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[alsa-devel] [PATCH] ASoC: Multi-component - Fix tlv320aic23 hw_write/hw_read functions
by Ryan Mallon 01 Jul '10

01 Jul '10

Commit 5ce5888769c4a453bb5a852e156ab4ac72822c40 removed the initialisations of the codec->hw_write/read functions for the tlv320aic23. Fix this by adding the initialisations in tlv320aic23_probe. Signed-off-by: Ryan Mallon <ryan(a)bluewatersys.com> --- diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index fbc95c5..fe89521 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -637,6 +637,8 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec) printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); codec->control_data = aic23->control_data; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->hw_read = NULL; /* Reset codec */ tlv320aic23_write(codec, TLV320AIC23_RESET, 0);

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[alsa-devel] [RFC 3/5] ASoC multi-component support : core
by Liam Girdwood 01 Jul '10

01 Jul '10

This patch adds multi-component support to the ASoC core and reflects the new ASoC driver/device structure model defined in soc.h and soc-dai.h. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/soc-core.c | 1595 +++++++++++++++++++++++++++++--------------------- 1 files changed, 914 insertions(+), 681 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3deb226..280c2f0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -39,6 +39,8 @@ #include <sound/soc-dapm.h> #include <sound/initval.h> +#define NAME_SIZE 32 + static DEFINE_MUTEX(pcm_mutex); static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); @@ -54,6 +56,7 @@ static LIST_HEAD(codec_list); static int snd_soc_register_card(struct snd_soc_card *card); static int snd_soc_unregister_card(struct snd_soc_card *card); +static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); /* * This is a timeout to do a DAPM powerdown after a stream is closed(). @@ -88,27 +91,27 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) { int i, step = 1, count = 0; - if (!codec->reg_cache_size) + if (!codec->driver->reg_cache_size) return 0; - if (codec->reg_cache_step) - step = codec->reg_cache_step; + if (codec->driver->reg_cache_step) + step = codec->driver->reg_cache_step; count += sprintf(buf, "%s registers\n", codec->name); - for (i = 0; i < codec->reg_cache_size; i += step) { - if (codec->readable_register && !codec->readable_register(i)) + for (i = 0; i < codec->driver->reg_cache_size; i += step) { + if (codec->driver->readable_register && !codec->driver->readable_register(i)) continue; count += sprintf(buf + count, "%2x: ", i); if (count >= PAGE_SIZE - 1) break; - if (codec->display_register) - count += codec->display_register(codec, buf + count, + if (codec->driver->display_register) + count += codec->driver->display_register(codec, buf + count, PAGE_SIZE - count, i); else count += snprintf(buf + count, PAGE_SIZE - count, - "%4x", codec->read(codec, i)); + "%4x", codec->driver->read(codec, i)); if (count >= PAGE_SIZE - 1) break; @@ -127,8 +130,8 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) static ssize_t codec_reg_show(struct device *dev, struct device_attribute *attr, char *buf) { - struct snd_soc_device *devdata = dev_get_drvdata(dev); - return soc_codec_reg_show(devdata->card->codec, buf); + struct snd_soc_codec *codec = dev_get_drvdata(dev); + return soc_codec_reg_show(codec, buf); } static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); @@ -136,20 +139,20 @@ static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); static ssize_t pmdown_time_show(struct device *dev, struct device_attribute *attr, char *buf) { - struct snd_soc_device *socdev = dev_get_drvdata(dev); - struct snd_soc_card *card = socdev->card; + struct snd_soc_pcm_runtime *rtd = + container_of(dev, struct snd_soc_pcm_runtime, dev); - return sprintf(buf, "%ld\n", card->pmdown_time); + return sprintf(buf, "%ld\n", rtd->pmdown_time); } static ssize_t pmdown_time_set(struct device *dev, struct device_attribute *attr, const char *buf, size_t count) { - struct snd_soc_device *socdev = dev_get_drvdata(dev); - struct snd_soc_card *card = socdev->card; + struct snd_soc_pcm_runtime *rtd = + container_of(dev, struct snd_soc_pcm_runtime, dev); - strict_strtol(buf, 10, &card->pmdown_time); + strict_strtol(buf, 10, &rtd->pmdown_time); return count; } @@ -193,19 +196,19 @@ static ssize_t codec_reg_write_file(struct file *file, return -EFAULT; buf[buf_size] = 0; - if (codec->reg_cache_step) - step = codec->reg_cache_step; + if (codec->driver->reg_cache_step) + step = codec->driver->reg_cache_step; while (*start == ' ') start++; reg = simple_strtoul(start, &start, 16); - if ((reg >= codec->reg_cache_size) || (reg % step)) + if ((reg >= codec->driver->reg_cache_size) || (reg % step)) return -EINVAL; while (*start == ' ') start++; if (strict_strtoul(start, 16, &value)) return -EINVAL; - codec->write(codec, reg, value); + codec->driver->write(codec, reg, value); return buf_size; } @@ -295,7 +298,7 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) codec->ac97->dev.release = soc_ac97_device_release; dev_set_name(&codec->ac97->dev, "%d-%d:%s", - codec->card->number, 0, codec->name); + codec->id, 0, codec->name); err = device_register(&codec->ac97->dev); if (err < 0) { snd_printk(KERN_ERR "Can't register ac97 bus\n"); @@ -309,24 +312,21 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_dai *codec_dai = machine->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - if (codec_dai->symmetric_rates || cpu_dai->symmetric_rates || - machine->symmetric_rates) { - dev_dbg(card->dev, "Symmetry forces %dHz rate\n", - machine->rate); + if (codec_dai->driver->symmetric_rates || cpu_dai->driver->symmetric_rates || + rtd->dai_link->symmetric_rates) { + dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", + rtd->rate); ret = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, - machine->rate, - machine->rate); + rtd->rate, + rtd->rate); if (ret < 0) { - dev_err(card->dev, + dev_err(&rtd->dev, "Unable to apply rate symmetry constraint: %d\n", ret); return ret; } @@ -343,48 +343,47 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) static int soc_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_card *card = socdev->card; struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_dai *codec_dai = machine->codec_dai; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; + struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; int ret = 0; mutex_lock(&pcm_mutex); /* startup the audio subsystem */ - if (cpu_dai->ops->startup) { - ret = cpu_dai->ops->startup(substream, cpu_dai); + if (cpu_dai->driver->ops->startup) { + ret = cpu_dai->driver->ops->startup(substream, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open interface %s\n", - cpu_dai->name); + cpu_dai->driver->name); goto out; } } - if (platform->pcm_ops->open) { - ret = platform->pcm_ops->open(substream); + if (platform->driver->ops->open) { + ret = platform->driver->ops->open(substream); if (ret < 0) { - printk(KERN_ERR "asoc: can't open platform %s\n", platform->name); + printk(KERN_ERR "asoc: can't open platform %s\n", platform->driver->name); goto platform_err; } } - if (codec_dai->ops->startup) { - ret = codec_dai->ops->startup(substream, codec_dai); + if (codec_dai->driver->ops->startup) { + ret = codec_dai->driver->ops->startup(substream, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open codec %s\n", - codec_dai->name); + codec_dai->driver->name); goto codec_dai_err; } } - if (machine->ops && machine->ops->startup) { - ret = machine->ops->startup(substream); + if (rtd->dai_link->ops && rtd->dai_link->ops->startup) { + ret = rtd->dai_link->ops->startup(substream); if (ret < 0) { - printk(KERN_ERR "asoc: %s startup failed\n", machine->name); + printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name); goto machine_err; } } @@ -392,66 +391,66 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) /* Check that the codec and cpu DAI's are compatible */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { runtime->hw.rate_min = - max(codec_dai->playback.rate_min, - cpu_dai->playback.rate_min); + max(codec_dai_drv->playback.rate_min, + cpu_dai_drv->playback.rate_min); runtime->hw.rate_max = - min(codec_dai->playback.rate_max, - cpu_dai->playback.rate_max); + min(codec_dai_drv->playback.rate_max, + cpu_dai_drv->playback.rate_max); runtime->hw.channels_min = - max(codec_dai->playback.channels_min, - cpu_dai->playback.channels_min); + max(codec_dai_drv->playback.channels_min, + cpu_dai_drv->playback.channels_min); runtime->hw.channels_max = - min(codec_dai->playback.channels_max, - cpu_dai->playback.channels_max); + min(codec_dai_drv->playback.channels_max, + cpu_dai_drv->playback.channels_max); runtime->hw.formats = - codec_dai->playback.formats & cpu_dai->playback.formats; + codec_dai_drv->playback.formats & cpu_dai_drv->playback.formats; runtime->hw.rates = - codec_dai->playback.rates & cpu_dai->playback.rates; - if (codec_dai->playback.rates + codec_dai_drv->playback.rates & cpu_dai_drv->playback.rates; + if (codec_dai_drv->playback.rates & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= cpu_dai->playback.rates; - if (cpu_dai->playback.rates + runtime->hw.rates |= cpu_dai_drv->playback.rates; + if (cpu_dai_drv->playback.rates & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= codec_dai->playback.rates; + runtime->hw.rates |= codec_dai_drv->playback.rates; } else { runtime->hw.rate_min = - max(codec_dai->capture.rate_min, - cpu_dai->capture.rate_min); + max(codec_dai_drv->capture.rate_min, + cpu_dai_drv->capture.rate_min); runtime->hw.rate_max = - min(codec_dai->capture.rate_max, - cpu_dai->capture.rate_max); + min(codec_dai_drv->capture.rate_max, + cpu_dai_drv->capture.rate_max); runtime->hw.channels_min = - max(codec_dai->capture.channels_min, - cpu_dai->capture.channels_min); + max(codec_dai_drv->capture.channels_min, + cpu_dai_drv->capture.channels_min); runtime->hw.channels_max = - min(codec_dai->capture.channels_max, - cpu_dai->capture.channels_max); + min(codec_dai_drv->capture.channels_max, + cpu_dai_drv->capture.channels_max); runtime->hw.formats = - codec_dai->capture.formats & cpu_dai->capture.formats; + codec_dai_drv->capture.formats & cpu_dai_drv->capture.formats; runtime->hw.rates = - codec_dai->capture.rates & cpu_dai->capture.rates; - if (codec_dai->capture.rates + codec_dai_drv->capture.rates & cpu_dai_drv->capture.rates; + if (codec_dai_drv->capture.rates & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= cpu_dai->capture.rates; - if (cpu_dai->capture.rates + runtime->hw.rates |= cpu_dai_drv->capture.rates; + if (cpu_dai_drv->capture.rates & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) - runtime->hw.rates |= codec_dai->capture.rates; + runtime->hw.rates |= codec_dai_drv->capture.rates; } snd_pcm_limit_hw_rates(runtime); if (!runtime->hw.rates) { printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", - codec_dai->name, cpu_dai->name); + codec_dai->driver->name, cpu_dai->driver->name); goto config_err; } if (!runtime->hw.formats) { printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", - codec_dai->name, cpu_dai->name); + codec_dai->driver->name, cpu_dai->driver->name); goto config_err; } if (!runtime->hw.channels_min || !runtime->hw.channels_max) { printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", - codec_dai->name, cpu_dai->name); + codec_dai->driver->name, cpu_dai->driver->name); goto config_err; } @@ -462,7 +461,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto config_err; } - pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name); + pr_debug("asoc: %s <-> %s info:\n", + codec_dai->driver->name, cpu_dai->driver->name); pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates); pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, runtime->hw.channels_max); @@ -470,33 +470,33 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) runtime->hw.rate_max); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback.active++; - codec_dai->playback.active++; + cpu_dai->playback_active++; + codec_dai->playback_active++; } else { - cpu_dai->capture.active++; - codec_dai->capture.active++; + cpu_dai->capture_active++; + codec_dai->capture_active++; } cpu_dai->active++; codec_dai->active++; - card->codec->active++; + rtd->codec->active++; mutex_unlock(&pcm_mutex); return 0; config_err: - if (machine->ops && machine->ops->shutdown) - machine->ops->shutdown(substream); + if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + rtd->dai_link->ops->shutdown(substream); machine_err: - if (codec_dai->ops->shutdown) - codec_dai->ops->shutdown(substream, codec_dai); + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); codec_dai_err: - if (platform->pcm_ops->close) - platform->pcm_ops->close(substream); + if (platform->driver->ops->close) + platform->driver->ops->close(substream); platform_err: - if (cpu_dai->ops->shutdown) - cpu_dai->ops->shutdown(substream, cpu_dai); + if (cpu_dai->driver->ops->shutdown) + cpu_dai->driver->ops->shutdown(substream, cpu_dai); out: mutex_unlock(&pcm_mutex); return ret; @@ -509,29 +509,25 @@ out: */ static void close_delayed_work(struct work_struct *work) { - struct snd_soc_card *card = container_of(work, struct snd_soc_card, - delayed_work.work); - struct snd_soc_codec *codec = card->codec; - struct snd_soc_dai *codec_dai; - int i; + struct snd_soc_pcm_runtime *rtd = + container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); + struct snd_soc_dai *codec_dai = rtd->codec_dai; mutex_lock(&pcm_mutex); - for (i = 0; i < codec->num_dai; i++) { - codec_dai = &codec->dai[i]; - - pr_debug("pop wq checking: %s status: %s waiting: %s\n", - codec_dai->playback.stream_name, - codec_dai->playback.active ? "active" : "inactive", - codec_dai->pop_wait ? "yes" : "no"); - - /* are we waiting on this codec DAI stream */ - if (codec_dai->pop_wait == 1) { - codec_dai->pop_wait = 0; - snd_soc_dapm_stream_event(codec, - codec_dai->playback.stream_name, - SND_SOC_DAPM_STREAM_STOP); - } + + pr_debug("pop wq checking: %s status: %s waiting: %s\n", + codec_dai->driver->playback.stream_name, + codec_dai->playback_active ? "active" : "inactive", + codec_dai->pop_wait ? "yes" : "no"); + + /* are we waiting on this codec DAI stream */ + if (codec_dai->pop_wait == 1) { + codec_dai->pop_wait = 0; + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->playback.stream_name, + SND_SOC_DAPM_STREAM_STOP); } + mutex_unlock(&pcm_mutex); } @@ -543,22 +539,19 @@ static void close_delayed_work(struct work_struct *work) static int soc_codec_close(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_dai *codec_dai = machine->codec_dai; - struct snd_soc_codec *codec = card->codec; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; mutex_lock(&pcm_mutex); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback.active--; - codec_dai->playback.active--; + cpu_dai->playback_active--; + codec_dai->playback_active--; } else { - cpu_dai->capture.active--; - codec_dai->capture.active--; + cpu_dai->capture_active--; + codec_dai->capture_active--; } cpu_dai->active--; @@ -571,27 +564,28 @@ static int soc_codec_close(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) snd_soc_dai_digital_mute(codec_dai, 1); - if (cpu_dai->ops->shutdown) - cpu_dai->ops->shutdown(substream, cpu_dai); + if (cpu_dai->driver->ops->shutdown) + cpu_dai->driver->ops->shutdown(substream, cpu_dai); - if (codec_dai->ops->shutdown) - codec_dai->ops->shutdown(substream, codec_dai); + if (codec_dai->driver->ops->shutdown) + codec_dai->driver->ops->shutdown(substream, codec_dai); - if (machine->ops && machine->ops->shutdown) - machine->ops->shutdown(substream); + if (rtd->dai_link->ops && rtd->dai_link->ops->shutdown) + rtd->dai_link->ops->shutdown(substream); - if (platform->pcm_ops->close) - platform->pcm_ops->close(substream); + if (platform->driver->ops->close) + platform->driver->ops->close(substream); + cpu_dai->runtime = NULL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* start delayed pop wq here for playback streams */ codec_dai->pop_wait = 1; - schedule_delayed_work(&card->delayed_work, - msecs_to_jiffies(card->pmdown_time)); + schedule_delayed_work(&rtd->delayed_work, + msecs_to_jiffies(rtd->pmdown_time)); } else { /* capture streams can be powered down now */ - snd_soc_dapm_stream_event(codec, - codec_dai->capture.stream_name, + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->capture.stream_name, SND_SOC_DAPM_STREAM_STOP); } @@ -607,43 +601,39 @@ static int soc_codec_close(struct snd_pcm_substream *substream) static int soc_pcm_prepare(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_dai *codec_dai = machine->codec_dai; - struct snd_soc_codec *codec = card->codec; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret = 0; mutex_lock(&pcm_mutex); - if (machine->ops && machine->ops->prepare) { - ret = machine->ops->prepare(substream); + if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) { + ret = rtd->dai_link->ops->prepare(substream); if (ret < 0) { printk(KERN_ERR "asoc: machine prepare error\n"); goto out; } } - if (platform->pcm_ops->prepare) { - ret = platform->pcm_ops->prepare(substream); + if (platform->driver->ops->prepare) { + ret = platform->driver->ops->prepare(substream); if (ret < 0) { printk(KERN_ERR "asoc: platform prepare error\n"); goto out; } } - if (codec_dai->ops->prepare) { - ret = codec_dai->ops->prepare(substream, codec_dai); + if (codec_dai->driver->ops->prepare) { + ret = codec_dai->driver->ops->prepare(substream, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: codec DAI prepare error\n"); goto out; } } - if (cpu_dai->ops->prepare) { - ret = cpu_dai->ops->prepare(substream, cpu_dai); + if (cpu_dai->driver->ops->prepare) { + ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: cpu DAI prepare error\n"); goto out; @@ -654,16 +644,16 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && codec_dai->pop_wait) { codec_dai->pop_wait = 0; - cancel_delayed_work(&card->delayed_work); + cancel_delayed_work(&rtd->delayed_work); } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dapm_stream_event(codec, - codec_dai->playback.stream_name, + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->playback.stream_name, SND_SOC_DAPM_STREAM_START); else - snd_soc_dapm_stream_event(codec, - codec_dai->capture.stream_name, + snd_soc_dapm_stream_event(rtd, + codec_dai->driver->capture.stream_name, SND_SOC_DAPM_STREAM_START); snd_soc_dai_digital_mute(codec_dai, 0); @@ -682,68 +672,65 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_card *card = socdev->card; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_dai *codec_dai = machine->codec_dai; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret = 0; mutex_lock(&pcm_mutex); - if (machine->ops && machine->ops->hw_params) { - ret = machine->ops->hw_params(substream, params); + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { + ret = rtd->dai_link->ops->hw_params(substream, params); if (ret < 0) { printk(KERN_ERR "asoc: machine hw_params failed\n"); goto out; } } - if (codec_dai->ops->hw_params) { - ret = codec_dai->ops->hw_params(substream, params, codec_dai); + if (codec_dai->driver->ops->hw_params) { + ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't set codec %s hw params\n", - codec_dai->name); + codec_dai->driver->name); goto codec_err; } } - if (cpu_dai->ops->hw_params) { - ret = cpu_dai->ops->hw_params(substream, params, cpu_dai); + if (cpu_dai->driver->ops->hw_params) { + ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: interface %s hw params failed\n", - cpu_dai->name); + cpu_dai->driver->name); goto interface_err; } } - if (platform->pcm_ops->hw_params) { - ret = platform->pcm_ops->hw_params(substream, params); + if (platform->driver->ops->hw_params) { + ret = platform->driver->ops->hw_params(substream, params); if (ret < 0) { printk(KERN_ERR "asoc: platform %s hw params failed\n", - platform->name); + platform->driver->name); goto platform_err; } } - machine->rate = params_rate(params); + rtd->rate = params_rate(params); out: mutex_unlock(&pcm_mutex); return ret; platform_err: - if (cpu_dai->ops->hw_free) - cpu_dai->ops->hw_free(substream, cpu_dai); + if (cpu_dai->driver->ops->hw_free) + cpu_dai->driver->ops->hw_free(substream, cpu_dai); interface_err: - if (codec_dai->ops->hw_free) - codec_dai->ops->hw_free(substream, codec_dai); + if (codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); codec_err: - if (machine->ops && machine->ops->hw_free) - machine->ops->hw_free(substream); + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + rtd->dai_link->ops->hw_free(substream); mutex_unlock(&pcm_mutex); return ret; @@ -755,13 +742,10 @@ codec_err: static int soc_pcm_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_card *card = socdev->card; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_dai *codec_dai = machine->codec_dai; - struct snd_soc_codec *codec = card->codec; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; mutex_lock(&pcm_mutex); @@ -770,19 +754,19 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) snd_soc_dai_digital_mute(codec_dai, 1); /* free any machine hw params */ - if (machine->ops && machine->ops->hw_free) - machine->ops->hw_free(substream); + if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) + rtd->dai_link->ops->hw_free(substream); /* free any DMA resources */ - if (platform->pcm_ops->hw_free) - platform->pcm_ops->hw_free(substream); + if (platform->driver->ops->hw_free) + platform->driver->ops->hw_free(substream); /* now free hw params for the DAI's */ - if (codec_dai->ops->hw_free) - codec_dai->ops->hw_free(substream, codec_dai); + if (codec_dai->driver->ops->hw_free) + codec_dai->driver->ops->hw_free(substream, codec_dai); - if (cpu_dai->ops->hw_free) - cpu_dai->ops->hw_free(substream, cpu_dai); + if (cpu_dai->driver->ops->hw_free) + cpu_dai->driver->ops->hw_free(substream, cpu_dai); mutex_unlock(&pcm_mutex); return 0; @@ -791,28 +775,25 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_card *card= socdev->card; - struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_dai *codec_dai = machine->codec_dai; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - if (codec_dai->ops->trigger) { - ret = codec_dai->ops->trigger(substream, cmd, codec_dai); + if (codec_dai->driver->ops->trigger) { + ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); if (ret < 0) return ret; } - if (platform->pcm_ops->trigger) { - ret = platform->pcm_ops->trigger(substream, cmd); + if (platform->driver->ops->trigger) { + ret = platform->driver->ops->trigger(substream, cmd); if (ret < 0) return ret; } - if (cpu_dai->ops->trigger) { - ret = cpu_dai->ops->trigger(substream, cmd, cpu_dai); + if (cpu_dai->driver->ops->trigger) { + ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai); if (ret < 0) return ret; } @@ -827,27 +808,24 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_card *card = socdev->card; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_dai_link *machine = rtd->dai; - struct snd_soc_dai *cpu_dai = machine->cpu_dai; - struct snd_soc_dai *codec_dai = machine->codec_dai; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t offset = 0; snd_pcm_sframes_t delay = 0; - if (platform->pcm_ops->pointer) - offset = platform->pcm_ops->pointer(substream); + if (platform->driver->ops->pointer) + offset = platform->driver->ops->pointer(substream); - if (cpu_dai->ops->delay) - delay += cpu_dai->ops->delay(substream, cpu_dai); + if (cpu_dai->driver->ops->delay) + delay += cpu_dai->driver->ops->delay(substream, cpu_dai); - if (codec_dai->ops->delay) - delay += codec_dai->ops->delay(substream, codec_dai); + if (codec_dai->driver->ops->delay) + delay += codec_dai->driver->ops->delay(substream, codec_dai); - if (platform->delay) - delay += platform->delay(substream, codec_dai); + if (platform->driver->delay) + delay += platform->driver->delay(substream, codec_dai); runtime->delay = delay; @@ -870,104 +848,111 @@ static struct snd_pcm_ops soc_pcm_ops = { static int soc_suspend(struct device *dev) { struct platform_device *pdev = to_platform_device(dev); - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_codec_device *codec_dev = socdev->codec_dev; - struct snd_soc_codec *codec = card->codec; + struct snd_soc_card *card = platform_get_drvdata(pdev); int i; /* If the initialization of this soc device failed, there is no codec * associated with it. Just bail out in this case. */ - if (!codec) + if (list_empty(&card->codec_dev_list)) return 0; /* Due to the resume being scheduled into a workqueue we could * suspend before that's finished - wait for it to complete. */ - snd_power_lock(codec->card); - snd_power_wait(codec->card, SNDRV_CTL_POWER_D0); - snd_power_unlock(codec->card); + snd_power_lock(card->snd_card); + snd_power_wait(card->snd_card, SNDRV_CTL_POWER_D0); + snd_power_unlock(card->snd_card); /* we're going to block userspace touching us until resume completes */ - snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3hot); + snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D3hot); /* mute any active DAC's */ - for (i = 0; i < card->num_links; i++) { - struct snd_soc_dai *dai = card->dai_link[i].codec_dai; + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *dai = card->rtd[i].codec_dai; + struct snd_soc_dai_driver *drv = dai->driver; - if (card->dai_link[i].ignore_suspend) + if (card->rtd[i].dai_link->ignore_suspend) continue; - if (dai->ops->digital_mute && dai->playback.active) - dai->ops->digital_mute(dai, 1); + if (drv->ops->digital_mute && dai->playback_active) + drv->ops->digital_mute(dai, 1); } /* suspend all pcms */ - for (i = 0; i < card->num_links; i++) { - if (card->dai_link[i].ignore_suspend) + for (i = 0; i < card->num_rtd; i++) { + if (card->rtd[i].dai_link->ignore_suspend) continue; - snd_pcm_suspend_all(card->dai_link[i].pcm); + snd_pcm_suspend_all(card->rtd[i].pcm); } if (card->suspend_pre) card->suspend_pre(pdev, PMSG_SUSPEND); - for (i = 0; i < card->num_links; i++) { - struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + struct snd_soc_platform *platform = card->rtd[i].platform; - if (card->dai_link[i].ignore_suspend) + if (card->rtd[i].dai_link->ignore_suspend) continue; - if (cpu_dai->suspend && !cpu_dai->ac97_control) - cpu_dai->suspend(cpu_dai); - if (platform->suspend) - platform->suspend(&card->dai_link[i]); + if (cpu_dai->driver->suspend && !cpu_dai->driver->ac97_control) + cpu_dai->driver->suspend(cpu_dai); + if (platform->driver->suspend && !platform->suspended) { + platform->driver->suspend(cpu_dai); + platform->suspended = 1; + } } /* close any waiting streams and save state */ - run_delayed_work(&card->delayed_work); - codec->suspend_bias_level = codec->bias_level; + for (i = 0; i < card->num_rtd; i++) { + run_delayed_work(&card->rtd[i].delayed_work); + card->rtd[i].codec->suspend_bias_level = card->rtd[i].codec->bias_level; + } - for (i = 0; i < codec->num_dai; i++) { - char *stream = codec->dai[i].playback.stream_name; + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai_driver *driver = card->rtd[i].codec_dai->driver; - if (card->dai_link[i].ignore_suspend) + if (card->rtd[i].dai_link->ignore_suspend) continue; - if (stream != NULL) - snd_soc_dapm_stream_event(codec, stream, + if (driver->playback.stream_name != NULL) + snd_soc_dapm_stream_event(&card->rtd[i], driver->playback.stream_name, SND_SOC_DAPM_STREAM_SUSPEND); - stream = codec->dai[i].capture.stream_name; - if (stream != NULL) - snd_soc_dapm_stream_event(codec, stream, + + if (driver->capture.stream_name != NULL) + snd_soc_dapm_stream_event(&card->rtd[i], driver->capture.stream_name, SND_SOC_DAPM_STREAM_SUSPEND); } - /* If there are paths active then the CODEC will be held with - * bias _ON and should not be suspended. */ - if (codec_dev->suspend) { - switch (codec->bias_level) { - case SND_SOC_BIAS_STANDBY: - case SND_SOC_BIAS_OFF: - codec_dev->suspend(pdev, PMSG_SUSPEND); - break; - default: - dev_dbg(socdev->dev, "CODEC is on over suspend\n"); - break; + /* suspend all CODECs */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_codec *codec = card->rtd[i].codec; + /* If there are paths active then the CODEC will be held with + * bias _ON and should not be suspended. */ + if (!codec->suspended && codec->driver->suspend) { + switch (codec->bias_level) { + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + codec->driver->suspend(codec, PMSG_SUSPEND); + codec->suspended = 1; + break; + default: + dev_dbg(codec->dev, "CODEC is on over suspend\n"); + break; + } } } - for (i = 0; i < card->num_links; i++) { - struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; - if (card->dai_link[i].ignore_suspend) + if (card->rtd[i].dai_link->ignore_suspend) continue; - if (cpu_dai->suspend && cpu_dai->ac97_control) - cpu_dai->suspend(cpu_dai); + if (cpu_dai->driver->suspend && cpu_dai->driver->ac97_control) + cpu_dai->driver->suspend(cpu_dai); } if (card->suspend_post) @@ -981,127 +966,127 @@ static int soc_suspend(struct device *dev) */ static void soc_resume_deferred(struct work_struct *work) { - struct snd_soc_card *card = container_of(work, - struct snd_soc_card, - deferred_resume_work); - struct snd_soc_device *socdev = card->socdev; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_codec_device *codec_dev = socdev->codec_dev; - struct snd_soc_codec *codec = card->codec; - struct platform_device *pdev = to_platform_device(socdev->dev); + struct snd_soc_card *card = + container_of(work, struct snd_soc_card, deferred_resume_work); + struct platform_device *pdev = to_platform_device(card->dev); int i; /* our power state is still SNDRV_CTL_POWER_D3hot from suspend time, * so userspace apps are blocked from touching us */ - dev_dbg(socdev->dev, "starting resume work\n"); + dev_dbg(card->dev, "starting resume work\n"); /* Bring us up into D2 so that DAPM starts enabling things */ - snd_power_change_state(codec->card, SNDRV_CTL_POWER_D2); + snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D2); if (card->resume_pre) card->resume_pre(pdev); - for (i = 0; i < card->num_links; i++) { - struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; + /* resume AC97 DAIs */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; - if (card->dai_link[i].ignore_suspend) + if (card->rtd[i].dai_link->ignore_suspend) continue; - if (cpu_dai->resume && cpu_dai->ac97_control) - cpu_dai->resume(cpu_dai); - } - - /* If the CODEC was idle over suspend then it will have been - * left with bias OFF or STANDBY and suspended so we must now - * resume. Otherwise the suspend was suppressed. - */ - if (codec_dev->resume) { - switch (codec->bias_level) { - case SND_SOC_BIAS_STANDBY: - case SND_SOC_BIAS_OFF: - codec_dev->resume(pdev); - break; - default: - dev_dbg(socdev->dev, "CODEC was on over suspend\n"); - break; + if (cpu_dai->driver->resume && cpu_dai->driver->ac97_control) + cpu_dai->driver->resume(cpu_dai); + } + + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_codec *codec = card->rtd[i].codec; + /* If the CODEC was idle over suspend then it will have been + * left with bias OFF or STANDBY and suspended so we must now + * resume. Otherwise the suspend was suppressed. + */ + if (codec->driver->resume && codec->suspended) { + switch (codec->bias_level) { + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + codec->driver->resume(codec); + codec->suspended = 0; + break; + default: + dev_dbg(codec->dev, "CODEC was on over suspend\n"); + break; + } } } - for (i = 0; i < codec->num_dai; i++) { - char *stream = codec->dai[i].playback.stream_name; + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai_driver *driver = card->rtd[i].codec_dai->driver; - if (card->dai_link[i].ignore_suspend) + if (card->rtd[i].dai_link->ignore_suspend) continue; - if (stream != NULL) - snd_soc_dapm_stream_event(codec, stream, + if (driver->playback.stream_name != NULL) + snd_soc_dapm_stream_event(&card->rtd[i], driver->playback.stream_name, SND_SOC_DAPM_STREAM_RESUME); - stream = codec->dai[i].capture.stream_name; - if (stream != NULL) - snd_soc_dapm_stream_event(codec, stream, + + if (driver->capture.stream_name != NULL) + snd_soc_dapm_stream_event(&card->rtd[i], driver->capture.stream_name, SND_SOC_DAPM_STREAM_RESUME); } /* unmute any active DACs */ - for (i = 0; i < card->num_links; i++) { - struct snd_soc_dai *dai = card->dai_link[i].codec_dai; + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *dai = card->rtd[i].codec_dai; + struct snd_soc_dai_driver *drv = dai->driver; - if (card->dai_link[i].ignore_suspend) + if (card->rtd[i].dai_link->ignore_suspend) continue; - if (dai->ops->digital_mute && dai->playback.active) - dai->ops->digital_mute(dai, 0); + if (drv->ops->digital_mute && dai->playback_active) + drv->ops->digital_mute(dai, 0); } - for (i = 0; i < card->num_links; i++) { - struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + struct snd_soc_platform *platform = card->rtd[i].platform; - if (card->dai_link[i].ignore_suspend) + if (card->rtd[i].dai_link->ignore_suspend) continue; - if (cpu_dai->resume && !cpu_dai->ac97_control) - cpu_dai->resume(cpu_dai); - if (platform->resume) - platform->resume(&card->dai_link[i]); + if (cpu_dai->driver->resume && !cpu_dai->driver->ac97_control) + cpu_dai->driver->resume(cpu_dai); + if (platform->driver->resume && platform->suspended) { + platform->driver->resume(cpu_dai); + platform->suspended = 0; + } } if (card->resume_post) card->resume_post(pdev); - dev_dbg(socdev->dev, "resume work completed\n"); + dev_dbg(card->dev, "resume work completed\n"); /* userspace can access us now we are back as we were before */ - snd_power_change_state(codec->card, SNDRV_CTL_POWER_D0); + snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D0); } /* powers up audio subsystem after a suspend */ static int soc_resume(struct device *dev) { struct platform_device *pdev = to_platform_device(dev); - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - struct snd_soc_dai *cpu_dai = card->dai_link[0].cpu_dai; - - /* If the initialization of this soc device failed, there is no codec - * associated with it. Just bail out in this case. - */ - if (!card->codec) - return 0; + struct snd_soc_card *card = platform_get_drvdata(pdev); + int i; /* AC97 devices might have other drivers hanging off them so * need to resume immediately. Other drivers don't have that * problem and may take a substantial amount of time to resume * due to I/O costs and anti-pop so handle them out of line. */ - if (cpu_dai->ac97_control) { - dev_dbg(socdev->dev, "Resuming AC97 immediately\n"); - soc_resume_deferred(&card->deferred_resume_work); - } else { - dev_dbg(socdev->dev, "Scheduling resume work\n"); - if (!schedule_work(&card->deferred_resume_work)) - dev_err(socdev->dev, "resume work item may be lost\n"); + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + if (cpu_dai->driver->ac97_control) { + dev_dbg(dev, "Resuming AC97 immediately\n"); + soc_resume_deferred(&card->deferred_resume_work); + } else { + dev_dbg(dev, "Scheduling resume work\n"); + if (!schedule_work(&card->deferred_resume_work)) + dev_err(dev, "resume work item may be lost\n"); + } } return 0; @@ -1114,198 +1099,408 @@ static int soc_resume(struct device *dev) static struct snd_soc_dai_ops null_dai_ops = { }; -static void snd_soc_instantiate_card(struct snd_soc_card *card) +static void soc_bind_dai_link(struct snd_soc_card *card, int num) { - struct platform_device *pdev = container_of(card->dev, - struct platform_device, - dev); - struct snd_soc_codec_device *codec_dev = card->socdev->codec_dev; + struct snd_soc_dai_link *dai_link = &card->dai_link[num]; + struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_codec *codec; struct snd_soc_platform *platform; - struct snd_soc_dai *dai; - int i, found, ret, ac97; + struct snd_soc_dai *codec_dai, *cpu_dai; + int devices = 0; - if (card->instantiated) + if (rtd->complete) return; - found = 0; - list_for_each_entry(platform, &platform_list, list) - if (card->platform == platform) { - found = 1; - break; + /* do we already have the CPU DAI for this link ? */ + if (rtd->cpu_dai) { + devices++; + goto find_codec_dai; + } + /* no, then find CPU DAI from registered DAIs*/ + list_for_each_entry(cpu_dai, &dai_list, list) { + if ((cpu_dai->id == dai_link->cpu_dai_id || cpu_dai->id == -1) && + cpu_dai->driver == dai_link->cpu_dai_drv) { + rtd->cpu_dai = cpu_dai; + devices++; + goto find_codec_dai; } - if (!found) { - dev_dbg(card->dev, "Platform %s not registered\n", - card->platform->name); - return; } - - ac97 = 0; - for (i = 0; i < card->num_links; i++) { - found = 0; - list_for_each_entry(dai, &dai_list, list) - if (card->dai_link[i].cpu_dai == dai) { - found = 1; - break; - } - if (!found) { - dev_dbg(card->dev, "DAI %s not registered\n", - card->dai_link[i].cpu_dai->name); - return; + dev_dbg(card->dev, "CPU DAI %s not registered\n", + dai_link->cpu_dai_drv->name); + +find_codec_dai: + /* do we already have the CODEC DAI for this link ? */ + if (rtd->codec_dai) { + devices++; + goto find_codec; + } + /* no, then find CPU DAI from registered DAIs*/ + list_for_each_entry(codec_dai, &dai_list, list) { + if ((codec_dai->id == dai_link->codec_dai_id || codec_dai->id == -1) && + codec_dai->driver == dai_link->codec_dai_drv) { + rtd->codec_dai = codec_dai; + devices++; + goto find_codec; + } + } + dev_dbg(card->dev, "CODEC DAI %s not registered\n", + dai_link->codec_dai_drv->name); + +find_codec: + /* do we already have the CODEC for this link ? */ + if (rtd->codec) { + devices++; + goto find_platform; + } + /* no, then find CODEC from registered CODECs*/ + list_for_each_entry(codec, &codec_list, list) { + if ((codec->id == dai_link->codec_id || codec->id == -1) && + codec->driver == dai_link->codec_drv) { + rtd->codec = codec; + devices++; + goto find_platform; } + } + dev_dbg(card->dev, "CODEC %s not registered\n", + dai_link->codec_drv->name); + +find_platform: + /* do we already have the CODEC DAI for this link ? */ + if (rtd->platform) { + devices++; + goto out; + } + /* no, then find CPU DAI from registered DAIs*/ + list_for_each_entry(platform, &platform_list, list) { + if ((platform->id == dai_link->platform_id || platform->id == -1) && + platform->driver == dai_link->platform_drv) { + rtd->platform = platform; + devices++; + goto out; + } + } - if (card->dai_link[i].cpu_dai->ac97_control) - ac97 = 1; + dev_dbg(card->dev, "platform %s not registered\n", + dai_link->platform_drv->name); + return; + +out: + /* mark rtd as complete if we found all 4 of our client devices */ + if (devices == 4) { + rtd->complete = 1; + card->num_rtd++; } +} - for (i = 0; i < card->num_links; i++) { - if (!card->dai_link[i].codec_dai->ops) - card->dai_link[i].codec_dai->ops = &null_dai_ops; +static void soc_remove_dai_link(struct snd_soc_card *card, int num) +{ + struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai; + int err; + + /* unregister the rtd device */ + if (rtd->dev_registered) { + device_remove_file(&rtd->dev, &dev_attr_pmdown_time); + device_unregister(&rtd->dev); + rtd->dev_registered = 0; } - /* If we have AC97 in the system then don't wait for the - * codec. This will need revisiting if we have to handle - * systems with mixed AC97 and non-AC97 parts. Only check for - * DAIs currently; we can't do this per link since some AC97 - * codecs have non-AC97 DAIs. - */ - if (!ac97) - for (i = 0; i < card->num_links; i++) { - found = 0; - list_for_each_entry(dai, &dai_list, list) - if (card->dai_link[i].codec_dai == dai) { - found = 1; - break; - } - if (!found) { - dev_dbg(card->dev, "DAI %s not registered\n", - card->dai_link[i].codec_dai->name); - return; - } + /* remove the CODEC DAI */ + if (codec_dai && codec_dai->probed) { + if (codec_dai->driver->remove) { + err = codec_dai->driver->remove(codec_dai); + if (err < 0) + printk(KERN_ERR "asoc: failed to remove %s\n", codec_dai->name); } + codec_dai->probed = 0; + list_del(&codec_dai->card_list); + } - /* Note that we do not current check for codec components */ + /* remove the platform */ + if (platform && platform->probed) { + if (platform->driver->remove) { + err = platform->driver->remove(platform); + if (err < 0) + printk(KERN_ERR "asoc: failed to remove %s\n", platform->name); + } + platform->probed = 0; + list_del(&platform->card_list); + } - dev_dbg(card->dev, "All components present, instantiating\n"); + /* remove the CODEC */ + if (codec && codec->probed) { + if (codec->driver->remove) { + err = codec->driver->remove(codec); + if (err < 0) + printk(KERN_ERR "asoc: failed to remove %s\n", codec->name); + } - /* Found everything, bring it up */ - card->pmdown_time = pmdown_time; + /* Make sure all DAPM widgets are freed */ + snd_soc_dapm_free(codec); - if (card->probe) { - ret = card->probe(pdev); - if (ret < 0) - return; + soc_cleanup_codec_debugfs(codec); + device_remove_file(codec->dev, &dev_attr_codec_reg); + codec->probed = 0; + list_del(&codec->card_list); } - for (i = 0; i < card->num_links; i++) { - struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; - if (cpu_dai->probe) { - ret = cpu_dai->probe(pdev, cpu_dai); + /* remove the cpu_dai */ + if (cpu_dai && cpu_dai->probed) { + if (cpu_dai->driver->remove) { + err = cpu_dai->driver->remove(cpu_dai); + if (err < 0) + printk(KERN_ERR "asoc: failed to remove %s\n", cpu_dai->name); + } + cpu_dai->probed = 0; + list_del(&cpu_dai->card_list); + } +} + +static int soc_probe_dai_link(struct snd_soc_card *card, int num) +{ + struct snd_soc_dai_link *dai_link = &card->dai_link[num]; + struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai; + int ret; + + dev_dbg(card->dev, "probe %s dai link %d\n", card->name, num); + + /* config components */ + codec->snd_card = card->snd_card; + codec_dai->codec = codec; + codec->card = card; + cpu_dai->platform = platform; + rtd->card = card; + rtd->dev.parent = card->dev; + + /* set default power off timeout */ + rtd->pmdown_time = pmdown_time; + + /* probe the cpu_dai */ + if (!cpu_dai->probed) { + if (cpu_dai->driver->probe) { + ret = cpu_dai->driver->probe(cpu_dai); if (ret < 0) - goto cpu_dai_err; + return ret; } + cpu_dai->probed = 1; + /* mark cpu_dai as probed and add to card cpu_dai list */ + list_add(&cpu_dai->card_list, &card->dai_dev_list); } - if (codec_dev->probe) { - ret = codec_dev->probe(pdev); + /* probe the CODEC */ + if (!codec->probed) { + if (codec->driver->probe) { + ret = codec->driver->probe(codec); + if (ret < 0) + return ret; + } + /* Make sure all DAPM widgets are instantiated */ + snd_soc_dapm_new_widgets(codec); + + /* mark codec as probed and add to card codec list */ + codec->probed = 1; + list_add(&codec->card_list, &card->codec_dev_list); + + /* add DAPM sysfs entries for this codec */ + ret = snd_soc_dapm_sys_add(codec->dev); if (ret < 0) - goto cpu_dai_err; - } - codec = card->codec; + printk(KERN_WARNING "asoc: failed to add codec dapm sysfs entries\n"); - if (platform->probe) { - ret = platform->probe(pdev); + /* add codec sysfs entries */ + ret = device_create_file(codec->dev, &dev_attr_codec_reg); if (ret < 0) - goto platform_err; + printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); + + soc_init_codec_debugfs(codec); } - /* DAPM stream work */ - INIT_DELAYED_WORK(&card->delayed_work, close_delayed_work); -#ifdef CONFIG_PM - /* deferred resume work */ - INIT_WORK(&card->deferred_resume_work, soc_resume_deferred); -#endif + /* probe the platform */ + if (!platform->probed) { + if (platform->driver->probe) { + ret = platform->driver->probe(platform); + if (ret < 0) + return ret; + } + /* mark platform as probed and add to card platform list */ + platform->probed = 1; + list_add(&platform->card_list, &card->platform_dev_list); + } - for (i = 0; i < card->num_links; i++) { - if (card->dai_link[i].init) { - ret = card->dai_link[i].init(codec); - if (ret < 0) { - printk(KERN_ERR "asoc: failed to init %s\n", - card->dai_link[i].stream_name); - continue; - } + /* probe the CODEC DAI */ + if (!codec_dai->probed) { + if (codec_dai->driver->probe) { + ret = codec_dai->driver->probe(codec_dai); + if (ret < 0) + return ret; } - if (card->dai_link[i].codec_dai->ac97_control) - ac97 = 1; + + /* mark cpu_dai as probed and add to card cpu_dai list */ + codec_dai->probed = 1; + list_add(&codec_dai->card_list, &card->dai_dev_list); } - snprintf(codec->card->shortname, sizeof(codec->card->shortname), - "%s", card->name); - snprintf(codec->card->longname, sizeof(codec->card->longname), - "%s (%s)", card->name, codec->name); + /* DAPM dai link stream work */ + INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); - /* Make sure all DAPM widgets are instantiated */ - snd_soc_dapm_new_widgets(codec); + /* now that all clients have probed, initialise the DAI link */ + if (dai_link->init) { + ret = dai_link->init(rtd); + if (ret < 0) { + printk(KERN_ERR "asoc: failed to init %s\n", dai_link->stream_name); + return ret; + } + } - ret = snd_card_register(codec->card); + /* register the rtd device */ + rtd->dev.init_name = rtd->dai_link->stream_name; + ret = device_register(&rtd->dev); if (ret < 0) { - printk(KERN_ERR "asoc: failed to register soundcard for %s\n", - codec->name); - goto card_err; + printk(KERN_ERR "asoc: failed to register DAI runtime device %d\n", ret); + return ret; } + rtd->dev_registered = 1; + ret = device_create_file(&rtd->dev, &dev_attr_pmdown_time); + if (ret < 0) + printk(KERN_WARNING "asoc: failed to add pmdown_time sysfs\n"); + + /* create the pcm */ + ret = soc_new_pcm(rtd, num); + if (ret < 0) { + printk(KERN_ERR "asoc: can't create pcm %s\n", dai_link->stream_name); + return ret; + } + + /* add platform data for AC97 devices */ + if (dai_link->codec_dai_drv->ac97_control) + snd_ac97_dev_add_pdata(codec->ac97, rtd->cpu_dai->ac97_pdata); + + return 0; +} - mutex_lock(&codec->mutex); #ifdef CONFIG_SND_SOC_AC97_BUS +static int soc_register_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + /* Only instantiate AC97 if not already done by the adaptor * for the generic AC97 subsystem. */ - if (ac97 && strcmp(codec->name, "AC97") != 0) { - ret = soc_ac97_dev_register(codec); + if (rtd->codec_dai->driver->ac97_control && !rtd->codec->ac97_registered) { + + ret = soc_ac97_dev_register(rtd->codec); if (ret < 0) { printk(KERN_ERR "asoc: AC97 device register failed\n"); - snd_card_free(codec->card); - mutex_unlock(&codec->mutex); - goto card_err; + return ret; } + + rtd->codec->ac97_registered = 1; } + return 0; +} + +static void soc_unregister_ac97_dai_link(struct snd_soc_codec *codec) +{ + if (codec->ac97_registered) { + soc_ac97_dev_unregister(codec); + codec->ac97_registered = 0; + } +} #endif - ret = snd_soc_dapm_sys_add(card->socdev->dev); - if (ret < 0) - printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n"); +static void snd_soc_instantiate_card(struct snd_soc_card *card) +{ + struct platform_device *pdev = to_platform_device(card->dev); + int ret, i; - ret = device_create_file(card->socdev->dev, &dev_attr_pmdown_time); - if (ret < 0) - printk(KERN_WARNING "asoc: failed to add pmdown_time sysfs\n"); + mutex_lock(&card->mutex); - ret = device_create_file(card->socdev->dev, &dev_attr_codec_reg); - if (ret < 0) - printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); + if (card->instantiated) { + mutex_unlock(&card->mutex); + return; + } - soc_init_codec_debugfs(codec); - mutex_unlock(&codec->mutex); + /* bind DAIs */ + for (i = 0; i < card->num_links; i++) + soc_bind_dai_link(card, i); - card->instantiated = 1; + /* bind completed ? */ + if (card->num_rtd != card->num_links) { + mutex_unlock(&card->mutex); + return; + } - return; + /* card bind complete so register a sound card */ + ret = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, + card->owner, 0, &card->snd_card); + if (ret < 0) { + printk(KERN_ERR "asoc: can't create sound card for card %s\n", + card->name); + mutex_unlock(&card->mutex); + return; + } + card->snd_card->dev = card->dev; -card_err: - if (platform->remove) - platform->remove(pdev); + /* initialise the sound card only once */ + if (card->probe) { +#ifdef CONFIG_PM + /* deferred resume work */ + INIT_WORK(&card->deferred_resume_work, soc_resume_deferred); +#endif + ret = card->probe(pdev); + if (ret < 0) + goto card_probe_error; + } -platform_err: - if (codec_dev->remove) - codec_dev->remove(pdev); + for (i = 0; i < card->num_links; i++) { + ret = soc_probe_dai_link(card, i); + if (ret < 0) { + printk(KERN_ERR "asoc: failed to instanciate card %s\n", card->name); + goto probe_dai_err; + } + } + + snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), + "%s", card->name); + snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), + "%s", card->name); -cpu_dai_err: - for (i--; i >= 0; i--) { - struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; - if (cpu_dai->remove) - cpu_dai->remove(pdev, cpu_dai); + ret = snd_card_register(card->snd_card); + if (ret < 0) { + printk(KERN_ERR "asoc: failed to register soundcard for %s\n", card->name); + goto probe_dai_err; } +#ifdef CONFIG_SND_SOC_AC97_BUS + /* register any AC97 codecs */ + for (i = 0; i < card->num_rtd; i++) { + ret = soc_register_ac97_dai_link(&card->rtd[i]); + if (ret < 0) { + printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name); + goto probe_dai_err; + } + } +#endif + + card->instantiated = 1; + mutex_unlock(&card->mutex); + return; + +probe_dai_err: + for (i = 0; i < card->num_links; i++) + soc_remove_dai_link(card, i); + +card_probe_error: if (card->remove) card->remove(pdev); + + snd_card_free(card->snd_card); + + mutex_unlock(&card->mutex); } /* @@ -1322,15 +1517,15 @@ static void snd_soc_instantiate_cards(void) /* probes a new socdev */ static int soc_probe(struct platform_device *pdev) { + struct snd_soc_card *card = platform_get_drvdata(pdev); int ret = 0; - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - - /* Bodge while we push things out of socdev */ - card->socdev = socdev; /* Bodge while we unpick instantiation */ card->dev = &pdev->dev; + INIT_LIST_HEAD(&card->dai_dev_list); + INIT_LIST_HEAD(&card->codec_dev_list); + INIT_LIST_HEAD(&card->platform_dev_list); + ret = snd_soc_register_card(card); if (ret != 0) { dev_err(&pdev->dev, "Failed to register card\n"); @@ -1343,50 +1538,49 @@ static int soc_probe(struct platform_device *pdev) /* removes a socdev */ static int soc_remove(struct platform_device *pdev) { + struct snd_soc_card *card = platform_get_drvdata(pdev); int i; - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_codec_device *codec_dev = socdev->codec_dev; - if (card->instantiated) { - run_delayed_work(&card->delayed_work); - - if (platform->remove) - platform->remove(pdev); - - if (codec_dev->remove) - codec_dev->remove(pdev); + if (card->instantiated) { - for (i = 0; i < card->num_links; i++) { - struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; - if (cpu_dai->remove) - cpu_dai->remove(pdev, cpu_dai); + /* make sure any delayed work runs */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + run_delayed_work(&rtd->delayed_work); } + /* remove and free each DAI */ + for (i = 0; i < card->num_rtd; i++) + soc_remove_dai_link(card, i); + + /* remove the card */ if (card->remove) card->remove(pdev); - } + kfree(card->rtd); + snd_card_free(card->snd_card); + } snd_soc_unregister_card(card); - return 0; } static int soc_poweroff(struct device *dev) { struct platform_device *pdev = to_platform_device(dev); - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_card *card = socdev->card; + struct snd_soc_card *card = platform_get_drvdata(pdev); + int i; if (!card->instantiated) return 0; /* Flush out pmdown_time work - we actually do want to run it * now, we're shutting down so no imminent restart. */ - run_delayed_work(&card->delayed_work); + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + run_delayed_work(&rtd->delayed_work); + } - snd_soc_dapm_shutdown(socdev); + snd_soc_dapm_shutdown(card); return 0; } @@ -1409,53 +1603,42 @@ static struct platform_driver soc_driver = { }; /* create a new pcm */ -static int soc_new_pcm(struct snd_soc_device *socdev, - struct snd_soc_dai_link *dai_link, int num) -{ - struct snd_soc_card *card = socdev->card; - struct snd_soc_codec *codec = card->codec; - struct snd_soc_platform *platform = card->platform; - struct snd_soc_dai *codec_dai = dai_link->codec_dai; - struct snd_soc_dai *cpu_dai = dai_link->cpu_dai; - struct snd_soc_pcm_runtime *rtd; +static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; - rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime), GFP_KERNEL); - if (rtd == NULL) - return -ENOMEM; - - rtd->dai = dai_link; - rtd->socdev = socdev; - codec_dai->codec = card->codec; - /* check client and interface hw capabilities */ snprintf(new_name, sizeof(new_name), "%s %s-%d", - dai_link->stream_name, codec_dai->name, num); + rtd->dai_link->stream_name, codec_dai->driver->name, num); - if (codec_dai->playback.channels_min) + if (codec_dai->driver->playback.channels_min) playback = 1; - if (codec_dai->capture.channels_min) + if (codec_dai->driver->capture.channels_min) capture = 1; - ret = snd_pcm_new(codec->card, new_name, codec->pcm_devs++, playback, - capture, &pcm); + dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num,new_name); + ret = snd_pcm_new(rtd->card->snd_card, new_name, + num, playback, capture, &pcm); if (ret < 0) { - printk(KERN_ERR "asoc: can't create pcm for codec %s\n", - codec->name); - kfree(rtd); + printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); return ret; } - dai_link->pcm = pcm; + rtd->pcm = pcm; pcm->private_data = rtd; - soc_pcm_ops.mmap = platform->pcm_ops->mmap; - soc_pcm_ops.ioctl = platform->pcm_ops->ioctl; - soc_pcm_ops.copy = platform->pcm_ops->copy; - soc_pcm_ops.silence = platform->pcm_ops->silence; - soc_pcm_ops.ack = platform->pcm_ops->ack; - soc_pcm_ops.page = platform->pcm_ops->page; + soc_pcm_ops.mmap = platform->driver->ops->mmap; + soc_pcm_ops.pointer = platform->driver->ops->pointer; + soc_pcm_ops.ioctl = platform->driver->ops->ioctl; + soc_pcm_ops.copy = platform->driver->ops->copy; + soc_pcm_ops.silence = platform->driver->ops->silence; + soc_pcm_ops.ack = platform->driver->ops->ack; + soc_pcm_ops.page = platform->driver->ops->page; if (playback) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); @@ -1463,16 +1646,15 @@ static int soc_new_pcm(struct snd_soc_device *socdev, if (capture) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); - ret = platform->pcm_new(codec->card, codec_dai, pcm); + ret = platform->driver->pcm_new(rtd->card->snd_card, codec_dai, pcm); if (ret < 0) { printk(KERN_ERR "asoc: platform pcm constructor failed\n"); - kfree(rtd); return ret; } - pcm->private_free = platform->pcm_free; - printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, - cpu_dai->name); + pcm->private_free = platform->driver->pcm_free; + printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->driver->name, + cpu_dai->driver->name); return ret; } @@ -1486,8 +1668,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, */ int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg) { - if (codec->volatile_register) - return codec->volatile_register(reg); + if (codec->driver->volatile_register) + return codec->driver->volatile_register(reg); else return 0; } @@ -1522,7 +1704,6 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, codec->ac97->bus->ops = ops; codec->ac97->num = num; - codec->dev = &codec->ac97->dev; mutex_unlock(&codec->mutex); return 0; } @@ -1537,6 +1718,9 @@ EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); void snd_soc_free_ac97_codec(struct snd_soc_codec *codec) { mutex_lock(&codec->mutex); +#ifdef CONFIG_SND_SOC_AC97_BUS + soc_unregister_ac97_dai_link(codec); +#endif kfree(codec->ac97->bus); kfree(codec->ac97); codec->ac97 = NULL; @@ -1623,95 +1807,6 @@ int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, EXPORT_SYMBOL_GPL(snd_soc_test_bits); /** - * snd_soc_new_pcms - create new sound card and pcms - * @socdev: the SoC audio device - * @idx: ALSA card index - * @xid: card identification - * - * Create a new sound card based upon the codec and interface pcms. - * - * Returns 0 for success, else error. - */ -int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) -{ - struct snd_soc_card *card = socdev->card; - struct snd_soc_codec *codec = card->codec; - int ret, i; - - mutex_lock(&codec->mutex); - - /* register a sound card */ - ret = snd_card_create(idx, xid, codec->owner, 0, &codec->card); - if (ret < 0) { - printk(KERN_ERR "asoc: can't create sound card for codec %s\n", - codec->name); - mutex_unlock(&codec->mutex); - return ret; - } - - codec->socdev = socdev; - codec->card->dev = socdev->dev; - codec->card->private_data = codec; - strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver)); - - /* create the pcms */ - for (i = 0; i < card->num_links; i++) { - ret = soc_new_pcm(socdev, &card->dai_link[i], i); - if (ret < 0) { - printk(KERN_ERR "asoc: can't create pcm %s\n", - card->dai_link[i].stream_name); - mutex_unlock(&codec->mutex); - return ret; - } - /* Check for codec->ac97 to handle the ac97.c fun */ - if (card->dai_link[i].codec_dai->ac97_control && codec->ac97) { - snd_ac97_dev_add_pdata(codec->ac97, - card->dai_link[i].cpu_dai->ac97_pdata); - } - } - - mutex_unlock(&codec->mutex); - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_new_pcms); - -/** - * snd_soc_free_pcms - free sound card and pcms - * @socdev: the SoC audio device - * - * Frees sound card and pcms associated with the socdev. - * Also unregister the codec if it is an AC97 device. - */ -void snd_soc_free_pcms(struct snd_soc_device *socdev) -{ - struct snd_soc_codec *codec = socdev->card->codec; -#ifdef CONFIG_SND_SOC_AC97_BUS - struct snd_soc_dai *codec_dai; - int i; -#endif - - mutex_lock(&codec->mutex); - soc_cleanup_codec_debugfs(codec); -#ifdef CONFIG_SND_SOC_AC97_BUS - for (i = 0; i < codec->num_dai; i++) { - codec_dai = &codec->dai[i]; - if (codec_dai->ac97_control && codec->ac97 && - strcmp(codec->name, "AC97") != 0) { - soc_ac97_dev_unregister(codec); - goto free_card; - } - } -free_card: -#endif - - if (codec->card) - snd_card_free(codec->card); - device_remove_file(socdev->dev, &dev_attr_codec_reg); - mutex_unlock(&codec->mutex); -} -EXPORT_SYMBOL_GPL(snd_soc_free_pcms); - -/** * snd_soc_set_runtime_hwparams - set the runtime hardware parameters * @substream: the pcm substream * @hw: the hardware parameters @@ -1772,7 +1867,7 @@ EXPORT_SYMBOL_GPL(snd_soc_cnew); int snd_soc_add_controls(struct snd_soc_codec *codec, const struct snd_kcontrol_new *controls, int num_controls) { - struct snd_card *card = codec->card; + struct snd_card *card = codec->snd_card; int err, i; for (i = 0; i < num_controls; i++) { @@ -2327,7 +2422,7 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); int snd_soc_limit_volume(struct snd_soc_codec *codec, const char *name, int max) { - struct snd_card *card = codec->card; + struct snd_card *card = codec->snd_card; struct snd_kcontrol *kctl; struct soc_mixer_control *mc; int found = 0; @@ -2459,8 +2554,8 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r_sx); int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { - if (dai->ops && dai->ops->set_sysclk) - return dai->ops->set_sysclk(dai, clk_id, freq, dir); + if (dai->driver && dai->driver->ops->set_sysclk) + return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir); else return -EINVAL; } @@ -2479,8 +2574,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { - if (dai->ops && dai->ops->set_clkdiv) - return dai->ops->set_clkdiv(dai, div_id, div); + if (dai->driver && dai->driver->ops->set_clkdiv) + return dai->driver->ops->set_clkdiv(dai, div_id, div); else return -EINVAL; } @@ -2499,8 +2594,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { - if (dai->ops && dai->ops->set_pll) - return dai->ops->set_pll(dai, pll_id, source, + if (dai->driver && dai->driver->ops->set_pll) + return dai->driver->ops->set_pll(dai, pll_id, source, freq_in, freq_out); else return -EINVAL; @@ -2516,8 +2611,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - if (dai->ops && dai->ops->set_fmt) - return dai->ops->set_fmt(dai, fmt); + if (dai->driver && dai->driver->ops->set_fmt) + return dai->driver->ops->set_fmt(dai, fmt); else return -EINVAL; } @@ -2537,8 +2632,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { - if (dai->ops && dai->ops->set_tdm_slot) - return dai->ops->set_tdm_slot(dai, tx_mask, rx_mask, + if (dai->driver && dai->driver->ops->set_tdm_slot) + return dai->driver->ops->set_tdm_slot(dai, tx_mask, rx_mask, slots, slot_width); else return -EINVAL; @@ -2561,8 +2656,8 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, unsigned int tx_num, unsigned int *tx_slot, unsigned int rx_num, unsigned int *rx_slot) { - if (dai->ops && dai->ops->set_channel_map) - return dai->ops->set_channel_map(dai, tx_num, tx_slot, + if (dai->driver && dai->driver->ops->set_channel_map) + return dai->driver->ops->set_channel_map(dai, tx_num, tx_slot, rx_num, rx_slot); else return -EINVAL; @@ -2578,8 +2673,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map); */ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) { - if (dai->ops && dai->ops->set_tristate) - return dai->ops->set_tristate(dai, tristate); + if (dai->driver && dai->driver->ops->set_tristate) + return dai->driver->ops->set_tristate(dai, tristate); else return -EINVAL; } @@ -2594,8 +2689,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); */ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) { - if (dai->ops && dai->ops->digital_mute) - return dai->ops->digital_mute(dai, mute); + if (dai->driver && dai->driver->ops->digital_mute) + return dai->driver->ops->digital_mute(dai, mute); else return -EINVAL; } @@ -2612,11 +2707,22 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); */ static int snd_soc_register_card(struct snd_soc_card *card) { + int i; + if (!card->name || !card->dev) return -EINVAL; + card->rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime) * card->num_links, + GFP_KERNEL); + if (card->rtd == NULL) + return -ENOMEM; + + for (i = 0; i < card->num_links; i++) + card->rtd[i].dai_link = &card->dai_link[i]; + INIT_LIST_HEAD(&card->list); card->instantiated = 0; + mutex_init(&card->mutex); mutex_lock(&client_mutex); list_add(&card->list, &card_list); @@ -2642,7 +2748,6 @@ static int snd_soc_unregister_card(struct snd_soc_card *card) mutex_lock(&client_mutex); list_del(&card->list); mutex_unlock(&client_mutex); - dev_dbg(card->dev, "Unregistered card '%s'\n", card->name); return 0; @@ -2653,19 +2758,32 @@ static int snd_soc_unregister_card(struct snd_soc_card *card) * * @dai: DAI to register */ -int snd_soc_register_dai(struct snd_soc_dai *dai) +int snd_soc_register_dai(struct device *dev, int id, + struct snd_soc_dai_driver *dai_drv) { - if (!dai->name) - return -EINVAL; + struct snd_soc_dai *dai; + char name[NAME_SIZE]; - /* The device should become mandatory over time */ - if (!dai->dev) - printk(KERN_WARNING "No device for DAI %s\n", dai->name); + dev_dbg(dev, "dai register %s %d\n", dai_drv->name, id); - if (!dai->ops) - dai->ops = &null_dai_ops; + if (!dai_drv->name) + return -EINVAL; - INIT_LIST_HEAD(&dai->list); + dai = kzalloc(sizeof(struct snd_soc_dai), GFP_KERNEL); + if (dai == NULL) + return -ENOMEM; + snprintf(name, NAME_SIZE, "%s.%d", dai_drv->name, id); + dai->name = kstrdup(name, GFP_KERNEL); + if (dai->name == NULL) { + kfree(dai); + return -ENOMEM; + } + + dai->dev = dev; + dai->driver = dai_drv; + if (!dai->driver->ops) + dai->driver->ops = &null_dai_ops; + dai->id = id; mutex_lock(&client_mutex); list_add(&dai->list, &dai_list); @@ -2683,13 +2801,24 @@ EXPORT_SYMBOL_GPL(snd_soc_register_dai); * * @dai: DAI to unregister */ -void snd_soc_unregister_dai(struct snd_soc_dai *dai) +void snd_soc_unregister_dai(struct device *dev, int id) { + struct snd_soc_dai *dai; + + list_for_each_entry(dai, &dai_list, list) { + if (dai->id == id && dev == dai->dev) + goto found; + } + return; + +found: mutex_lock(&client_mutex); list_del(&dai->list); mutex_unlock(&client_mutex); pr_debug("Unregistered DAI '%s'\n", dai->name); + kfree(dai->name); + kfree(dai); } EXPORT_SYMBOL_GPL(snd_soc_unregister_dai); @@ -2699,12 +2828,13 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_dai); * @dai: Array of DAIs to register * @count: Number of DAIs */ -int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count) +int snd_soc_register_dais(struct device *dev, + struct snd_soc_dai_driver *dai_drv, size_t count) { int i, ret; for (i = 0; i < count; i++) { - ret = snd_soc_register_dai(&dai[i]); + ret = snd_soc_register_dai(dev, i, &dai_drv[i]); if (ret != 0) goto err; } @@ -2713,7 +2843,7 @@ int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count) err: for (i--; i >= 0; i--) - snd_soc_unregister_dai(&dai[i]); + snd_soc_unregister_dai(dev, i); return ret; } @@ -2725,12 +2855,12 @@ EXPORT_SYMBOL_GPL(snd_soc_register_dais); * @dai: Array of DAIs to unregister * @count: Number of DAIs */ -void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count) +void snd_soc_unregister_dais(struct device *dev, size_t count) { int i; for (i = 0; i < count; i++) - snd_soc_unregister_dai(&dai[i]); + snd_soc_unregister_dai(dev, i); } EXPORT_SYMBOL_GPL(snd_soc_unregister_dais); @@ -2739,19 +2869,37 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_dais); * * @platform: platform to register */ -int snd_soc_register_platform(struct snd_soc_platform *platform) +int snd_soc_register_platform(struct device *dev, int id, + struct snd_soc_platform_driver *platform_drv) { - if (!platform->name) + struct snd_soc_platform *platform; + char name[NAME_SIZE]; + + dev_dbg(dev, "platform register %s %d\n", platform_drv->name, id); + + if (!platform_drv->name) return -EINVAL; - INIT_LIST_HEAD(&platform->list); + platform = kzalloc(sizeof(struct snd_soc_platform), GFP_KERNEL); + if (platform == NULL) + return -ENOMEM; + snprintf(name, NAME_SIZE, "%s.%d", platform_drv->name, id); + platform->name = kstrdup(name, GFP_KERNEL); + if (platform->name == NULL) { + kfree(platform); + return -ENOMEM; + } + + platform->dev = dev; + platform->driver = platform_drv; + platform->id = id; mutex_lock(&client_mutex); list_add(&platform->list, &platform_list); snd_soc_instantiate_cards(); mutex_unlock(&client_mutex); - pr_debug("Registered platform '%s'\n", platform->name); + pr_debug("Registered platform '%s'\n", platform_drv->name); return 0; } @@ -2762,13 +2910,24 @@ EXPORT_SYMBOL_GPL(snd_soc_register_platform); * * @platform: platform to unregister */ -void snd_soc_unregister_platform(struct snd_soc_platform *platform) +void snd_soc_unregister_platform(struct device *dev, int id) { + struct snd_soc_platform *platform; + + list_for_each_entry(platform, &platform_list, list) { + if (platform->id == id && dev == platform->dev) + goto found; + } + return; + +found: mutex_lock(&client_mutex); list_del(&platform->list); mutex_unlock(&client_mutex); pr_debug("Unregistered platform '%s'\n", platform->name); + kfree(platform->name); + kfree(platform); } EXPORT_SYMBOL_GPL(snd_soc_unregister_platform); @@ -2810,22 +2969,58 @@ static void fixup_codec_formats(struct snd_soc_pcm_stream *stream) * * @codec: codec to register */ -int snd_soc_register_codec(struct snd_soc_codec *codec) +int snd_soc_register_codec(struct device *dev, int id, + struct snd_soc_codec_driver *codec_drv, + struct snd_soc_dai_driver *dai_drv, int num_dai) { - int i; + struct snd_soc_codec *codec; + char name[NAME_SIZE]; + int ret, i; + + dev_dbg(dev, "codec register %s %d\n", codec_drv->name, id); - if (!codec->name) + if (!codec_drv->name) return -EINVAL; - /* The device should become mandatory over time */ - if (!codec->dev) - printk(KERN_WARNING "No device for codec %s\n", codec->name); + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + snprintf(name, NAME_SIZE, "%s.%d", codec_drv->name, id); + codec->name = kstrdup(name, GFP_KERNEL); + if (codec->name == NULL) { + kfree(codec); + return -ENOMEM; + } + + /* allocate CODEC register cache */ + if (codec_drv->reg_cache_size && codec_drv->reg_word_size) { - INIT_LIST_HEAD(&codec->list); + if (codec_drv->reg_cache_default) + codec->reg_cache = kmemdup(codec_drv->reg_cache_default, + codec_drv->reg_cache_size * codec_drv->reg_word_size, GFP_KERNEL); + else + codec->reg_cache = kzalloc(codec_drv->reg_cache_size * + codec_drv->reg_word_size, GFP_KERNEL); - for (i = 0; i < codec->num_dai; i++) { - fixup_codec_formats(&codec->dai[i].playback); - fixup_codec_formats(&codec->dai[i].capture); + if (codec->reg_cache == NULL) { + kfree(codec->name); + kfree(codec); + return -ENOMEM; + } + } + + codec->id = id; + codec->dev = dev; + codec->driver = codec_drv; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->num_dai = num_dai; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + for (i = 0; i < num_dai; i++) { + fixup_codec_formats(&dai_drv[i].playback); + fixup_codec_formats(&dai_drv[i].capture); } mutex_lock(&client_mutex); @@ -2833,9 +3028,29 @@ int snd_soc_register_codec(struct snd_soc_codec *codec) snd_soc_instantiate_cards(); mutex_unlock(&client_mutex); - pr_debug("Registered codec '%s'\n", codec->name); + /* register DAIs */ + for (i = 0; i < num_dai; i++) { + ret = snd_soc_register_dai(dev, i, &dai_drv[i]); + if (ret != 0) + goto error; + } + pr_debug("Registered codec '%s'\n", codec->name); return 0; + +error: + for (i--; i >= 0; i--) + snd_soc_unregister_dai(dev, i); + + mutex_lock(&client_mutex); + list_del(&codec->list); + mutex_unlock(&client_mutex); + + if (codec->reg_cache) + kfree(codec->reg_cache); + kfree(codec->name); + kfree(codec); + return ret; } EXPORT_SYMBOL_GPL(snd_soc_register_codec); @@ -2844,13 +3059,31 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec); * * @codec: codec to unregister */ -void snd_soc_unregister_codec(struct snd_soc_codec *codec) +void snd_soc_unregister_codec(struct device *dev, int id) { + struct snd_soc_codec *codec; + int i; + + list_for_each_entry(codec, &codec_list, list) { + if (codec->id == id && dev == codec->dev) + goto found; + } + return; + +found: + for (i = 0; i < codec->num_dai; i++) + snd_soc_unregister_dai(dev, id); + mutex_lock(&client_mutex); list_del(&codec->list); mutex_unlock(&client_mutex); pr_debug("Unregistered codec '%s'\n", codec->name); + + if (codec->reg_cache) + kfree(codec->reg_cache); + kfree(codec->name); + kfree(codec); } EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); -- 1.7.0.4

2 2

[alsa-devel] Introduction from Robert.
by robert 01 Jul '10

01 Jul '10

Dear All, My name is Robert. I have chosen ALSA to develop my own audio application for Linux. How are you? Cheers, Robert

1 0

[alsa-devel] [PATCH 1/2] ASoC: uda134x: add DATA011 register found in codecs family
by Vladimir Zapolskiy 30 Jun '10

30 Jun '10

In UDA1340, UDA1344 and UDA1345 codecs there is one more functional register in part of DATA0 tranfser. For UDA1341 this register coincides with EA register. Signed-off-by: Vladimir Zapolskiy <vzapolskiy(a)gmail.com> --- sound/soc/codecs/uda134x.c | 3 ++- sound/soc/codecs/uda134x.h | 5 +++-- 2 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index b3039b1..e770029 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -57,7 +57,7 @@ static const char uda134x_reg[UDA134X_REGS_NUM] = { /* Extended address registers */ 0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, /* Status, data regs */ - 0x00, 0x83, 0x00, 0x40, 0x80, 0x00, + 0x00, 0x83, 0x00, 0x40, 0x80, 0xC0, 0x00, }; /* @@ -116,6 +116,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg, case UDA134X_DATA000: case UDA134X_DATA001: case UDA134X_DATA010: + case UDA134X_DATA011: addr = UDA134X_DATA0_ADDR; break; case UDA134X_DATA1: diff --git a/sound/soc/codecs/uda134x.h b/sound/soc/codecs/uda134x.h index 94f4404..205f03b 100644 --- a/sound/soc/codecs/uda134x.h +++ b/sound/soc/codecs/uda134x.h @@ -23,9 +23,10 @@ #define UDA134X_DATA000 10 #define UDA134X_DATA001 11 #define UDA134X_DATA010 12 -#define UDA134X_DATA1 13 +#define UDA134X_DATA011 13 +#define UDA134X_DATA1 14 -#define UDA134X_REGS_NUM 14 +#define UDA134X_REGS_NUM 15 #define STATUS0_DAIFMT_MASK (~(7<<1)) #define STATUS0_SYSCLK_MASK (~(3<<4)) -- 1.7.0.4

2 10

[alsa-devel] [PATCH] ASoC: DaVinci: Added support for cpu clocking I2S
by Raffaele Recalcati 30 Jun '10

30 Jun '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Added audio playback support with [frame sync master - clock master] mode and with [frame sync master - clock slave]. 1. SND_SOC_DAIFMT_CBS_CFS (the cpu generates clock and frame sync) I need to clarify how the McBSP is clocked internally. Reading sprufi3a-1.pdf (TMS320DM36x DMSoC Multichannel Buffered Serial Port (McBSP) Interface) at "2.5.3 Data Clock Generation" paragraph this info not appears really clear if we are in the case of "McBSP internal input clock". I hope some Ti developers can help me. There is a FIXME message in the patch. i2s_fast_clock switch can be used to have better approximate or symmetric waveforms. clk_input_pin board info can be used to select it depending on hw connections 2. SND_SOC_DAIFMT_CBS_CFM (the cpu get clock from external pin and generates frame sync) Clock slave can be important when the external codec need system clock and bit clock synchronized (tested with uda1345). 3. We haven't changed the evmdm365 support (due also to CPLD that doesn't help to understand) We don't know in this mode if audio stereo works on evmdm365. Probably it does. This patch has been developed against the http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm, but using uda1345 as external audio codec). Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- arch/arm/mach-davinci/include/mach/asp.h | 16 ++++ sound/soc/davinci/davinci-i2s.c | 126 ++++++++++++++++++++++++++---- 2 files changed, 128 insertions(+), 14 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 834725f..f9b6da2 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -63,6 +63,16 @@ struct snd_platform_data { unsigned sram_size_playback; unsigned sram_size_capture; + /* + * This define works when both clock and FS are output for the cpu + * and makes clock very fast (FS is not simmetrical, but sampling + * frequency is better approximated + */ + int i2s_fast_clock; + + /* To be used when cpu gets clock from extenal pin */ + int clk_input_pin; + /* McASP specific fields */ int tdm_slots; u8 op_mode; @@ -78,6 +88,12 @@ enum { MCASP_VERSION_2, /* DA8xx/OMAPL1x */ }; +enum { + MCBSP_CLKR = 0, /* DM365 */ + MCBSP_CLKS, +}; + + #define INACTIVE_MODE 0 #define TX_MODE 1 #define RX_MODE 2 diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index adadcd3..786ade6 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -68,16 +68,21 @@ #define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_RCR_RFIG (1 << 18) #define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_RCR_RFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_RCR_RPHASE (1 << 31) #define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5) #define DAVINCI_MCBSP_XCR_XFRLEN1(v) ((v) << 8) #define DAVINCI_MCBSP_XCR_XDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_XCR_XFIG (1 << 18) #define DAVINCI_MCBSP_XCR_XWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_XCR_XFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_XCR_XPHASE (1 << 31) #define DAVINCI_MCBSP_SRGR_FWID(v) ((v) << 8) #define DAVINCI_MCBSP_SRGR_FPER(v) ((v) << 16) #define DAVINCI_MCBSP_SRGR_FSGM (1 << 28) +#define DAVINCI_MCBSP_SRGR_CLKSM (1 << 29) #define DAVINCI_MCBSP_PCR_CLKRP (1 << 0) #define DAVINCI_MCBSP_PCR_CLKXP (1 << 1) @@ -144,6 +149,11 @@ struct davinci_mcbsp_dev { * won't end up being swapped because of the underrun. */ unsigned enable_channel_combine:1; + + int i2s_fast_clock; + unsigned int fmt; + int clk_div; + int clk_input_pin; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -254,10 +264,12 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, struct davinci_mcbsp_dev *dev = cpu_dai->private_data; unsigned int pcr; unsigned int srgr; + /* Attention srgr is updated by hw_params! */ srgr = DAVINCI_MCBSP_SRGR_FSGM | DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) | DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1); + dev->fmt = fmt; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: @@ -268,11 +280,18 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, DAVINCI_MCBSP_PCR_CLKRM; break; case SND_SOC_DAIFMT_CBM_CFS: - /* McBSP CLKR pin is the input for the Sample Rate Generator. - * McBSP FSR and FSX are driven by the Sample Rate Generator. */ - pcr = DAVINCI_MCBSP_PCR_SCLKME | - DAVINCI_MCBSP_PCR_FSXM | - DAVINCI_MCBSP_PCR_FSRM; + pcr = DAVINCI_MCBSP_PCR_FSRM | DAVINCI_MCBSP_PCR_FSXM; + if (dev->clk_input_pin == MCBSP_CLKS) + pcr |= DAVINCI_MCBSP_PCR_CLKXM | + DAVINCI_MCBSP_PCR_CLKRM; + else + /* + * McBSP CLKR pin is the input for the Sample Rate + * Generator. + * McBSP FSR and FSX are driven by the Sample Rate + * Generator. + */ + pcr |= DAVINCI_MCBSP_PCR_SCLKME; break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is master */ @@ -372,6 +391,16 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return 0; } +static int davinci_i2s_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + int srgr; + + dev->clk_div = div; + return 0; +} + static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -380,11 +409,12 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct davinci_pcm_dma_params *dma_params = &dev->dma_params[substream->stream]; struct snd_interval *i = NULL; - int mcbsp_word_length; - unsigned int rcr, xcr, srgr; + int mcbsp_word_length, master; + unsigned int rcr, xcr, srgr, clk_div, freq, framesize; u32 spcr; snd_pcm_format_t fmt; unsigned element_cnt = 1; + struct clk *clk; /* general line settings */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); @@ -396,12 +426,56 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); } - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); - srgr = DAVINCI_MCBSP_SRGR_FSGM; - srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + fmt = params_format(params); + mcbsp_word_length = asp_word_length[fmt]; - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); - srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + if (master == SND_SOC_DAIFMT_CBS_CFS) { + clk = clk_get(NULL, "pll1_sysclk6"); + if (clk) + freq = clk_get_rate(clk); + freq = 122000000; /* FIXME ask to Texas */ + srgr = DAVINCI_MCBSP_SRGR_FSGM | + DAVINCI_MCBSP_SRGR_CLKSM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * + 8 - 1); + if (dev->i2s_fast_clock) { + clk_div = 256; + do { + framesize = (freq / (--clk_div)) / + params->rate_num * + params->rate_den; + } while (((framesize < 33) || (framesize > 4095)) && + (clk_div)); + clk_div--; + srgr |= DAVINCI_MCBSP_SRGR_FPER(framesize - 1); + } else { + /* symmetric waveforms */ + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * + 16 - 1); + } + clk_div &= 0xFF; + srgr |= clk_div; + } else if (master == SND_SOC_DAIFMT_CBM_CFS) { + /* Clock given on CLKS */ + srgr = DAVINCI_MCBSP_SRGR_FSGM; + clk_div = dev->clk_div - 1; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * 16 - 1); + clk_div &= 0xFF; + srgr |= clk_div; + } else { + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); + srgr = DAVINCI_MCBSP_SRGR_FSGM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + pr_debug("%s - %d FWID set: re-read srgr = %X\n", + __func__, __LINE__, snd_interval_value(i) - 1); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); + srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + } davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr); rcr = DAVINCI_MCBSP_RCR_RFIG; @@ -426,12 +500,29 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, element_cnt = 1; fmt = double_fmt[fmt]; } + if (master == SND_SOC_DAIFMT_CBS_CFS || + master == SND_SOC_DAIFMT_CBS_CFM) { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(0); + rcr |= DAVINCI_MCBSP_RCR_RPHASE; + xcr |= DAVINCI_MCBSP_XCR_XPHASE; + } else { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(element_cnt - 1); + } } dma_params->acnt = dma_params->data_type = data_type[fmt]; dma_params->fifo_level = 0; mcbsp_word_length = asp_word_length[fmt]; - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + + if (master == SND_SOC_DAIFMT_CBS_CFS || + master == SND_SOC_DAIFMT_CBS_CFM) { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(0); + } else { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + } rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); @@ -442,6 +533,10 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr); else davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr); + + pr_debug("%s - %d srgr=%X\n", __func__, __LINE__, srgr); + pr_debug("%s - %d xcr=%X\n", __func__, __LINE__, xcr); + pr_debug("%s - %d rcr=%X\n", __func__, __LINE__, rcr); return 0; } @@ -500,6 +595,7 @@ static struct snd_soc_dai_ops davinci_i2s_dai_ops = { .trigger = davinci_i2s_trigger, .hw_params = davinci_i2s_hw_params, .set_fmt = davinci_i2s_set_dai_fmt, + .set_clkdiv = davinci_i2s_dai_set_clkdiv, }; @@ -552,6 +648,8 @@ static int davinci_i2s_probe(struct platform_device *pdev) pdata->sram_size_playback; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = pdata->sram_size_capture; + dev->i2s_fast_clock = pdata->i2s_fast_clock; + dev->clk_input_pin = pdata->clk_input_pin; } dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { -- 1.7.0.4

3 6

[alsa-devel] [PATCH v2] ASoC: ak4642: Add Digital Playback Volume control
by Kuninori Morimoto 30 Jun '10

30 Jun '10

This patch add Digital Playback Volume control. And it moves default digital playback volume settings from ak4642_dai_startup to ak4642_init Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 o "Playback Volume" => "Digital Playback Volume" o Add explain on commit log sound/soc/codecs/ak4642.c | 28 +++++++++++++++++++++++++--- 1 files changed, 25 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 7528a54..7b96311 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -36,6 +36,7 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "ak4642.h" @@ -111,6 +112,23 @@ struct snd_soc_codec_device soc_codec_dev_ak4642; +/* + * Playback Volume (table 39) + * + * max : 0x00 : +12.0 dB + * ( 0.5 dB step ) + * min : 0xFE : -115.0 dB + * mute: 0xFF + */ +static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1); + +static const struct snd_kcontrol_new ak4642_snd_controls[] = { + + SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC, + 0, 0xFF, 1, out_tlv), +}; + + /* codec private data */ struct ak4642_priv { struct snd_soc_codec codec; @@ -204,7 +222,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * * PLL, Master Mode * Audio I/F Format :MSB justified (ADC & DAC) - * Digital Volume: -8dB * Bass Boost Level : Middle * * This operation came from example code of @@ -214,8 +231,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, ak4642_write(codec, 0x0e, 0x19); ak4642_write(codec, 0x09, 0x91); ak4642_write(codec, 0x0c, 0x91); - ak4642_write(codec, 0x0a, 0x28); - ak4642_write(codec, 0x0d, 0x28); ak4642_write(codec, 0x00, 0x64); snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN); @@ -463,6 +478,10 @@ static int ak4642_init(struct ak4642_priv *ak4642) goto reg_cache_err; } + /* default volume */ + ak4642_write(codec, L_DVC, 0x28); + ak4642_write(codec, R_DVC, 0x28); + return ret; reg_cache_err: @@ -548,6 +567,9 @@ static int ak4642_probe(struct platform_device *pdev) goto pcm_err; } + snd_soc_add_controls(ak4642_codec, ak4642_snd_controls, + ARRAY_SIZE(ak4642_snd_controls)); + dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); return ret; -- 1.7.0.4

2 1

[alsa-devel] alsamixer-qt4 release
by Sebastian H. 30 Jun '10

30 Jun '10

Hello list Here is (yet another) mixer application for ALSA: alsamixer-qt4 It took a bit longer that expected but today I've erased the last items from the TODO list for the first release version 0.1.0. All files are hosted at sourceforge http://sourceforge.net/projects/alsamixer-qt4/files/ And here's a small web page http://xwmw.org/alsamixer-qt4 At the moment it is a volume only mixer and the dB value is only displayed but the scaling sliders are worth a look (The style is homebrewn ;). So far. Regards, Sebastian H.

2 2

[alsa-devel] snddevices issue
by Adish Kuvelker 29 Jun '10

29 Jun '10

Hello, I find that the device nodes created in the /dev/snd for Control, Timer and pcmC0D0 have different minor numbers from the class devices created by the ALSA core. I just did kernel prints to capture the minor numbers of audio Control, Timer and pcmC0D0p creatd by ALSA core and found that they are different. My ALSA core version is ALSA 1.0.13 and the snddevices file is also from the ALSA 1.0.13. How do I do a workabout for this? Regards; Aadish

2 2

[alsa-devel] alsa-utils configuration query
by Adish Kuvelker 29 Jun '10

29 Jun '10

Hello, I am cross-compiling alsa-utils for my ARM target. I have successfully configured and installed the alsa-lib (alsa-lib-1.0.16). Now to configure the alsa-utils(alsa-utils-1.0.16) i pass the following command LD=arm-linux-ld ./configure --with-alsa-prefix=/usr/lib --build=i686-linux --disable-nls The libasound.la & libasound.a are present in the /usr/lib but the above command gives an error checking for snd_ctl_open in -lasound... no configure: error: No linkable libasound was found Can anyone give me a pointer of how to tackle this? Regards; Aadish

1 0

Re: [alsa-devel] [PATCH] - remove "front" , "surround" , "rear" , "surround40" , "surround51" from output of arecord -L
by Raymond Yau 29 Jun '10

29 Jun '10

2010/6/29 Jerone Young <jerone.young(a)canonical.com> > Hey Raymod, > A bunch of things wrong with this. > > 1) You need to post the whole diff patch .. not must the lines .. > the > diff patch will contain the file this is patching. > > 2) Needs a Signed-off-by: ME <me(a)somewhere.com> line. > > Jerone > > On Tue, 2010-06-29 at 12:05 +0800, Raymond Yau wrote: > > Since arecord -L will list "front", ..... devices which are not suitable > > for recording (e.g. HDA has softvol plugin in front device , emu10k1 and > > other drivers has hook ctl and lock ) > > > > > > > > > > while (*n != NULL) { > > name = snd_device_name_get_hint(*n, "NAME"); > > descr = snd_device_name_get_hint(*n, "DESC"); > > io = snd_device_name_get_hint(*n, "IOID"); > > + if ( stream == SND_PCM_STREAM_CAPTURE ) > > + if (( strstr(descr,"speaker") > 0 ) || ( > > strstr(descr,"Output") > 0 )) > > + goto __end; > > if (io != NULL && strcmp(io, filter) != 0) > > goto __end; > > printf("%s\n", name); > > if ((descr1 = descr) != NULL) { > I am not sure the fix should be at arecord or namehint since those "Output" and "speaker" may be translated to different language and use "front" device for capturing is not recommeded since there are many drivers have hook ctl and lock (e.g. "Channel Mode" , "Line-In As Surround", Surround Down Mix) in front or surroundxx device and emu10k1 have ctl of pcm_route/pcm_volume ATIIXP.conf: lock true ATIIXP-SPDMA.conf: lock true Audigy2.conf: lock true Audigy.conf: lock true Aureon51.conf: lock true Aureon71.conf: lock true CA0106.conf: lock true CMI8338.conf: lock true CMI8738-MC6.conf: lock true CMI8738-MC8.conf: lock true CMI8788.conf: lock true CS46xx.conf: lock true EMU10K1.conf: lock true EMU10K1X.conf: lock true ENS1370.conf: lock true ENS1371.conf: lock true HDA-Intel.conf: lock true ICE1712.conf: lock true ICE1724.conf: lock true ICH4.conf: lock true ICH.conf: lock true NFORCE.conf: lock true PS3.conf: lock true RME9652.conf: lock true SB-XFi.conf: lock true SI7018.conf: lock true TRID4DWAVENX.conf: lock true VIA686A.conf: lock true VIA8233A.conf: lock true VIA8237.conf: lock true VX222.conf: lock true VXPocket.conf: lock true YMF744.conf: lock true

1 0

[alsa-devel] Card information in /proc or /sys
by Florian Faber 29 Jun '10

29 Jun '10

Hi, I want to expose more information about the sound card's ports (names, connectors etc.) to userspace in a standardized manner. Does ALSA have predefined locations for this in /sys, or would /proc (snd_card_proc_new) still be the way to go? Flo -- Machines can do the work, so people have time to think. public key DA43FEF4 x-hkp://wwwkeys.eu.pgp.net

2 1

[alsa-devel] [PATCH] - remove "front" , "surround" , "rear" , "surround40" , "surround51" from output of arecord -L
by Raymond Yau 29 Jun '10

29 Jun '10

Since arecord -L will list "front", ..... devices which are not suitable for recording (e.g. HDA has softvol plugin in front device , emu10k1 and other drivers has hook ctl and lock ) while (*n != NULL) { name = snd_device_name_get_hint(*n, "NAME"); descr = snd_device_name_get_hint(*n, "DESC"); io = snd_device_name_get_hint(*n, "IOID"); + if ( stream == SND_PCM_STREAM_CAPTURE ) + if (( strstr(descr,"speaker") > 0 ) || ( strstr(descr,"Output") > 0 )) + goto __end; if (io != NULL && strcmp(io, filter) != 0) goto __end; printf("%s\n", name); if ((descr1 = descr) != NULL) {

1 0

Re: [alsa-devel] [PATCH 1/4] dump_audiostreaming_interface(): fix ac-3 decoder
by Daniel Mack 28 Jun '10

28 Jun '10

Hi Greg, On Tue, Jun 01, 2010 at 10:06:58AM -0700, Greg KH wrote: > On Tue, Jun 01, 2010 at 06:52:12PM +0200, Daniel Mack wrote: > > Greg, > > > > I believe you're maintaining usb-utils now, right? At least it was > > you posting an announcement for a new release the other day. > > Yes I am. > > > Let me know if I should do anything else to get these patches merged :) > > Let me wake up and get a chance to get around to these patches? :) > > I'll get to them in a day or so, they look great to me. Just a reminder - could you have a look at them already? For easier access, the six patches I sent can now also be pulled from git://github.com/zonque/usbutils.git FWIW, I got positive feedback from people developing USB audio class 2 devices - my patches seem to display the right things for them :) Thanks, Daniel

2 3

[alsa-devel] newbie, arecord
by javi perez 28 Jun '10

28 Jun '10

I am trying to do a speech recognition system. In the aplay file I would like to conver the unsigned char buffer to a float buffer to be able to make comparisons the way I would like. The floats that I am getting after the conversion are these. How should I do the conversion from unsigned char to float array? thanks in advance. 819 -464604244644772930012428885483847680.000000 820 -542490915911345743841838949652758528.000000 821 -495758857691688295559156595670646784.000000 822 -480181610589352498484045935735209984.000000 823 -441239146465853748477054432634208256.000000 824 -376333296573780517983308202832297984.000000 825 -503547144523165508473277153076576256.000000 826 -485374263974618640302095603012206592.000000 827 -495758501164956981369637424722870272.000000 828 -433450265423157678580401923648651264.000000 829 -415277147190123267616207592952430592.000000 830 -425662216276168008459294146874572800.000000 831 -477585818686816398859299858518376448.000000 832 -394508395510877785555942372127277056.000000 833 -472393086073387742776912597697429504.000000 834 -482778551300245054941687119339323392.000000 835 -584029726534518193323939684778246144.000000 836 -511336897075649235277643191045586944.000000 837 -558067806486950226727430438640418816.000000 838 -513933085118997906224077235982172160.000000 839 -302397753578834695235772573852631040.000000 float* float_array; while (rest > 0 && recycle_capture_file == 0) { size_t c = (rest <= (off64_t)chunk_bytes) ? (size_t)rest : chunk_bytes; size_t f = c * 8 / bits_per_frame; if (pcm_read(audiobuf, f) != f) break; float_array=(float*)malloc(f); memcpy(float_array,audiobuf,f); int z=0; for(z=0; z<f/sizeof(float);z++){ printf("%d %f\n",z,float_array[z]); } /*if (write(fd, audiobuf, c) != c) { perror(name); prg_exit(EXIT_FAILURE); }*/ count -= c; rest -= c; fdcount += c; }

2 1

Re: [alsa-devel] [PATCH RFC 1/3] ASoC: uda1380: use callbacks instead of gpiolib
by Mark Brown 28 Jun '10

28 Jun '10

On Mon, Jun 28, 2010 at 04:20:43PM +0200, pHilipp Zabel wrote: > The reset GPIO is connected directly to the UDA1380 RESET pin. The > power source is external. OK, the power source is definitely a regulator then.

1 0

[alsa-devel] [PATCH RFC 0/3] asoc: uda1380 cleanup
by Vasily Khoruzhick 28 Jun '10

28 Jun '10

This patch series cleanups uda1380 codec driver a bit: - it introduces callbacks instead of direct gpiolib usage, as some machines requires more actions than just plain gpio toggle to enable/disable/reset codec; - it makes driver more powersave-friendly and fixes suspend/resume

2 24

[alsa-devel] [PATCH] ASoC: ak4642: Add Playback Volume control
by Kuninori Morimoto 28 Jun '10

28 Jun '10

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This patch works for me. But I'm not sure it is correct settings. Please check it carefully sound/soc/codecs/ak4642.c | 28 +++++++++++++++++++++++++--- 1 files changed, 25 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 9a6157f..ce4e6ec 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -28,6 +28,7 @@ #include <linux/slab.h> #include <sound/soc-dapm.h> #include <sound/initval.h> +#include <sound/tlv.h> #include "ak4642.h" @@ -103,6 +104,23 @@ struct snd_soc_codec_device soc_codec_dev_ak4642; +/* + * Playback Volume (table 39) + * + * max : 0x00 : +12.0 dB + * ( 0.5 dB step ) + * min : 0xFE : -115.0 dB + * mute: 0xFF + */ +static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1); + +static const struct snd_kcontrol_new ak4642_snd_controls[] = { + + SOC_DOUBLE_R_TLV("Playback Volume", L_DVC, R_DVC, + 0, 0xFF, 1, out_tlv), +}; + + /* codec private data */ struct ak4642_priv { struct snd_soc_codec codec; @@ -190,7 +208,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * * PLL, Master Mode * Audio I/F Format :MSB justified (ADC & DAC) - * Digital Volume: -8dB * Bass Boost Level : Middle * * This operation came from example code of @@ -200,8 +217,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, ak4642_write(codec, 0x0e, 0x19); ak4642_write(codec, 0x09, 0x91); ak4642_write(codec, 0x0c, 0x91); - ak4642_write(codec, 0x0a, 0x28); - ak4642_write(codec, 0x0d, 0x28); ak4642_write(codec, 0x00, 0x64); snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN); @@ -446,6 +461,10 @@ static int ak4642_init(struct ak4642_priv *ak4642) return ret; } + /* default volume */ + ak4642_write(codec, L_DVC, 0x28); + ak4642_write(codec, R_DVC, 0x28); + return ret; } @@ -524,6 +543,9 @@ static int ak4642_probe(struct platform_device *pdev) goto pcm_err; } + snd_soc_add_controls(ak4642_codec, ak4642_snd_controls, + ARRAY_SIZE(ak4642_snd_controls)); + dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); return ret; -- 1.7.0.4

3 5

[alsa-devel] [PATCH] sis7019: increase reset delays
by David Dillow 28 Jun '10

28 Jun '10

A few boards using this controller are reported to need a little extra time during their reset cycle. Reported-by: Michael Goeke <michael.goeke(a)icachip.de> Signed-off-by: Dave Dillow <dave(a)thedillows.org> --- Wow, a horrible maintainer am I. Working on the virtual timing period reminded me of of another timing issue from two years back that I forgot to push upstream. This applies on top of the vperiod fixes, but should not conflict if the ordering is changed. diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 614ff6e..1b8f674 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1049,7 +1049,7 @@ static int sis_chip_free(struct sis7019 *sis) /* Reset the chip, and disable all interrputs. */ outl(SIS_GCR_SOFTWARE_RESET, sis->ioport + SIS_GCR); - udelay(10); + udelay(25); outl(0, sis->ioport + SIS_GCR); outl(0, sis->ioport + SIS_GIER); @@ -1085,7 +1085,7 @@ static int sis_chip_init(struct sis7019 *sis) /* Reset the audio controller */ outl(SIS_GCR_SOFTWARE_RESET, io + SIS_GCR); - udelay(10); + udelay(25); outl(0, io + SIS_GCR); /* Get the AC-link semaphore, and reset the codecs @@ -1098,7 +1098,7 @@ static int sis_chip_init(struct sis7019 *sis) return -EIO; outl(SIS_AC97_CMD_CODEC_COLD_RESET, io + SIS_AC97_CMD); - udelay(10); + udelay(250); count = 0xffff; while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count)

2 1

[alsa-devel] [PATCH] ASoC: ak4642: Add ak4642_fill_cache function
by Kuninori Morimoto 28 Jun '10

28 Jun '10

ak4642 driver is used for ak4642/ak4643. So, reading real register value is better than using constant value. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4642.c | 50 +++++++++++++++----------------------------- 1 files changed, 17 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 8d56811..9a6157f 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -106,25 +106,19 @@ struct snd_soc_codec_device soc_codec_dev_ak4642; /* codec private data */ struct ak4642_priv { struct snd_soc_codec codec; + u8 reg_cache[AK4642_CACHEREGNUM]; }; static struct snd_soc_codec *ak4642_codec; -/* - * ak4642 register cache - */ -static const u16 ak4642_reg[AK4642_CACHEREGNUM] = { - 0x0000, 0x0000, 0x0001, 0x0000, - 0x0002, 0x0000, 0x0000, 0x0000, - 0x00e1, 0x00e1, 0x0018, 0x0000, - 0x00e1, 0x0018, 0x0011, 0x0008, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, -}; +static void ak4642_fill_cache(struct snd_soc_codec *codec) +{ + int i; + u8 *cache = codec->reg_cache; + + for (i = 0; i < AK4642_CACHEREGNUM; i++) + i2c_master_recv(codec->control_data, cache + i, 1); +} /* * read ak4642 register cache @@ -132,7 +126,7 @@ static const u16 ak4642_reg[AK4642_CACHEREGNUM] = { static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { - u16 *cache = codec->reg_cache; + u8 *cache = codec->reg_cache; if (reg >= AK4642_CACHEREGNUM) return -1; return cache[reg]; @@ -144,7 +138,7 @@ static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec, static inline void ak4642_write_reg_cache(struct snd_soc_codec *codec, u16 reg, unsigned int value) { - u16 *cache = codec->reg_cache; + u8 *cache = codec->reg_cache; if (reg >= AK4642_CACHEREGNUM) return; @@ -175,7 +169,7 @@ static int ak4642_write(struct snd_soc_codec *codec, unsigned int reg, static int ak4642_sync(struct snd_soc_codec *codec) { - u16 *cache = codec->reg_cache; + u8 *cache = codec->reg_cache; int i, r = 0; for (i = 0; i < AK4642_CACHEREGNUM; i++) @@ -432,12 +426,9 @@ static int ak4642_init(struct ak4642_priv *ak4642) codec->dai = &ak4642_dai; codec->num_dai = 1; codec->hw_write = (hw_write_t)i2c_master_send; - codec->reg_cache_size = ARRAY_SIZE(ak4642_reg); - codec->reg_cache = kmemdup(ak4642_reg, - sizeof(ak4642_reg), GFP_KERNEL); - - if (!codec->reg_cache) - return -ENOMEM; + codec->reg_cache_size = AK4642_CACHEREGNUM; + codec->reg_cache = ak4642->reg_cache; + ak4642_fill_cache(codec); ak4642_dai.dev = codec->dev; ak4642_codec = codec; @@ -445,23 +436,17 @@ static int ak4642_init(struct ak4642_priv *ak4642) ret = snd_soc_register_codec(codec); if (ret) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - goto reg_cache_err; + return ret; } ret = snd_soc_register_dai(&ak4642_dai); if (ret) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); snd_soc_unregister_codec(codec); - goto reg_cache_err; + return ret; } return ret; - -reg_cache_err: - kfree(codec->reg_cache); - codec->reg_cache = NULL; - - return ret; } #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) @@ -495,7 +480,6 @@ static int ak4642_i2c_remove(struct i2c_client *client) snd_soc_unregister_dai(&ak4642_dai); snd_soc_unregister_codec(&ak4642->codec); - kfree(ak4642->codec.reg_cache); kfree(ak4642); ak4642_codec = NULL; -- 1.7.0.4

3 3

[alsa-devel] ALSA SoC device tree bindings
by Jeremy Kerr 27 Jun '10

27 Jun '10

Hi all, From some discussions at the previous Ubuntu Developer Summit, I've put up a page on the devicetree.org wiki with the ultimate goal of describing device tree bindings for ALSA System on Chip devices. At this stage, we're just looking at how to structure device in the device tree. The bindings for specific devices will come as required (ie, as we add device tree support for them). So - if you're interested in the ALSA and/or device tree areas, please take a look at the notes from UDS: http://devicetree.org/SoC_Audio - if you have any comments or suggestions, please let me know, or add to the page (or the discussion for the page). I intend for this document to become a more formal specification, but we still have a way to go before that happens. If you'd like an introduction to device trees, Grant has a good walkthrough- style document at: http://devicetree.org/Device_Tree_Usage Cheers, Jeremy

4 5

[alsa-devel] Modifying the AD193X driver and integrating with the Blackfin platform
by Adam Rosenberg 27 Jun '10

27 Jun '10

I see that Liam just modified the code (bf5xx-ad193x.c) that I was looking at modifying to add support for the CS42448 CODEC. I would like to understand the architecture of the ASoC and Blackfin TDM PCM driver that it uses because I need to enable the secondary data lines to support multiple chips. Where can I find the documentation for this code? If I want to integrate this code into the uClinux-dist-2009R1.1-RC4 distribution, is it simply a matter of copying the files in "[linux-2.6-asoc.git]/sound/" to the linux-2.6/sound directory for uClinux? Or is it better if I git clone linux-2.6-asoc and move my custom board development files there? Is there a driver installation guide somewhere that I can read about this? I am concerned that the official Blackfin distribution (http://blackfin.uclinux.org/gf/download/frsrelease/473/7285/uClinux-dist-20…) will not be compatible with the ASoC distribution (git://opensource.wolfsonmicro.com/linux-2.6-asoc). Can you tell me if there are any differences between these? To sum up: 1. Where can I find the ASoC documentation? 2. Is there an installation guide to use ASoC with an existing Kernel? 3. What are the differences between the wolfsonmicro.com linux-2.6-asoc and blackfin.uclinux.org uClinux-dist-2009R1.1-RC4? Thank you, Adam Adam Rosenberg Software Engineer Alcorn McBride Inc. 3300 South Hiawassee Building 105 Orlando, FL 32835 (407) 296 - 5800 ext. 5490

2 1

[alsa-devel] Detecting if a pcm device is a digital or analog one
by Jean-Yves Avenard 25 Jun '10

25 Jun '10

Hi there... I was wondering if there was any way to accurately distinguish between audio device, if it has an analog or a digital output. I would prefer not parsing the name (like analog71 is obviously an analog device, while "hdmi" is digital) I see that on some devices the IEC958 bit is set, but this doesn't seem reliable (on another PC, the IEC958 bit isn't set). Any suggestions would be greatly appreciated... Thank you Jean-Yves

3 2

[alsa-devel] [PATCH] ASoC: SAMSUNG: Change s3c24xx directory name to samsung
by 최찬우 25 Jun '10

25 Jun '10

This patch changes s3c24xx directory name to samsung. (sound/soc/s3c24xx -> sound/soc/samsung) Currently, the source files related to samsung SoCs(s3c24xx & s3c64xx) are included in sound/soc/s3c24xx. It would be better to change s3c24xx directory name to samsung, because s3c24xx directory will support many board based on samsung SoCs(s5p, etc). Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/Kconfig | 2 +- sound/soc/Makefile | 2 +- sound/soc/s3c24xx/Kconfig | 127 ---- sound/soc/s3c24xx/Makefile | 43 -- sound/soc/s3c24xx/jive_wm8750.c | 196 ------ sound/soc/s3c24xx/lm4857.h | 32 - sound/soc/s3c24xx/ln2440sbc_alc650.c | 85 --- sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 506 --------------- sound/soc/s3c24xx/neo1973_wm8753.c | 707 --------------------- sound/soc/s3c24xx/regs-i2s-v2.h | 115 ---- sound/soc/s3c24xx/s3c-ac97.c | 521 ---------------- sound/soc/s3c24xx/s3c-ac97.h | 23 - sound/soc/s3c24xx/s3c-dma.c | 483 --------------- sound/soc/s3c24xx/s3c-dma.h | 31 - sound/soc/s3c24xx/s3c-i2s-v2.c | 776 ------------------------ sound/soc/s3c24xx/s3c-i2s-v2.h | 103 ---- sound/soc/s3c24xx/s3c-pcm.c | 554 ----------------- sound/soc/s3c24xx/s3c-pcm.h | 123 ---- sound/soc/s3c24xx/s3c2412-i2s.c | 190 ------ sound/soc/s3c24xx/s3c2412-i2s.h | 29 - sound/soc/s3c24xx/s3c24xx-i2s.c | 503 --------------- sound/soc/s3c24xx/s3c24xx-i2s.h | 37 -- sound/soc/s3c24xx/s3c24xx_simtec.c | 394 ------------ sound/soc/s3c24xx/s3c24xx_simtec.h | 22 - sound/soc/s3c24xx/s3c24xx_simtec_hermes.c | 153 ----- sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c | 137 ----- sound/soc/s3c24xx/s3c24xx_uda134x.c | 373 ------------ sound/soc/s3c24xx/s3c64xx-i2s-v4.c | 209 ------- sound/soc/s3c24xx/s3c64xx-i2s.c | 190 ------ sound/soc/s3c24xx/s3c64xx-i2s.h | 42 -- sound/soc/s3c24xx/smdk2443_wm9710.c | 81 --- sound/soc/s3c24xx/smdk64xx_wm8580.c | 266 -------- sound/soc/s3c24xx/smdk_wm9713.c | 94 --- sound/soc/samsung/Kconfig | 127 ++++ sound/soc/samsung/Makefile | 43 ++ sound/soc/samsung/jive_wm8750.c | 196 ++++++ sound/soc/samsung/lm4857.h | 32 + sound/soc/samsung/ln2440sbc_alc650.c | 85 +++ sound/soc/samsung/neo1973_gta02_wm8753.c | 506 +++++++++++++++ sound/soc/samsung/neo1973_wm8753.c | 707 +++++++++++++++++++++ sound/soc/samsung/regs-i2s-v2.h | 115 ++++ sound/soc/samsung/s3c-ac97.c | 521 ++++++++++++++++ sound/soc/samsung/s3c-ac97.h | 23 + sound/soc/samsung/s3c-dma.c | 483 +++++++++++++++ sound/soc/samsung/s3c-dma.h | 31 + sound/soc/samsung/s3c-i2s-v2.c | 776 ++++++++++++++++++++++++ sound/soc/samsung/s3c-i2s-v2.h | 103 ++++ sound/soc/samsung/s3c-pcm.c | 554 +++++++++++++++++ sound/soc/samsung/s3c-pcm.h | 123 ++++ sound/soc/samsung/s3c2412-i2s.c | 190 ++++++ sound/soc/samsung/s3c2412-i2s.h | 29 + sound/soc/samsung/s3c24xx-i2s.c | 503 +++++++++++++++ sound/soc/samsung/s3c24xx-i2s.h | 37 ++ sound/soc/samsung/s3c24xx_simtec.c | 394 ++++++++++++ sound/soc/samsung/s3c24xx_simtec.h | 22 + sound/soc/samsung/s3c24xx_simtec_hermes.c | 153 +++++ sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 137 +++++ sound/soc/samsung/s3c24xx_uda134x.c | 373 ++++++++++++ sound/soc/samsung/s3c64xx-i2s-v4.c | 209 +++++++ sound/soc/samsung/s3c64xx-i2s.c | 190 ++++++ sound/soc/samsung/s3c64xx-i2s.h | 42 ++ sound/soc/samsung/smdk2443_wm9710.c | 81 +++ sound/soc/samsung/smdk64xx_wm8580.c | 266 ++++++++ sound/soc/samsung/smdk_wm9713.c | 94 +++ 64 files changed, 7147 insertions(+), 7147 deletions(-) delete mode 100644 sound/soc/s3c24xx/Kconfig delete mode 100644 sound/soc/s3c24xx/Makefile delete mode 100644 sound/soc/s3c24xx/jive_wm8750.c delete mode 100644 sound/soc/s3c24xx/lm4857.h delete mode 100644 sound/soc/s3c24xx/ln2440sbc_alc650.c delete mode 100644 sound/soc/s3c24xx/neo1973_gta02_wm8753.c delete mode 100644 sound/soc/s3c24xx/neo1973_wm8753.c delete mode 100644 sound/soc/s3c24xx/regs-i2s-v2.h delete mode 100644 sound/soc/s3c24xx/s3c-ac97.c delete mode 100644 sound/soc/s3c24xx/s3c-ac97.h delete mode 100644 sound/soc/s3c24xx/s3c-dma.c delete mode 100644 sound/soc/s3c24xx/s3c-dma.h delete mode 100644 sound/soc/s3c24xx/s3c-i2s-v2.c delete mode 100644 sound/soc/s3c24xx/s3c-i2s-v2.h delete mode 100644 sound/soc/s3c24xx/s3c-pcm.c delete mode 100644 sound/soc/s3c24xx/s3c-pcm.h delete mode 100644 sound/soc/s3c24xx/s3c2412-i2s.c delete mode 100644 sound/soc/s3c24xx/s3c2412-i2s.h delete mode 100644 sound/soc/s3c24xx/s3c24xx-i2s.c delete mode 100644 sound/soc/s3c24xx/s3c24xx-i2s.h delete mode 100644 sound/soc/s3c24xx/s3c24xx_simtec.c delete mode 100644 sound/soc/s3c24xx/s3c24xx_simtec.h delete mode 100644 sound/soc/s3c24xx/s3c24xx_simtec_hermes.c delete mode 100644 sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c delete mode 100644 sound/soc/s3c24xx/s3c24xx_uda134x.c delete mode 100644 sound/soc/s3c24xx/s3c64xx-i2s-v4.c delete mode 100644 sound/soc/s3c24xx/s3c64xx-i2s.c delete mode 100644 sound/soc/s3c24xx/s3c64xx-i2s.h delete mode 100644 sound/soc/s3c24xx/smdk2443_wm9710.c delete mode 100644 sound/soc/s3c24xx/smdk64xx_wm8580.c delete mode 100644 sound/soc/s3c24xx/smdk_wm9713.c create mode 100644 sound/soc/samsung/Kconfig create mode 100644 sound/soc/samsung/Makefile create mode 100644 sound/soc/samsung/jive_wm8750.c create mode 100644 sound/soc/samsung/lm4857.h create mode 100644 sound/soc/samsung/ln2440sbc_alc650.c create mode 100644 sound/soc/samsung/neo1973_gta02_wm8753.c create mode 100644 sound/soc/samsung/neo1973_wm8753.c create mode 100644 sound/soc/samsung/regs-i2s-v2.h create mode 100644 sound/soc/samsung/s3c-ac97.c create mode 100644 sound/soc/samsung/s3c-ac97.h create mode 100644 sound/soc/samsung/s3c-dma.c create mode 100644 sound/soc/samsung/s3c-dma.h create mode 100644 sound/soc/samsung/s3c-i2s-v2.c create mode 100644 sound/soc/samsung/s3c-i2s-v2.h create mode 100644 sound/soc/samsung/s3c-pcm.c create mode 100644 sound/soc/samsung/s3c-pcm.h create mode 100644 sound/soc/samsung/s3c2412-i2s.c create mode 100644 sound/soc/samsung/s3c2412-i2s.h create mode 100644 sound/soc/samsung/s3c24xx-i2s.c create mode 100644 sound/soc/samsung/s3c24xx-i2s.h create mode 100644 sound/soc/samsung/s3c24xx_simtec.c create mode 100644 sound/soc/samsung/s3c24xx_simtec.h create mode 100644 sound/soc/samsung/s3c24xx_simtec_hermes.c create mode 100644 sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c create mode 100644 sound/soc/samsung/s3c24xx_uda134x.c create mode 100644 sound/soc/samsung/s3c64xx-i2s-v4.c create mode 100644 sound/soc/samsung/s3c64xx-i2s.c create mode 100644 sound/soc/samsung/s3c64xx-i2s.h create mode 100644 sound/soc/samsung/smdk2443_wm9710.c create mode 100644 sound/soc/samsung/smdk64xx_wm8580.c create mode 100644 sound/soc/samsung/smdk_wm9713.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index b1749bc..5d467ef 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -32,7 +32,7 @@ source "sound/soc/fsl/Kconfig" source "sound/soc/imx/Kconfig" source "sound/soc/omap/Kconfig" source "sound/soc/pxa/Kconfig" -source "sound/soc/s3c24xx/Kconfig" +source "sound/soc/samsung/Kconfig" source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" source "sound/soc/txx9/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 1470141..9572925 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -10,7 +10,7 @@ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += imx/ obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += pxa/ -obj-$(CONFIG_SND_SOC) += s3c24xx/ +obj-$(CONFIG_SND_SOC) += samsung/ obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ obj-$(CONFIG_SND_SOC) += txx9/ diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig deleted file mode 100644 index 2a7cc22..0000000 --- a/sound/soc/s3c24xx/Kconfig +++ /dev/null @@ -1,127 +0,0 @@ -config SND_S3C24XX_SOC - tristate "SoC Audio for the Samsung S3CXXXX chips" - depends on ARCH_S3C2410 || ARCH_S3C64XX - select S3C64XX_DMA if ARCH_S3C64XX - help - Say Y or M if you want to add support for codecs attached to - the S3C24XX AC97 or I2S interfaces. You will also need to - select the audio interfaces to support below. - -config SND_S3C24XX_SOC_I2S - tristate - select S3C2410_DMA - -config SND_S3C_I2SV2_SOC - tristate - -config SND_S3C2412_SOC_I2S - tristate - select SND_S3C_I2SV2_SOC - select S3C2410_DMA - -config SND_S3C64XX_SOC_I2S - tristate - select SND_S3C_I2SV2_SOC - select S3C64XX_DMA - -config SND_S3C64XX_SOC_I2S_V4 - tristate - select SND_S3C_I2SV2_SOC - select S3C64XX_DMA - -config SND_S3C_SOC_PCM - tristate - -config SND_S3C_SOC_AC97 - tristate - select SND_SOC_AC97_BUS - -config SND_S3C24XX_SOC_NEO1973_WM8753 - tristate "SoC I2S Audio support for NEO1973 - WM8753" - depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA01 - select SND_S3C24XX_SOC_I2S - select SND_SOC_WM8753 - help - Say Y if you want to add support for SoC audio on smdk2440 - with the WM8753. - -config SND_S3C24XX_SOC_NEO1973_GTA02_WM8753 - tristate "Audio support for the Openmoko Neo FreeRunner (GTA02)" - depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA02 - select SND_S3C24XX_SOC_I2S - select SND_SOC_WM8753 - help - This driver provides audio support for the Openmoko Neo FreeRunner - smartphone. - -config SND_S3C24XX_SOC_JIVE_WM8750 - tristate "SoC I2S Audio support for Jive" - depends on SND_S3C24XX_SOC && MACH_JIVE - select SND_SOC_WM8750 - select SND_S3C2412_SOC_I2S - help - Sat Y if you want to add support for SoC audio on the Jive. - -config SND_S3C64XX_SOC_WM8580 - tristate "SoC I2S Audio support for WM8580 on SMDK64XX" - depends on SND_S3C24XX_SOC && MACH_SMDK6410 - select SND_SOC_WM8580 - select SND_S3C64XX_SOC_I2S_V4 - help - Say Y if you want to add support for SoC audio on the SMDK6410. - -config SND_S3C24XX_SOC_SMDK2443_WM9710 - tristate "SoC AC97 Audio support for SMDK2443 - WM9710" - depends on SND_S3C24XX_SOC && MACH_SMDK2443 - select S3C2410_DMA - select AC97_BUS - select SND_SOC_AC97_CODEC - select SND_S3C_SOC_AC97 - help - Say Y if you want to add support for SoC audio on smdk2443 - with the WM9710. - -config SND_S3C24XX_SOC_LN2440SBC_ALC650 - tristate "SoC AC97 Audio support for LN2440SBC - ALC650" - depends on SND_S3C24XX_SOC && ARCH_S3C2410 - select S3C2410_DMA - select AC97_BUS - select SND_SOC_AC97_CODEC - select SND_S3C_SOC_AC97 - help - Say Y if you want to add support for SoC audio on ln2440sbc - with the ALC650. - -config SND_S3C24XX_SOC_S3C24XX_UDA134X - tristate "SoC I2S Audio support UDA134X wired to a S3C24XX" - depends on SND_S3C24XX_SOC && ARCH_S3C2410 - select SND_S3C24XX_SOC_I2S - select SND_SOC_L3 - select SND_SOC_UDA134X - -config SND_S3C24XX_SOC_SIMTEC - tristate - help - Internal node for common S3C24XX/Simtec suppor - -config SND_S3C24XX_SOC_SIMTEC_TLV320AIC23 - tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards" - depends on SND_S3C24XX_SOC && ARCH_S3C2410 - select SND_S3C24XX_SOC_I2S - select SND_SOC_TLV320AIC23 - select SND_S3C24XX_SOC_SIMTEC - -config SND_S3C24XX_SOC_SIMTEC_HERMES - tristate "SoC I2S Audio support for Simtec Hermes board" - depends on SND_S3C24XX_SOC && ARCH_S3C2410 - select SND_S3C24XX_SOC_I2S - select SND_SOC_TLV320AIC3X - select SND_S3C24XX_SOC_SIMTEC - -config SND_SOC_SMDK_WM9713 - tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_S3C24XX_SOC && MACH_SMDK6410 - select SND_SOC_WM9713 - select SND_S3C_SOC_AC97 - help - Sat Y if you want to add support for SoC audio on the SMDK. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile deleted file mode 100644 index 81d8dc5..0000000 --- a/sound/soc/s3c24xx/Makefile +++ /dev/null @@ -1,43 +0,0 @@ -# S3c24XX Platform Support -snd-soc-s3c24xx-objs := s3c-dma.o -snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o -snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o -snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o -snd-soc-s3c-ac97-objs := s3c-ac97.o -snd-soc-s3c64xx-i2s-v4-objs := s3c64xx-i2s-v4.o -snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o -snd-soc-s3c-pcm-objs := s3c-pcm.o - -obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o -obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o -obj-$(CONFIG_SND_S3C_SOC_AC97) += snd-soc-s3c-ac97.o -obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o -obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o -obj-$(CONFIG_SND_S3C64XX_SOC_I2S_V4) += snd-soc-s3c64xx-i2s-v4.o -obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o -obj-$(CONFIG_SND_S3C_SOC_PCM) += snd-soc-s3c-pcm.o - -# S3C24XX Machine Support -snd-soc-jive-wm8750-objs := jive_wm8750.o -snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o -snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o -snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o -snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o -snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o -snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o -snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o -snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o -snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o -snd-soc-smdk-wm9713-objs := smdk_wm9713.o - -obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o -obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o -obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o -obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o -obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o -obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o -obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o -obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o -obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o -obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o -obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c deleted file mode 100644 index 8c108b1..0000000 --- a/sound/soc/s3c24xx/jive_wm8750.c +++ /dev/null @@ -1,196 +0,0 @@ -/* sound/soc/s3c24xx/jive_wm8750.c - * - * Copyright 2007,2008 Simtec Electronics - * - * Based on sound/soc/pxa/spitz.c - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. -*/ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/clk.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <asm/mach-types.h> - -#include "s3c-dma.h" -#include "s3c2412-i2s.h" - -#include "../codecs/wm8750.h" - -static const struct snd_soc_dapm_route audio_map[] = { - { "Headphone Jack", NULL, "LOUT1" }, - { "Headphone Jack", NULL, "ROUT1" }, - { "Internal Speaker", NULL, "LOUT2" }, - { "Internal Speaker", NULL, "ROUT2" }, - { "LINPUT1", NULL, "Line Input" }, - { "RINPUT1", NULL, "Line Input" }, -}; - -static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Internal Speaker", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), -}; - -static int jive_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - struct s3c_i2sv2_rate_calc div; - unsigned int clk = 0; - int ret = 0; - - switch (params_rate(params)) { - case 8000: - case 16000: - case 48000: - case 96000: - clk = 12288000; - break; - case 11025: - case 22050: - case 44100: - clk = 11289600; - break; - } - - s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params), - s3c_i2sv2_get_clock(cpu_dai)); - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_RCLK, div.fs_div); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_PRESCALER, - div.clk_div - 1); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops jive_ops = { - .hw_params = jive_hw_params, -}; - -static int jive_wm8750_init(struct snd_soc_codec *codec) -{ - int err; - - /* These endpoints are not being used. */ - snd_soc_dapm_nc_pin(codec, "LINPUT2"); - snd_soc_dapm_nc_pin(codec, "RINPUT2"); - snd_soc_dapm_nc_pin(codec, "LINPUT3"); - snd_soc_dapm_nc_pin(codec, "RINPUT3"); - snd_soc_dapm_nc_pin(codec, "OUT3"); - snd_soc_dapm_nc_pin(codec, "MONO"); - - /* Add jive specific widgets */ - err = snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, - ARRAY_SIZE(wm8750_dapm_widgets)); - if (err) { - printk(KERN_ERR "%s: failed to add widgets (%d)\n", - __func__, err); - return err; - } - - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_sync(codec); - - return 0; -} - -static struct snd_soc_dai_link jive_dai = { - .name = "wm8750", - .stream_name = "WM8750", - .cpu_dai = &s3c2412_i2s_dai, - .codec_dai = &wm8750_dai, - .init = jive_wm8750_init, - .ops = &jive_ops, -}; - -/* jive audio machine driver */ -static struct snd_soc_card snd_soc_machine_jive = { - .name = "Jive", - .platform = &s3c24xx_soc_platform, - .dai_link = &jive_dai, - .num_links = 1, -}; - -/* jive audio subsystem */ -static struct snd_soc_device jive_snd_devdata = { - .card = &snd_soc_machine_jive, - .codec_dev = &soc_codec_dev_wm8750, -}; - -static struct platform_device *jive_snd_device; - -static int __init jive_init(void) -{ - int ret; - - if (!machine_is_jive()) - return 0; - - printk("JIVE WM8750 Audio support\n"); - - jive_snd_device = platform_device_alloc("soc-audio", -1); - if (!jive_snd_device) - return -ENOMEM; - - platform_set_drvdata(jive_snd_device, &jive_snd_devdata); - jive_snd_devdata.dev = &jive_snd_device->dev; - ret = platform_device_add(jive_snd_device); - - if (ret) - platform_device_put(jive_snd_device); - - return ret; -} - -static void __exit jive_exit(void) -{ - platform_device_unregister(jive_snd_device); -} - -module_init(jive_init); -module_exit(jive_exit); - -MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("ALSA SoC Jive Audio support"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/lm4857.h b/sound/soc/s3c24xx/lm4857.h deleted file mode 100644 index 0cf5b70..0000000 --- a/sound/soc/s3c24xx/lm4857.h +++ /dev/null @@ -1,32 +0,0 @@ -/* - * lm4857.h -- ALSA Soc Audio Layer - * - * Copyright 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * Revision history - * 18th Jun 2007 Initial version. - */ - -#ifndef LM4857_H_ -#define LM4857_H_ - -/* The register offsets in the cache array */ -#define LM4857_MVOL 0 -#define LM4857_LVOL 1 -#define LM4857_RVOL 2 -#define LM4857_CTRL 3 - -/* the shifts required to set these bits */ -#define LM4857_3D 5 -#define LM4857_WAKEUP 5 -#define LM4857_EPGAIN 4 - -#endif /*LM4857_H_*/ - diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c deleted file mode 100644 index ffa954f..0000000 --- a/sound/soc/s3c24xx/ln2440sbc_alc650.c +++ /dev/null @@ -1,85 +0,0 @@ -/* - * SoC audio for ln2440sbc - * - * Copyright 2007 KonekTel, a.s. - * Author: Ivan Kuten - * ivan.kuten(a)promwad.com - * - * Heavily based on smdk2443_wm9710.c - * Copyright 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - */ - -#include <linux/module.h> -#include <linux/device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include "../codecs/ac97.h" -#include "s3c-dma.h" -#include "s3c-ac97.h" - -static struct snd_soc_card ln2440sbc; - -static struct snd_soc_dai_link ln2440sbc_dai[] = { -{ - .name = "AC97", - .stream_name = "AC97 HiFi", - .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], - .codec_dai = &ac97_dai, -}, -}; - -static struct snd_soc_card ln2440sbc = { - .name = "LN2440SBC", - .platform = &s3c24xx_soc_platform, - .dai_link = ln2440sbc_dai, - .num_links = ARRAY_SIZE(ln2440sbc_dai), -}; - -static struct snd_soc_device ln2440sbc_snd_ac97_devdata = { - .card = &ln2440sbc, - .codec_dev = &soc_codec_dev_ac97, -}; - -static struct platform_device *ln2440sbc_snd_ac97_device; - -static int __init ln2440sbc_init(void) -{ - int ret; - - ln2440sbc_snd_ac97_device = platform_device_alloc("soc-audio", -1); - if (!ln2440sbc_snd_ac97_device) - return -ENOMEM; - - platform_set_drvdata(ln2440sbc_snd_ac97_device, - &ln2440sbc_snd_ac97_devdata); - ln2440sbc_snd_ac97_devdata.dev = &ln2440sbc_snd_ac97_device->dev; - ret = platform_device_add(ln2440sbc_snd_ac97_device); - - if (ret) - platform_device_put(ln2440sbc_snd_ac97_device); - - return ret; -} - -static void __exit ln2440sbc_exit(void) -{ - platform_device_unregister(ln2440sbc_snd_ac97_device); -} - -module_init(ln2440sbc_init); -module_exit(ln2440sbc_exit); - -/* Module information */ -MODULE_AUTHOR("Ivan Kuten"); -MODULE_DESCRIPTION("ALSA SoC ALC650 LN2440SBC"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c deleted file mode 100644 index 209c259..0000000 --- a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c +++ /dev/null @@ -1,506 +0,0 @@ -/* - * neo1973_gta02_wm8753.c -- SoC audio for Openmoko Freerunner(GTA02) - * - * Copyright 2007 Openmoko Inc - * Author: Graeme Gregory <graeme(a)openmoko.org> - * Copyright 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory <linux(a)wolfsonmicro.com> - * Copyright 2009 Wolfson Microelectronics - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/gpio.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <asm/mach-types.h> - -#include <plat/regs-iis.h> - -#include <mach/regs-clock.h> -#include <asm/io.h> -#include <mach/gta02.h> -#include "../codecs/wm8753.h" -#include "s3c-dma.h" -#include "s3c24xx-i2s.h" - -static struct snd_soc_card neo1973_gta02; - -static int neo1973_gta02_hifi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - unsigned int pll_out = 0, bclk = 0; - int ret = 0; - unsigned long iis_clkrate; - - iis_clkrate = s3c24xx_i2s_get_clockrate(); - - switch (params_rate(params)) { - case 8000: - case 16000: - pll_out = 12288000; - break; - case 48000: - bclk = WM8753_BCLK_DIV_4; - pll_out = 12288000; - break; - case 96000: - bclk = WM8753_BCLK_DIV_2; - pll_out = 12288000; - break; - case 11025: - bclk = WM8753_BCLK_DIV_16; - pll_out = 11289600; - break; - case 22050: - bclk = WM8753_BCLK_DIV_8; - pll_out = 11289600; - break; - case 44100: - bclk = WM8753_BCLK_DIV_4; - pll_out = 11289600; - break; - case 88200: - bclk = WM8753_BCLK_DIV_2; - pll_out = 11289600; - break; - } - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set MCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, - S3C2410_IISMOD_32FS); - if (ret < 0) - return ret; - - /* set codec BCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(codec_dai, - WM8753_BCLKDIV, bclk); - if (ret < 0) - return ret; - - /* set prescaler division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, - S3C24XX_PRESCALE(4, 4)); - if (ret < 0) - return ret; - - /* codec PLL input is PCLK/4 */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, - iis_clkrate / 4, pll_out); - if (ret < 0) - return ret; - - return 0; -} - -static int neo1973_gta02_hifi_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - - /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); -} - -/* - * Neo1973 WM8753 HiFi DAI opserations. - */ -static struct snd_soc_ops neo1973_gta02_hifi_ops = { - .hw_params = neo1973_gta02_hifi_hw_params, - .hw_free = neo1973_gta02_hifi_hw_free, -}; - -static int neo1973_gta02_voice_hw_params( - struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - unsigned int pcmdiv = 0; - int ret = 0; - unsigned long iis_clkrate; - - iis_clkrate = s3c24xx_i2s_get_clockrate(); - - if (params_rate(params) != 8000) - return -EINVAL; - if (params_channels(params) != 1) - return -EINVAL; - - pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */ - - /* todo: gg check mode (DSP_B) against CSR datasheet */ - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, - 12288000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set codec PCM division for sample rate */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, - pcmdiv); - if (ret < 0) - return ret; - - /* configue and enable PLL for 12.288MHz output */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, - iis_clkrate / 4, 12288000); - if (ret < 0) - return ret; - - return 0; -} - -static int neo1973_gta02_voice_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - - /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); -} - -static struct snd_soc_ops neo1973_gta02_voice_ops = { - .hw_params = neo1973_gta02_voice_hw_params, - .hw_free = neo1973_gta02_voice_hw_free, -}; - -#define LM4853_AMP 1 -#define LM4853_SPK 2 - -static u8 lm4853_state; - -/* This has no effect, it exists only to maintain compatibility with - * existing ALSA state files. - */ -static int lm4853_set_state(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - int val = ucontrol->value.integer.value[0]; - - if (val) - lm4853_state |= LM4853_AMP; - else - lm4853_state &= ~LM4853_AMP; - - return 0; -} - -static int lm4853_get_state(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = lm4853_state & LM4853_AMP; - - return 0; -} - -static int lm4853_set_spk(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - int val = ucontrol->value.integer.value[0]; - - if (val) { - lm4853_state |= LM4853_SPK; - gpio_set_value(GTA02_GPIO_HP_IN, 0); - } else { - lm4853_state &= ~LM4853_SPK; - gpio_set_value(GTA02_GPIO_HP_IN, 1); - } - - return 0; -} - -static int lm4853_get_spk(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = (lm4853_state & LM4853_SPK) >> 1; - - return 0; -} - -static int lm4853_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, - int event) -{ - gpio_set_value(GTA02_GPIO_AMP_SHUT, SND_SOC_DAPM_EVENT_OFF(value)); - - return 0; -} - -static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { - SND_SOC_DAPM_SPK("Stereo Out", lm4853_event), - SND_SOC_DAPM_LINE("GSM Line Out", NULL), - SND_SOC_DAPM_LINE("GSM Line In", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Handset Mic", NULL), - SND_SOC_DAPM_SPK("Handset Spk", NULL), -}; - - -/* example machine audio_mapnections */ -static const struct snd_soc_dapm_route audio_map[] = { - - /* Connections to the lm4853 amp */ - {"Stereo Out", NULL, "LOUT1"}, - {"Stereo Out", NULL, "ROUT1"}, - - /* Connections to the GSM Module */ - {"GSM Line Out", NULL, "MONO1"}, - {"GSM Line Out", NULL, "MONO2"}, - {"RXP", NULL, "GSM Line In"}, - {"RXN", NULL, "GSM Line In"}, - - /* Connections to Headset */ - {"MIC1", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Headset Mic"}, - - /* Call Mic */ - {"MIC2", NULL, "Mic Bias"}, - {"MIC2N", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Handset Mic"}, - - /* Call Speaker */ - {"Handset Spk", NULL, "LOUT2"}, - {"Handset Spk", NULL, "ROUT2"}, - - /* Connect the ALC pins */ - {"ACIN", NULL, "ACOP"}, -}; - -static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = { - SOC_DAPM_PIN_SWITCH("Stereo Out"), - SOC_DAPM_PIN_SWITCH("GSM Line Out"), - SOC_DAPM_PIN_SWITCH("GSM Line In"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Handset Mic"), - SOC_DAPM_PIN_SWITCH("Handset Spk"), - - /* This has no effect, it exists only to maintain compatibility with - * existing ALSA state files. - */ - SOC_SINGLE_EXT("Amp State Switch", 6, 0, 1, 0, - lm4853_get_state, - lm4853_set_state), - SOC_SINGLE_EXT("Amp Spk Switch", 7, 0, 1, 0, - lm4853_get_spk, - lm4853_set_spk), -}; - -/* - * This is an example machine initialisation for a wm8753 connected to a - * neo1973 GTA02. - */ -static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec) -{ - int err; - - /* set up NC codec pins */ - snd_soc_dapm_nc_pin(codec, "OUT3"); - snd_soc_dapm_nc_pin(codec, "OUT4"); - snd_soc_dapm_nc_pin(codec, "LINE1"); - snd_soc_dapm_nc_pin(codec, "LINE2"); - - /* Add neo1973 gta02 specific widgets */ - snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, - ARRAY_SIZE(wm8753_dapm_widgets)); - - /* add neo1973 gta02 specific controls */ - err = snd_soc_add_controls(codec, wm8753_neo1973_gta02_controls, - ARRAY_SIZE(wm8753_neo1973_gta02_controls)); - - if (err < 0) - return err; - - /* set up neo1973 gta02 specific audio path audio_map */ - snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - - /* set endpoints to default off mode */ - snd_soc_dapm_disable_pin(codec, "Stereo Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Handset Mic"); - snd_soc_dapm_disable_pin(codec, "Handset Spk"); - - /* allow audio paths from the GSM modem to run during suspend */ - snd_soc_dapm_ignore_suspend(codec, "Stereo Out"); - snd_soc_dapm_ignore_suspend(codec, "GSM Line Out"); - snd_soc_dapm_ignore_suspend(codec, "GSM Line In"); - snd_soc_dapm_ignore_suspend(codec, "Headset Mic"); - snd_soc_dapm_ignore_suspend(codec, "Handset Mic"); - snd_soc_dapm_ignore_suspend(codec, "Handset Spk"); - - snd_soc_dapm_sync(codec); - - return 0; -} - -/* - * BT Codec DAI - */ -static struct snd_soc_dai bt_dai = { - .name = "Bluetooth", - .id = 0, - .playback = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, -}; - -static struct snd_soc_dai_link neo1973_gta02_dai[] = { -{ /* Hifi Playback - for similatious use with voice below */ - .name = "WM8753", - .stream_name = "WM8753 HiFi", - .cpu_dai = &s3c24xx_i2s_dai, - .codec_dai = &wm8753_dai[WM8753_DAI_HIFI], - .init = neo1973_gta02_wm8753_init, - .ops = &neo1973_gta02_hifi_ops, -}, -{ /* Voice via BT */ - .name = "Bluetooth", - .stream_name = "Voice", - .cpu_dai = &bt_dai, - .codec_dai = &wm8753_dai[WM8753_DAI_VOICE], - .ops = &neo1973_gta02_voice_ops, -}, -}; - -static struct snd_soc_card neo1973_gta02 = { - .name = "neo1973-gta02", - .platform = &s3c24xx_soc_platform, - .dai_link = neo1973_gta02_dai, - .num_links = ARRAY_SIZE(neo1973_gta02_dai), -}; - -static struct snd_soc_device neo1973_gta02_snd_devdata = { - .card = &neo1973_gta02, - .codec_dev = &soc_codec_dev_wm8753, -}; - -static struct platform_device *neo1973_gta02_snd_device; - -static int __init neo1973_gta02_init(void) -{ - int ret; - - if (!machine_is_neo1973_gta02()) { - printk(KERN_INFO - "Only GTA02 is supported by this ASoC driver\n"); - return -ENODEV; - } - - /* register bluetooth DAI here */ - ret = snd_soc_register_dai(&bt_dai); - if (ret) - return ret; - - neo1973_gta02_snd_device = platform_device_alloc("soc-audio", -1); - if (!neo1973_gta02_snd_device) - return -ENOMEM; - - platform_set_drvdata(neo1973_gta02_snd_device, - &neo1973_gta02_snd_devdata); - neo1973_gta02_snd_devdata.dev = &neo1973_gta02_snd_device->dev; - ret = platform_device_add(neo1973_gta02_snd_device); - - if (ret) { - platform_device_put(neo1973_gta02_snd_device); - return ret; - } - - /* Initialise GPIOs used by amp */ - ret = gpio_request(GTA02_GPIO_HP_IN, "GTA02_HP_IN"); - if (ret) { - pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_HP_IN); - goto err_unregister_device; - } - - ret = gpio_direction_output(GTA02_GPIO_AMP_HP_IN, 1); - if (ret) { - pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_HP_IN); - goto err_free_gpio_hp_in; - } - - ret = gpio_request(GTA02_GPIO_AMP_SHUT, "GTA02_AMP_SHUT"); - if (ret) { - pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_AMP_SHUT); - goto err_free_gpio_hp_in; - } - - ret = gpio_direction_output(GTA02_GPIO_AMP_SHUT, 1); - if (ret) { - pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_AMP_SHUT); - goto err_free_gpio_amp_shut; - } - - return 0; - -err_free_gpio_amp_shut: - gpio_free(GTA02_GPIO_AMP_SHUT); -err_free_gpio_hp_in: - gpio_free(GTA02_GPIO_HP_IN); -err_unregister_device: - platform_device_unregister(neo1973_gta02_snd_device); - return ret; -} -module_init(neo1973_gta02_init); - -static void __exit neo1973_gta02_exit(void) -{ - snd_soc_unregister_dai(&bt_dai); - platform_device_unregister(neo1973_gta02_snd_device); - gpio_free(GTA02_GPIO_HP_IN); - gpio_free(GTA02_GPIO_AMP_SHUT); -} -module_exit(neo1973_gta02_exit); - -/* Module information */ -MODULE_AUTHOR("Graeme Gregory, graeme(a)openmoko.org"); -MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 GTA02"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c deleted file mode 100644 index 0cb4f86..0000000 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ /dev/null @@ -1,707 +0,0 @@ -/* - * neo1973_wm8753.c -- SoC audio for Neo1973 - * - * Copyright 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/timer.h> -#include <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/i2c.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> -#include <sound/tlv.h> - -#include <asm/mach-types.h> -#include <asm/hardware/scoop.h> -#include <mach/regs-clock.h> -#include <mach/regs-gpio.h> -#include <mach/hardware.h> -#include <linux/io.h> -#include <mach/spi-gpio.h> - -#include <plat/regs-iis.h> - -#include "../codecs/wm8753.h" -#include "lm4857.h" -#include "s3c-dma.h" -#include "s3c24xx-i2s.h" - -/* define the scenarios */ -#define NEO_AUDIO_OFF 0 -#define NEO_GSM_CALL_AUDIO_HANDSET 1 -#define NEO_GSM_CALL_AUDIO_HEADSET 2 -#define NEO_GSM_CALL_AUDIO_BLUETOOTH 3 -#define NEO_STEREO_TO_SPEAKERS 4 -#define NEO_STEREO_TO_HEADPHONES 5 -#define NEO_CAPTURE_HANDSET 6 -#define NEO_CAPTURE_HEADSET 7 -#define NEO_CAPTURE_BLUETOOTH 8 - -static struct snd_soc_card neo1973; -static struct i2c_client *i2c; - -static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - unsigned int pll_out = 0, bclk = 0; - int ret = 0; - unsigned long iis_clkrate; - - pr_debug("Entered %s\n", __func__); - - iis_clkrate = s3c24xx_i2s_get_clockrate(); - - switch (params_rate(params)) { - case 8000: - case 16000: - pll_out = 12288000; - break; - case 48000: - bclk = WM8753_BCLK_DIV_4; - pll_out = 12288000; - break; - case 96000: - bclk = WM8753_BCLK_DIV_2; - pll_out = 12288000; - break; - case 11025: - bclk = WM8753_BCLK_DIV_16; - pll_out = 11289600; - break; - case 22050: - bclk = WM8753_BCLK_DIV_8; - pll_out = 11289600; - break; - case 44100: - bclk = WM8753_BCLK_DIV_4; - pll_out = 11289600; - break; - case 88200: - bclk = WM8753_BCLK_DIV_2; - pll_out = 11289600; - break; - } - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set MCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, - S3C2410_IISMOD_32FS); - if (ret < 0) - return ret; - - /* set codec BCLK division for sample rate */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk); - if (ret < 0) - return ret; - - /* set prescaler division for sample rate */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, - S3C24XX_PRESCALE(4, 4)); - if (ret < 0) - return ret; - - /* codec PLL input is PCLK/4 */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, - iis_clkrate / 4, pll_out); - if (ret < 0) - return ret; - - return 0; -} - -static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - - pr_debug("Entered %s\n", __func__); - - /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); -} - -/* - * Neo1973 WM8753 HiFi DAI opserations. - */ -static struct snd_soc_ops neo1973_hifi_ops = { - .hw_params = neo1973_hifi_hw_params, - .hw_free = neo1973_hifi_hw_free, -}; - -static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - unsigned int pcmdiv = 0; - int ret = 0; - unsigned long iis_clkrate; - - pr_debug("Entered %s\n", __func__); - - iis_clkrate = s3c24xx_i2s_get_clockrate(); - - if (params_rate(params) != 8000) - return -EINVAL; - if (params_channels(params) != 1) - return -EINVAL; - - pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */ - - /* todo: gg check mode (DSP_B) against CSR datasheet */ - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set codec PCM division for sample rate */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv); - if (ret < 0) - return ret; - - /* configue and enable PLL for 12.288MHz output */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, - iis_clkrate / 4, 12288000); - if (ret < 0) - return ret; - - return 0; -} - -static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - - pr_debug("Entered %s\n", __func__); - - /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); -} - -static struct snd_soc_ops neo1973_voice_ops = { - .hw_params = neo1973_voice_hw_params, - .hw_free = neo1973_voice_hw_free, -}; - -static int neo1973_scenario; - -static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = neo1973_scenario; - return 0; -} - -static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) -{ - pr_debug("Entered %s\n", __func__); - - switch (neo1973_scenario) { - case NEO_AUDIO_OFF: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_HANDSET: - snd_soc_dapm_enable_pin(codec, "Audio Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_enable_pin(codec, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_HEADSET: - snd_soc_dapm_enable_pin(codec, "Audio Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line In"); - snd_soc_dapm_enable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_BLUETOOTH: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line Out"); - snd_soc_dapm_enable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - break; - case NEO_STEREO_TO_SPEAKERS: - snd_soc_dapm_enable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - break; - case NEO_STEREO_TO_HEADPHONES: - snd_soc_dapm_enable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - break; - case NEO_CAPTURE_HANDSET: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_enable_pin(codec, "Call Mic"); - break; - case NEO_CAPTURE_HEADSET: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_enable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - break; - case NEO_CAPTURE_BLUETOOTH: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - break; - default: - snd_soc_dapm_disable_pin(codec, "Audio Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line Out"); - snd_soc_dapm_disable_pin(codec, "GSM Line In"); - snd_soc_dapm_disable_pin(codec, "Headset Mic"); - snd_soc_dapm_disable_pin(codec, "Call Mic"); - } - - snd_soc_dapm_sync(codec); - - return 0; -} - -static int neo1973_set_scenario(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - - pr_debug("Entered %s\n", __func__); - - if (neo1973_scenario == ucontrol->value.integer.value[0]) - return 0; - - neo1973_scenario = ucontrol->value.integer.value[0]; - set_scenario_endpoints(codec, neo1973_scenario); - return 1; -} - -static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0}; - -static void lm4857_write_regs(void) -{ - pr_debug("Entered %s\n", __func__); - - if (i2c_master_send(i2c, lm4857_regs, 4) != 4) - printk(KERN_ERR "lm4857: i2c write failed\n"); -} - -static int lm4857_get_reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int reg = mc->reg; - int shift = mc->shift; - int mask = mc->max; - - pr_debug("Entered %s\n", __func__); - - ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask; - return 0; -} - -static int lm4857_set_reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - int reg = mc->reg; - int shift = mc->shift; - int mask = mc->max; - - if (((lm4857_regs[reg] >> shift) & mask) == - ucontrol->value.integer.value[0]) - return 0; - - lm4857_regs[reg] &= ~(mask << shift); - lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift; - lm4857_write_regs(); - return 1; -} - -static int lm4857_get_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - u8 value = lm4857_regs[LM4857_CTRL] & 0x0F; - - pr_debug("Entered %s\n", __func__); - - if (value) - value -= 5; - - ucontrol->value.integer.value[0] = value; - return 0; -} - -static int lm4857_set_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - u8 value = ucontrol->value.integer.value[0]; - - pr_debug("Entered %s\n", __func__); - - if (value) - value += 5; - - if ((lm4857_regs[LM4857_CTRL] & 0x0F) == value) - return 0; - - lm4857_regs[LM4857_CTRL] &= 0xF0; - lm4857_regs[LM4857_CTRL] |= value; - lm4857_write_regs(); - return 1; -} - -static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { - SND_SOC_DAPM_LINE("Audio Out", NULL), - SND_SOC_DAPM_LINE("GSM Line Out", NULL), - SND_SOC_DAPM_LINE("GSM Line In", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Call Mic", NULL), -}; - - -static const struct snd_soc_dapm_route dapm_routes[] = { - - /* Connections to the lm4857 amp */ - {"Audio Out", NULL, "LOUT1"}, - {"Audio Out", NULL, "ROUT1"}, - - /* Connections to the GSM Module */ - {"GSM Line Out", NULL, "MONO1"}, - {"GSM Line Out", NULL, "MONO2"}, - {"RXP", NULL, "GSM Line In"}, - {"RXN", NULL, "GSM Line In"}, - - /* Connections to Headset */ - {"MIC1", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Headset Mic"}, - - /* Call Mic */ - {"MIC2", NULL, "Mic Bias"}, - {"MIC2N", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Call Mic"}, - - /* Connect the ALC pins */ - {"ACIN", NULL, "ACOP"}, -}; - -static const char *lm4857_mode[] = { - "Off", - "Call Speaker", - "Stereo Speakers", - "Stereo Speakers + Headphones", - "Headphones" -}; - -static const struct soc_enum lm4857_mode_enum[] = { - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode), -}; - -static const char *neo_scenarios[] = { - "Off", - "GSM Handset", - "GSM Headset", - "GSM Bluetooth", - "Speakers", - "Headphones", - "Capture Handset", - "Capture Headset", - "Capture Bluetooth" -}; - -static const struct soc_enum neo_scenario_enum[] = { - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios), -}; - -static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0); -static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0); - -static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { - SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg, stereo_tlv), - SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg, stereo_tlv), - SOC_SINGLE_EXT_TLV("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg, mono_tlv), - SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0], - lm4857_get_mode, lm4857_set_mode), - SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0], - neo1973_get_scenario, neo1973_set_scenario), - SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Fast Wakeup Playback Switch", LM4857_CTRL, 5, 1, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Earpiece 6dB Playback Switch", LM4857_CTRL, 4, 1, 0, - lm4857_get_reg, lm4857_set_reg), -}; - -/* - * This is an example machine initialisation for a wm8753 connected to a - * neo1973 II. It is missing logic to detect hp/mic insertions and logic - * to re-route the audio in such an event. - */ -static int neo1973_wm8753_init(struct snd_soc_codec *codec) -{ - int err; - - pr_debug("Entered %s\n", __func__); - - /* set up NC codec pins */ - snd_soc_dapm_nc_pin(codec, "LOUT2"); - snd_soc_dapm_nc_pin(codec, "ROUT2"); - snd_soc_dapm_nc_pin(codec, "OUT3"); - snd_soc_dapm_nc_pin(codec, "OUT4"); - snd_soc_dapm_nc_pin(codec, "LINE1"); - snd_soc_dapm_nc_pin(codec, "LINE2"); - - /* Add neo1973 specific widgets */ - snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, - ARRAY_SIZE(wm8753_dapm_widgets)); - - /* set endpoints to default mode */ - set_scenario_endpoints(codec, NEO_AUDIO_OFF); - - /* add neo1973 specific controls */ - err = snd_soc_add_controls(codec, wm8753_neo1973_controls, - ARRAY_SIZE(8753_neo1973_controls)); - if (err < 0) - return err; - - /* set up neo1973 specific audio routes */ - err = snd_soc_dapm_add_routes(codec, dapm_routes, - ARRAY_SIZE(dapm_routes)); - - snd_soc_dapm_sync(codec); - return 0; -} - -/* - * BT Codec DAI - */ -static struct snd_soc_dai bt_dai = { - .name = "Bluetooth", - .id = 0, - .playback = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, -}; - -static struct snd_soc_dai_link neo1973_dai[] = { -{ /* Hifi Playback - for similatious use with voice below */ - .name = "WM8753", - .stream_name = "WM8753 HiFi", - .cpu_dai = &s3c24xx_i2s_dai, - .codec_dai = &wm8753_dai[WM8753_DAI_HIFI], - .init = neo1973_wm8753_init, - .ops = &neo1973_hifi_ops, -}, -{ /* Voice via BT */ - .name = "Bluetooth", - .stream_name = "Voice", - .cpu_dai = &bt_dai, - .codec_dai = &wm8753_dai[WM8753_DAI_VOICE], - .ops = &neo1973_voice_ops, -}, -}; - -static struct snd_soc_card neo1973 = { - .name = "neo1973", - .platform = &s3c24xx_soc_platform, - .dai_link = neo1973_dai, - .num_links = ARRAY_SIZE(neo1973_dai), -}; - -static struct snd_soc_device neo1973_snd_devdata = { - .card = &neo1973, - .codec_dev = &soc_codec_dev_wm8753, -}; - -static int lm4857_i2c_probe(struct i2c_client *client, - const struct i2c_device_id *id) -{ - pr_debug("Entered %s\n", __func__); - - i2c = client; - - lm4857_write_regs(); - return 0; -} - -static int lm4857_i2c_remove(struct i2c_client *client) -{ - pr_debug("Entered %s\n", __func__); - - i2c = NULL; - - return 0; -} - -static u8 lm4857_state; - -static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) -{ - pr_debug("Entered %s\n", __func__); - - dev_dbg(&dev->dev, "lm4857_suspend\n"); - lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf; - if (lm4857_state) { - lm4857_regs[LM4857_CTRL] &= 0xf0; - lm4857_write_regs(); - } - return 0; -} - -static int lm4857_resume(struct i2c_client *dev) -{ - pr_debug("Entered %s\n", __func__); - - if (lm4857_state) { - lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f); - lm4857_write_regs(); - } - return 0; -} - -static void lm4857_shutdown(struct i2c_client *dev) -{ - pr_debug("Entered %s\n", __func__); - - dev_dbg(&dev->dev, "lm4857_shutdown\n"); - lm4857_regs[LM4857_CTRL] &= 0xf0; - lm4857_write_regs(); -} - -static const struct i2c_device_id lm4857_i2c_id[] = { - { "neo1973_lm4857", 0 }, - { } -}; - -static struct i2c_driver lm4857_i2c_driver = { - .driver = { - .name = "LM4857 I2C Amp", - .owner = THIS_MODULE, - }, - .suspend = lm4857_suspend, - .resume = lm4857_resume, - .shutdown = lm4857_shutdown, - .probe = lm4857_i2c_probe, - .remove = lm4857_i2c_remove, - .id_table = lm4857_i2c_id, -}; - -static struct platform_device *neo1973_snd_device; - -static int __init neo1973_init(void) -{ - int ret; - - pr_debug("Entered %s\n", __func__); - - if (!machine_is_neo1973_gta01()) { - printk(KERN_INFO - "Only GTA01 hardware supported by ASoC driver\n"); - return -ENODEV; - } - - neo1973_snd_device = platform_device_alloc("soc-audio", -1); - if (!neo1973_snd_device) - return -ENOMEM; - - platform_set_drvdata(neo1973_snd_device, &neo1973_snd_devdata); - neo1973_snd_devdata.dev = &neo1973_snd_device->dev; - ret = platform_device_add(neo1973_snd_device); - - if (ret) { - platform_device_put(neo1973_snd_device); - return ret; - } - - ret = i2c_add_driver(&lm4857_i2c_driver); - - if (ret != 0) - platform_device_unregister(neo1973_snd_device); - - return ret; -} - -static void __exit neo1973_exit(void) -{ - pr_debug("Entered %s\n", __func__); - - i2c_del_driver(&lm4857_i2c_driver); - platform_device_unregister(neo1973_snd_device); -} - -module_init(neo1973_init); -module_exit(neo1973_exit); - -/* Module information */ -MODULE_AUTHOR("Graeme Gregory, graeme(a)openmoko.org, www.openmoko.org"); -MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/regs-i2s-v2.h b/sound/soc/s3c24xx/regs-i2s-v2.h deleted file mode 100644 index 5e5e568..0000000 --- a/sound/soc/s3c24xx/regs-i2s-v2.h +++ /dev/null @@ -1,115 +0,0 @@ -/* linux/include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h - * - * Copyright 2007 Simtec Electronics <linux(a)simtec.co.uk> - * http://armlinux.simtec.co.uk/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * S3C2412 IIS register definition -*/ - -#ifndef __ASM_ARCH_REGS_S3C2412_IIS_H -#define __ASM_ARCH_REGS_S3C2412_IIS_H - -#define S3C2412_IISCON (0x00) -#define S3C2412_IISMOD (0x04) -#define S3C2412_IISFIC (0x08) -#define S3C2412_IISPSR (0x0C) -#define S3C2412_IISTXD (0x10) -#define S3C2412_IISRXD (0x14) - -#define S5PC1XX_IISFICS 0x18 -#define S5PC1XX_IISTXDS 0x1C - -#define S5PC1XX_IISCON_SW_RST (1 << 31) -#define S5PC1XX_IISCON_FRXOFSTATUS (1 << 26) -#define S5PC1XX_IISCON_FRXORINTEN (1 << 25) -#define S5PC1XX_IISCON_FTXSURSTAT (1 << 24) -#define S5PC1XX_IISCON_FTXSURINTEN (1 << 23) -#define S5PC1XX_IISCON_TXSDMAPAUSE (1 << 20) -#define S5PC1XX_IISCON_TXSDMACTIVE (1 << 18) - -#define S3C64XX_IISCON_FTXURSTATUS (1 << 17) -#define S3C64XX_IISCON_FTXURINTEN (1 << 16) -#define S3C64XX_IISCON_TXFIFO2_EMPTY (1 << 15) -#define S3C64XX_IISCON_TXFIFO1_EMPTY (1 << 14) -#define S3C64XX_IISCON_TXFIFO2_FULL (1 << 13) -#define S3C64XX_IISCON_TXFIFO1_FULL (1 << 12) - -#define S3C2412_IISCON_LRINDEX (1 << 11) -#define S3C2412_IISCON_TXFIFO_EMPTY (1 << 10) -#define S3C2412_IISCON_RXFIFO_EMPTY (1 << 9) -#define S3C2412_IISCON_TXFIFO_FULL (1 << 8) -#define S3C2412_IISCON_RXFIFO_FULL (1 << 7) -#define S3C2412_IISCON_TXDMA_PAUSE (1 << 6) -#define S3C2412_IISCON_RXDMA_PAUSE (1 << 5) -#define S3C2412_IISCON_TXCH_PAUSE (1 << 4) -#define S3C2412_IISCON_RXCH_PAUSE (1 << 3) -#define S3C2412_IISCON_TXDMA_ACTIVE (1 << 2) -#define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1) -#define S3C2412_IISCON_IIS_ACTIVE (1 << 0) - -#define S5PC1XX_IISMOD_OPCLK_CDCLK_OUT (0 << 30) -#define S5PC1XX_IISMOD_OPCLK_CDCLK_IN (1 << 30) -#define S5PC1XX_IISMOD_OPCLK_BCLK_OUT (2 << 30) -#define S5PC1XX_IISMOD_OPCLK_PCLK (3 << 30) -#define S5PC1XX_IISMOD_OPCLK_MASK (3 << 30) -#define S5PC1XX_IISMOD_TXS_IDMA (1 << 28) /* Sec_TXFIFO use I-DMA */ -#define S5PC1XX_IISMOD_BLCS_MASK 0x3 -#define S5PC1XX_IISMOD_BLCS_SHIFT 26 -#define S5PC1XX_IISMOD_BLCP_MASK 0x3 -#define S5PC1XX_IISMOD_BLCP_SHIFT 24 - -#define S3C64XX_IISMOD_C2DD_HHALF (1 << 21) /* Discard Higher-half */ -#define S3C64XX_IISMOD_C2DD_LHALF (1 << 20) /* Discard Lower-half */ -#define S3C64XX_IISMOD_C1DD_HHALF (1 << 19) -#define S3C64XX_IISMOD_C1DD_LHALF (1 << 18) -#define S3C64XX_IISMOD_DC2_EN (1 << 17) -#define S3C64XX_IISMOD_DC1_EN (1 << 16) -#define S3C64XX_IISMOD_BLC_16BIT (0 << 13) -#define S3C64XX_IISMOD_BLC_8BIT (1 << 13) -#define S3C64XX_IISMOD_BLC_24BIT (2 << 13) -#define S3C64XX_IISMOD_BLC_MASK (3 << 13) - -#define S3C2412_IISMOD_IMS_SYSMUX (1 << 10) -#define S3C2412_IISMOD_SLAVE (1 << 11) -#define S3C2412_IISMOD_MODE_TXONLY (0 << 8) -#define S3C2412_IISMOD_MODE_RXONLY (1 << 8) -#define S3C2412_IISMOD_MODE_TXRX (2 << 8) -#define S3C2412_IISMOD_MODE_MASK (3 << 8) -#define S3C2412_IISMOD_LR_LLOW (0 << 7) -#define S3C2412_IISMOD_LR_RLOW (1 << 7) -#define S3C2412_IISMOD_SDF_IIS (0 << 5) -#define S3C2412_IISMOD_SDF_MSB (1 << 5) -#define S3C2412_IISMOD_SDF_LSB (2 << 5) -#define S3C2412_IISMOD_SDF_MASK (3 << 5) -#define S3C2412_IISMOD_RCLK_256FS (0 << 3) -#define S3C2412_IISMOD_RCLK_512FS (1 << 3) -#define S3C2412_IISMOD_RCLK_384FS (2 << 3) -#define S3C2412_IISMOD_RCLK_768FS (3 << 3) -#define S3C2412_IISMOD_RCLK_MASK (3 << 3) -#define S3C2412_IISMOD_BCLK_32FS (0 << 1) -#define S3C2412_IISMOD_BCLK_48FS (1 << 1) -#define S3C2412_IISMOD_BCLK_16FS (2 << 1) -#define S3C2412_IISMOD_BCLK_24FS (3 << 1) -#define S3C2412_IISMOD_BCLK_MASK (3 << 1) -#define S3C2412_IISMOD_8BIT (1 << 0) - -#define S3C64XX_IISMOD_CDCLKCON (1 << 12) - -#define S3C2412_IISPSR_PSREN (1 << 15) - -#define S3C64XX_IISFIC_TX2COUNT(x) (((x) >> 24) & 0xf) -#define S3C64XX_IISFIC_TX1COUNT(x) (((x) >> 16) & 0xf) - -#define S3C2412_IISFIC_TXFLUSH (1 << 15) -#define S3C2412_IISFIC_RXFLUSH (1 << 7) -#define S3C2412_IISFIC_TXCOUNT(x) (((x) >> 8) & 0xf) -#define S3C2412_IISFIC_RXCOUNT(x) (((x) >> 0) & 0xf) - -#define S5PC1XX_IISFICS_TXFLUSH (1 << 15) -#define S5PC1XX_IISFICS_TXCOUNT(x) (((x) >> 8) & 0x7f) - -#endif /* __ASM_ARCH_REGS_S3C2412_IIS_H */ diff --git a/sound/soc/s3c24xx/s3c-ac97.c b/sound/soc/s3c24xx/s3c-ac97.c deleted file mode 100644 index ecf4fd0..0000000 --- a/sound/soc/s3c24xx/s3c-ac97.c +++ /dev/null @@ -1,521 +0,0 @@ -/* sound/soc/s3c24xx/s3c-ac97.c - * - * ALSA SoC Audio Layer - S3C AC97 Controller driver - * Evolved from s3c2443-ac97.c - * - * Copyright (c) 2010 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh <jassi.brar(a)samsung.com> - * Credits: Graeme Gregory, Sean Choi - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/io.h> -#include <linux/delay.h> -#include <linux/clk.h> - -#include <sound/soc.h> - -#include <plat/regs-ac97.h> -#include <mach/dma.h> -#include <plat/audio.h> - -#include "s3c-dma.h" -#include "s3c-ac97.h" - -#define AC_CMD_ADDR(x) (x << 16) -#define AC_CMD_DATA(x) (x & 0xffff) - -struct s3c_ac97_info { - unsigned state; - struct clk *ac97_clk; - void __iomem *regs; - struct mutex lock; - struct completion done; -}; -static struct s3c_ac97_info s3c_ac97; - -static struct s3c2410_dma_client s3c_dma_client_out = { - .name = "AC97 PCMOut" -}; - -static struct s3c2410_dma_client s3c_dma_client_in = { - .name = "AC97 PCMIn" -}; - -static struct s3c2410_dma_client s3c_dma_client_micin = { - .name = "AC97 MicIn" -}; - -static struct s3c_dma_params s3c_ac97_pcm_out = { - .client = &s3c_dma_client_out, - .dma_size = 4, -}; - -static struct s3c_dma_params s3c_ac97_pcm_in = { - .client = &s3c_dma_client_in, - .dma_size = 4, -}; - -static struct s3c_dma_params s3c_ac97_mic_in = { - .client = &s3c_dma_client_micin, - .dma_size = 4, -}; - -static void s3c_ac97_activate(struct snd_ac97 *ac97) -{ - u32 ac_glbctrl, stat; - - stat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT) & 0x7; - if (stat == S3C_AC97_GLBSTAT_MAINSTATE_ACTIVE) - return; /* Return if already active */ - - INIT_COMPLETION(s3c_ac97.done); - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON; - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE; - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - - if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) - printk(KERN_ERR "AC97: Unable to activate!"); -} - -static unsigned short s3c_ac97_read(struct snd_ac97 *ac97, - unsigned short reg) -{ - u32 ac_glbctrl, ac_codec_cmd; - u32 stat, addr, data; - - mutex_lock(&s3c_ac97.lock); - - s3c_ac97_activate(ac97); - - INIT_COMPLETION(s3c_ac97.done); - - ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); - ac_codec_cmd = S3C_AC97_CODEC_CMD_READ | AC_CMD_ADDR(reg); - writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); - - udelay(50); - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - - if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) - printk(KERN_ERR "AC97: Unable to read!"); - - stat = readl(s3c_ac97.regs + S3C_AC97_STAT); - addr = (stat >> 16) & 0x7f; - data = (stat & 0xffff); - - if (addr != reg) - printk(KERN_ERR "s3c-ac97: req addr = %02x, rep addr = %02x\n", reg, addr); - - mutex_unlock(&s3c_ac97.lock); - - return (unsigned short)data; -} - -static void s3c_ac97_write(struct snd_ac97 *ac97, unsigned short reg, - unsigned short val) -{ - u32 ac_glbctrl, ac_codec_cmd; - - mutex_lock(&s3c_ac97.lock); - - s3c_ac97_activate(ac97); - - INIT_COMPLETION(s3c_ac97.done); - - ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); - ac_codec_cmd = AC_CMD_ADDR(reg) | AC_CMD_DATA(val); - writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); - - udelay(50); - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - - if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) - printk(KERN_ERR "AC97: Unable to write!"); - - ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); - ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ; - writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); - - mutex_unlock(&s3c_ac97.lock); -} - -static void s3c_ac97_cold_reset(struct snd_ac97 *ac97) -{ - writel(S3C_AC97_GLBCTRL_COLDRESET, - s3c_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - writel(0, s3c_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); -} - -static void s3c_ac97_warm_reset(struct snd_ac97 *ac97) -{ - u32 stat; - - stat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT) & 0x7; - if (stat == S3C_AC97_GLBSTAT_MAINSTATE_ACTIVE) - return; /* Return if already active */ - - writel(S3C_AC97_GLBCTRL_WARMRESET, s3c_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - writel(0, s3c_ac97.regs + S3C_AC97_GLBCTRL); - msleep(1); - - s3c_ac97_activate(ac97); -} - -static irqreturn_t s3c_ac97_irq(int irq, void *dev_id) -{ - u32 ac_glbctrl, ac_glbstat; - - ac_glbstat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT); - - if (ac_glbstat & S3C_AC97_GLBSTAT_CODECREADY) { - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl &= ~S3C_AC97_GLBCTRL_CODECREADYIE; - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - - complete(&s3c_ac97.done); - } - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl |= (1<<30); /* Clear interrupt */ - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - - return IRQ_HANDLED; -} - -struct snd_ac97_bus_ops soc_ac97_ops = { - .read = s3c_ac97_read, - .write = s3c_ac97_write, - .warm_reset = s3c_ac97_warm_reset, - .reset = s3c_ac97_cold_reset, -}; -EXPORT_SYMBOL_GPL(soc_ac97_ops); - -static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - struct s3c_dma_params *dma_data; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c_ac97_pcm_out; - else - dma_data = &s3c_ac97_pcm_in; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - return 0; -} - -static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - u32 ac_glbctrl; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK; - else - ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMOUTTM_MASK; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA; - else - ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA; - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - break; - } - - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); - - return 0; -} - -static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return -ENODEV; - else - snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); - - return 0; -} - -static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - u32 ac_glbctrl; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); - - ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); - ac_glbctrl &= ~S3C_AC97_GLBCTRL_MICINTM_MASK; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ac_glbctrl |= S3C_AC97_GLBCTRL_MICINTM_DMA; - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - break; - } - - writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); - - return 0; -} - -static struct snd_soc_dai_ops s3c_ac97_dai_ops = { - .hw_params = s3c_ac97_hw_params, - .trigger = s3c_ac97_trigger, -}; - -static struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { - .hw_params = s3c_ac97_hw_mic_params, - .trigger = s3c_ac97_mic_trigger, -}; - -struct snd_soc_dai s3c_ac97_dai[] = { - [S3C_AC97_DAI_PCM] = { - .name = "s3c-ac97", - .id = S3C_AC97_DAI_PCM, - .ac97_control = 1, - .playback = { - .stream_name = "AC97 Playback", - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .stream_name = "AC97 Capture", - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &s3c_ac97_dai_ops, - }, - [S3C_AC97_DAI_MIC] = { - .name = "s3c-ac97-mic", - .id = S3C_AC97_DAI_MIC, - .ac97_control = 1, - .capture = { - .stream_name = "AC97 Mic Capture", - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &s3c_ac97_mic_dai_ops, - }, -}; -EXPORT_SYMBOL_GPL(s3c_ac97_dai); - -static __devinit int s3c_ac97_probe(struct platform_device *pdev) -{ - struct resource *mem_res, *dmatx_res, *dmarx_res, *dmamic_res, *irq_res; - struct s3c_audio_pdata *ac97_pdata; - int ret; - - ac97_pdata = pdev->dev.platform_data; - if (!ac97_pdata || !ac97_pdata->cfg_gpio) { - dev_err(&pdev->dev, "cfg_gpio callback not provided!\n"); - return -EINVAL; - } - - /* Check for availability of necessary resource */ - dmatx_res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dmatx_res) { - dev_err(&pdev->dev, "Unable to get AC97-TX dma resource\n"); - return -ENXIO; - } - - dmarx_res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!dmarx_res) { - dev_err(&pdev->dev, "Unable to get AC97-RX dma resource\n"); - return -ENXIO; - } - - dmamic_res = platform_get_resource(pdev, IORESOURCE_DMA, 2); - if (!dmamic_res) { - dev_err(&pdev->dev, "Unable to get AC97-MIC dma resource\n"); - return -ENXIO; - } - - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem_res) { - dev_err(&pdev->dev, "Unable to get register resource\n"); - return -ENXIO; - } - - irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!irq_res) { - dev_err(&pdev->dev, "AC97 IRQ not provided!\n"); - return -ENXIO; - } - - if (!request_mem_region(mem_res->start, - resource_size(mem_res), "s3c-ac97")) { - dev_err(&pdev->dev, "Unable to request register region\n"); - return -EBUSY; - } - - s3c_ac97_pcm_out.channel = dmatx_res->start; - s3c_ac97_pcm_out.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; - s3c_ac97_pcm_in.channel = dmarx_res->start; - s3c_ac97_pcm_in.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; - s3c_ac97_mic_in.channel = dmamic_res->start; - s3c_ac97_mic_in.dma_addr = mem_res->start + S3C_AC97_MIC_DATA; - - init_completion(&s3c_ac97.done); - mutex_init(&s3c_ac97.lock); - - s3c_ac97.regs = ioremap(mem_res->start, resource_size(mem_res)); - if (s3c_ac97.regs == NULL) { - dev_err(&pdev->dev, "Unable to ioremap register region\n"); - ret = -ENXIO; - goto err1; - } - - s3c_ac97.ac97_clk = clk_get(&pdev->dev, "ac97"); - if (IS_ERR(s3c_ac97.ac97_clk)) { - dev_err(&pdev->dev, "s3c-ac97 failed to get ac97_clock\n"); - ret = -ENODEV; - goto err2; - } - clk_enable(s3c_ac97.ac97_clk); - - if (ac97_pdata->cfg_gpio(pdev)) { - dev_err(&pdev->dev, "Unable to configure gpio\n"); - ret = -EINVAL; - goto err3; - } - - ret = request_irq(irq_res->start, s3c_ac97_irq, - IRQF_DISABLED, "AC97", NULL); - if (ret < 0) { - printk(KERN_ERR "s3c-ac97: interrupt request failed.\n"); - goto err4; - } - - s3c_ac97_dai[S3C_AC97_DAI_PCM].dev = &pdev->dev; - s3c_ac97_dai[S3C_AC97_DAI_MIC].dev = &pdev->dev; - - ret = snd_soc_register_dais(s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); - if (ret) - goto err5; - - return 0; - -err5: - free_irq(irq_res->start, NULL); -err4: -err3: - clk_disable(s3c_ac97.ac97_clk); - clk_put(s3c_ac97.ac97_clk); -err2: - iounmap(s3c_ac97.regs); -err1: - release_mem_region(mem_res->start, resource_size(mem_res)); - - return ret; -} - -static __devexit int s3c_ac97_remove(struct platform_device *pdev) -{ - struct resource *mem_res, *irq_res; - - snd_soc_unregister_dais(s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); - - irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (irq_res) - free_irq(irq_res->start, NULL); - - clk_disable(s3c_ac97.ac97_clk); - clk_put(s3c_ac97.ac97_clk); - - iounmap(s3c_ac97.regs); - - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (mem_res) - release_mem_region(mem_res->start, resource_size(mem_res)); - - return 0; -} - -static struct platform_driver s3c_ac97_driver = { - .probe = s3c_ac97_probe, - .remove = s3c_ac97_remove, - .driver = { - .name = "s3c-ac97", - .owner = THIS_MODULE, - }, -}; - -static int __init s3c_ac97_init(void) -{ - return platform_driver_register(&s3c_ac97_driver); -} -module_init(s3c_ac97_init); - -static void __exit s3c_ac97_exit(void) -{ - platform_driver_unregister(&s3c_ac97_driver); -} -module_exit(s3c_ac97_exit); - -MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); -MODULE_DESCRIPTION("AC97 driver for the Samsung SoC"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-ac97.h b/sound/soc/s3c24xx/s3c-ac97.h deleted file mode 100644 index 2781983..0000000 --- a/sound/soc/s3c24xx/s3c-ac97.h +++ /dev/null @@ -1,23 +0,0 @@ -/* sound/soc/s3c24xx/s3c-ac97.h - * - * ALSA SoC Audio Layer - S3C AC97 Controller driver - * Evolved from s3c2443-ac97.h - * - * Copyright (c) 2010 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh <jassi.brar(a)samsung.com> - * Credits: Graeme Gregory, Sean Choi - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __S3C_AC97_H_ -#define __S3C_AC97_H_ - -#define S3C_AC97_DAI_PCM 0 -#define S3C_AC97_DAI_MIC 1 - -extern struct snd_soc_dai s3c_ac97_dai[]; - -#endif /* __S3C_AC97_H_ */ diff --git a/sound/soc/s3c24xx/s3c-dma.c b/sound/soc/s3c24xx/s3c-dma.c deleted file mode 100644 index 1b61c23..0000000 --- a/sound/soc/s3c24xx/s3c-dma.c +++ /dev/null @@ -1,483 +0,0 @@ -/* - * s3c-dma.c -- ALSA Soc Audio Layer - * - * (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com - * - * Copyright 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/io.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <linux/dma-mapping.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <asm/dma.h> -#include <mach/hardware.h> -#include <mach/dma.h> - -#include "s3c-dma.h" - -static const struct snd_pcm_hardware s3c_dma_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_U16_LE | - SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S8, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 128*1024, - .period_bytes_min = PAGE_SIZE, - .period_bytes_max = PAGE_SIZE*2, - .periods_min = 2, - .periods_max = 128, - .fifo_size = 32, -}; - -struct s3c24xx_runtime_data { - spinlock_t lock; - int state; - unsigned int dma_loaded; - unsigned int dma_limit; - unsigned int dma_period; - dma_addr_t dma_start; - dma_addr_t dma_pos; - dma_addr_t dma_end; - struct s3c_dma_params *params; -}; - -/* s3c_dma_enqueue - * - * place a dma buffer onto the queue for the dma system - * to handle. -*/ -static void s3c_dma_enqueue(struct snd_pcm_substream *substream) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - dma_addr_t pos = prtd->dma_pos; - unsigned int limit; - int ret; - - pr_debug("Entered %s\n", __func__); - - if (s3c_dma_has_circular()) - limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; - else - limit = prtd->dma_limit; - - pr_debug("%s: loaded %d, limit %d\n", - __func__, prtd->dma_loaded, limit); - - while (prtd->dma_loaded < limit) { - unsigned long len = prtd->dma_period; - - pr_debug("dma_loaded: %d\n", prtd->dma_loaded); - - if ((pos + len) > prtd->dma_end) { - len = prtd->dma_end - pos; - pr_debug(KERN_DEBUG "%s: corrected dma len %ld\n", - __func__, len); - } - - ret = s3c2410_dma_enqueue(prtd->params->channel, - substream, pos, len); - - if (ret == 0) { - prtd->dma_loaded++; - pos += prtd->dma_period; - if (pos >= prtd->dma_end) - pos = prtd->dma_start; - } else - break; - } - - prtd->dma_pos = pos; -} - -static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, - void *dev_id, int size, - enum s3c2410_dma_buffresult result) -{ - struct snd_pcm_substream *substream = dev_id; - struct s3c24xx_runtime_data *prtd; - - pr_debug("Entered %s\n", __func__); - - if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR) - return; - - prtd = substream->runtime->private_data; - - if (substream) - snd_pcm_period_elapsed(substream); - - spin_lock(&prtd->lock); - if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) { - prtd->dma_loaded--; - s3c_dma_enqueue(substream); - } - - spin_unlock(&prtd->lock); -} - -static int s3c_dma_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - unsigned long totbytes = params_buffer_bytes(params); - struct s3c_dma_params *dma = - snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); - int ret = 0; - - - pr_debug("Entered %s\n", __func__); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!dma) - return 0; - - /* this may get called several times by oss emulation - * with different params -HW */ - if (prtd->params == NULL) { - /* prepare DMA */ - prtd->params = dma; - - pr_debug("params %p, client %p, channel %d\n", prtd->params, - prtd->params->client, prtd->params->channel); - - ret = s3c2410_dma_request(prtd->params->channel, - prtd->params->client, NULL); - - if (ret < 0) { - printk(KERN_ERR "failed to get dma channel\n"); - return ret; - } - - /* use the circular buffering if we have it available. */ - if (s3c_dma_has_circular()) - s3c2410_dma_setflags(prtd->params->channel, - S3C2410_DMAF_CIRCULAR); - } - - s3c2410_dma_set_buffdone_fn(prtd->params->channel, - s3c24xx_audio_buffdone); - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - runtime->dma_bytes = totbytes; - - spin_lock_irq(&prtd->lock); - prtd->dma_loaded = 0; - prtd->dma_limit = runtime->hw.periods_min; - prtd->dma_period = params_period_bytes(params); - prtd->dma_start = runtime->dma_addr; - prtd->dma_pos = prtd->dma_start; - prtd->dma_end = prtd->dma_start + totbytes; - spin_unlock_irq(&prtd->lock); - - return 0; -} - -static int s3c_dma_hw_free(struct snd_pcm_substream *substream) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - /* TODO - do we need to ensure DMA flushed */ - snd_pcm_set_runtime_buffer(substream, NULL); - - if (prtd->params) { - s3c2410_dma_free(prtd->params->channel, prtd->params->client); - prtd->params = NULL; - } - - return 0; -} - -static int s3c_dma_prepare(struct snd_pcm_substream *substream) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - /* return if this is a bufferless transfer e.g. - * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!prtd->params) - return 0; - - /* channel needs configuring for mem=>device, increment memory addr, - * sync to pclk, half-word transfers to the IIS-FIFO. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - s3c2410_dma_devconfig(prtd->params->channel, - S3C2410_DMASRC_MEM, - prtd->params->dma_addr); - } else { - s3c2410_dma_devconfig(prtd->params->channel, - S3C2410_DMASRC_HW, - prtd->params->dma_addr); - } - - s3c2410_dma_config(prtd->params->channel, - prtd->params->dma_size); - - /* flush the DMA channel */ - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_FLUSH); - prtd->dma_loaded = 0; - prtd->dma_pos = prtd->dma_start; - - /* enqueue dma buffers */ - s3c_dma_enqueue(substream); - - return ret; -} - -static int s3c_dma_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - spin_lock(&prtd->lock); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - prtd->state |= ST_RUNNING; - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - prtd->state &= ~ST_RUNNING; - s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STOP); - break; - - default: - ret = -EINVAL; - break; - } - - spin_unlock(&prtd->lock); - - return ret; -} - -static snd_pcm_uframes_t -s3c_dma_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd = runtime->private_data; - unsigned long res; - dma_addr_t src, dst; - - pr_debug("Entered %s\n", __func__); - - spin_lock(&prtd->lock); - s3c2410_dma_getposition(prtd->params->channel, &src, &dst); - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - res = dst - prtd->dma_start; - else - res = src - prtd->dma_start; - - spin_unlock(&prtd->lock); - - pr_debug("Pointer %x %x\n", src, dst); - - /* we seem to be getting the odd error from the pcm library due - * to out-of-bounds pointers. this is maybe due to the dma engine - * not having loaded the new values for the channel before being - * callled... (todo - fix ) - */ - - if (res >= snd_pcm_lib_buffer_bytes(substream)) { - if (res == snd_pcm_lib_buffer_bytes(substream)) - res = 0; - } - - return bytes_to_frames(substream->runtime, res); -} - -static int s3c_dma_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd; - - pr_debug("Entered %s\n", __func__); - - snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - snd_soc_set_runtime_hwparams(substream, &s3c_dma_hardware); - - prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL); - if (prtd == NULL) - return -ENOMEM; - - spin_lock_init(&prtd->lock); - - runtime->private_data = prtd; - return 0; -} - -static int s3c_dma_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct s3c24xx_runtime_data *prtd = runtime->private_data; - - pr_debug("Entered %s\n", __func__); - - if (!prtd) - pr_debug("s3c_dma_close called with prtd == NULL\n"); - - kfree(prtd); - - return 0; -} - -static int s3c_dma_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - pr_debug("Entered %s\n", __func__); - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops s3c_dma_ops = { - .open = s3c_dma_open, - .close = s3c_dma_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = s3c_dma_hw_params, - .hw_free = s3c_dma_hw_free, - .prepare = s3c_dma_prepare, - .trigger = s3c_dma_trigger, - .pointer = s3c_dma_pointer, - .mmap = s3c_dma_mmap, -}; - -static int s3c_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = s3c_dma_hardware.buffer_bytes_max; - - pr_debug("Entered %s\n", __func__); - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) - return -ENOMEM; - buf->bytes = size; - return 0; -} - -static void s3c_dma_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - pr_debug("Entered %s\n", __func__); - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -static u64 s3c_dma_mask = DMA_BIT_MASK(32); - -static int s3c_dma_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) -{ - int ret = 0; - - pr_debug("Entered %s\n", __func__); - - if (!card->dev->dma_mask) - card->dev->dma_mask = &s3c_dma_mask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; - - if (dai->playback.channels_min) { - ret = s3c_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (dai->capture.channels_min) { - ret = s3c_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} - -struct snd_soc_platform s3c24xx_soc_platform = { - .name = "s3c24xx-audio", - .pcm_ops = &s3c_dma_ops, - .pcm_new = s3c_dma_new, - .pcm_free = s3c_dma_free_dma_buffers, -}; -EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); - -static int __init s3c24xx_soc_platform_init(void) -{ - return snd_soc_register_platform(&s3c24xx_soc_platform); -} -module_init(s3c24xx_soc_platform_init); - -static void __exit s3c24xx_soc_platform_exit(void) -{ - snd_soc_unregister_platform(&s3c24xx_soc_platform); -} -module_exit(s3c24xx_soc_platform_exit); - -MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("Samsung S3C Audio DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-dma.h b/sound/soc/s3c24xx/s3c-dma.h deleted file mode 100644 index 69bb6bf..0000000 --- a/sound/soc/s3c24xx/s3c-dma.h +++ /dev/null @@ -1,31 +0,0 @@ -/* - * s3c-dma.h -- - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * ALSA PCM interface for the Samsung S3C24xx CPU - */ - -#ifndef _S3C_AUDIO_H -#define _S3C_AUDIO_H - -#define ST_RUNNING (1<<0) -#define ST_OPENED (1<<1) - -struct s3c_dma_params { - struct s3c2410_dma_client *client; /* stream identifier */ - int channel; /* Channel ID */ - dma_addr_t dma_addr; - int dma_size; /* Size of the DMA transfer */ -}; - -#define S3C24XX_DAI_I2S 0 - -/* platform data */ -extern struct snd_soc_platform s3c24xx_soc_platform; -extern struct snd_ac97_bus_ops s3c24xx_ac97_ops; - -#endif diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c deleted file mode 100644 index 13311c8..0000000 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ /dev/null @@ -1,776 +0,0 @@ -/* sound/soc/s3c24xx/s3c-i2c-v2.c - * - * ALSA Soc Audio Layer - I2S core for newer Samsung SoCs. - * - * Copyright (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory(a)wolfsonmicro.com - * linux(a)wolfsonmicro.com - * - * Copyright (c) 2008, 2007, 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/io.h> - -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <mach/dma.h> - -#include "regs-i2s-v2.h" -#include "s3c-i2s-v2.h" -#include "s3c-dma.h" - -#undef S3C_IIS_V2_SUPPORTED - -#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) -#define S3C_IIS_V2_SUPPORTED -#endif - -#ifdef CONFIG_PLAT_S3C64XX -#define S3C_IIS_V2_SUPPORTED -#endif - -#ifndef S3C_IIS_V2_SUPPORTED -#error Unsupported CPU model -#endif - -#define S3C2412_I2S_DEBUG_CON 0 - -static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) -{ - return cpu_dai->private_data; -} - -#define bit_set(v, b) (((v) & (b)) ? 1 : 0) - -#if S3C2412_I2S_DEBUG_CON -static void dbg_showcon(const char *fn, u32 con) -{ - printk(KERN_DEBUG "%s: LRI=%d, TXFEMPT=%d, RXFEMPT=%d, TXFFULL=%d, RXFFULL=%d\n", fn, - bit_set(con, S3C2412_IISCON_LRINDEX), - bit_set(con, S3C2412_IISCON_TXFIFO_EMPTY), - bit_set(con, S3C2412_IISCON_RXFIFO_EMPTY), - bit_set(con, S3C2412_IISCON_TXFIFO_FULL), - bit_set(con, S3C2412_IISCON_RXFIFO_FULL)); - - printk(KERN_DEBUG "%s: PAUSE: TXDMA=%d, RXDMA=%d, TXCH=%d, RXCH=%d\n", - fn, - bit_set(con, S3C2412_IISCON_TXDMA_PAUSE), - bit_set(con, S3C2412_IISCON_RXDMA_PAUSE), - bit_set(con, S3C2412_IISCON_TXCH_PAUSE), - bit_set(con, S3C2412_IISCON_RXCH_PAUSE)); - printk(KERN_DEBUG "%s: ACTIVE: TXDMA=%d, RXDMA=%d, IIS=%d\n", fn, - bit_set(con, S3C2412_IISCON_TXDMA_ACTIVE), - bit_set(con, S3C2412_IISCON_RXDMA_ACTIVE), - bit_set(con, S3C2412_IISCON_IIS_ACTIVE)); -} -#else -static inline void dbg_showcon(const char *fn, u32 con) -{ -} -#endif - - -/* Turn on or off the transmission path. */ -static void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on) -{ - void __iomem *regs = i2s->regs; - u32 fic, con, mod; - - pr_debug("%s(%d)\n", __func__, on); - - fic = readl(regs + S3C2412_IISFIC); - con = readl(regs + S3C2412_IISCON); - mod = readl(regs + S3C2412_IISMOD); - - pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); - - if (on) { - con |= S3C2412_IISCON_TXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; - con &= ~S3C2412_IISCON_TXDMA_PAUSE; - con &= ~S3C2412_IISCON_TXCH_PAUSE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_TXONLY: - case S3C2412_IISMOD_MODE_TXRX: - /* do nothing, we are in the right mode */ - break; - - case S3C2412_IISMOD_MODE_RXONLY: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_TXRX; - break; - - default: - dev_err(i2s->dev, "TXEN: Invalid MODE %x in IISMOD\n", - mod & S3C2412_IISMOD_MODE_MASK); - break; - } - - writel(con, regs + S3C2412_IISCON); - writel(mod, regs + S3C2412_IISMOD); - } else { - /* Note, we do not have any indication that the FIFO problems - * tha the S3C2410/2440 had apply here, so we should be able - * to disable the DMA and TX without resetting the FIFOS. - */ - - con |= S3C2412_IISCON_TXDMA_PAUSE; - con |= S3C2412_IISCON_TXCH_PAUSE; - con &= ~S3C2412_IISCON_TXDMA_ACTIVE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_TXRX: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_RXONLY; - break; - - case S3C2412_IISMOD_MODE_TXONLY: - mod &= ~S3C2412_IISMOD_MODE_MASK; - con &= ~S3C2412_IISCON_IIS_ACTIVE; - break; - - default: - dev_err(i2s->dev, "TXDIS: Invalid MODE %x in IISMOD\n", - mod & S3C2412_IISMOD_MODE_MASK); - break; - } - - writel(mod, regs + S3C2412_IISMOD); - writel(con, regs + S3C2412_IISCON); - } - - fic = readl(regs + S3C2412_IISFIC); - dbg_showcon(__func__, con); - pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); -} - -static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) -{ - void __iomem *regs = i2s->regs; - u32 fic, con, mod; - - pr_debug("%s(%d)\n", __func__, on); - - fic = readl(regs + S3C2412_IISFIC); - con = readl(regs + S3C2412_IISCON); - mod = readl(regs + S3C2412_IISMOD); - - pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); - - if (on) { - con |= S3C2412_IISCON_RXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; - con &= ~S3C2412_IISCON_RXDMA_PAUSE; - con &= ~S3C2412_IISCON_RXCH_PAUSE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_TXRX: - case S3C2412_IISMOD_MODE_RXONLY: - /* do nothing, we are in the right mode */ - break; - - case S3C2412_IISMOD_MODE_TXONLY: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_TXRX; - break; - - default: - dev_err(i2s->dev, "RXEN: Invalid MODE %x in IISMOD\n", - mod & S3C2412_IISMOD_MODE_MASK); - } - - writel(mod, regs + S3C2412_IISMOD); - writel(con, regs + S3C2412_IISCON); - } else { - /* See txctrl notes on FIFOs. */ - - con &= ~S3C2412_IISCON_RXDMA_ACTIVE; - con |= S3C2412_IISCON_RXDMA_PAUSE; - con |= S3C2412_IISCON_RXCH_PAUSE; - - switch (mod & S3C2412_IISMOD_MODE_MASK) { - case S3C2412_IISMOD_MODE_RXONLY: - con &= ~S3C2412_IISCON_IIS_ACTIVE; - mod &= ~S3C2412_IISMOD_MODE_MASK; - break; - - case S3C2412_IISMOD_MODE_TXRX: - mod &= ~S3C2412_IISMOD_MODE_MASK; - mod |= S3C2412_IISMOD_MODE_TXONLY; - break; - - default: - dev_err(i2s->dev, "RXDIS: Invalid MODE %x in IISMOD\n", - mod & S3C2412_IISMOD_MODE_MASK); - } - - writel(con, regs + S3C2412_IISCON); - writel(mod, regs + S3C2412_IISMOD); - } - - fic = readl(regs + S3C2412_IISFIC); - pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); -} - -#define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t) - -/* - * Wait for the LR signal to allow synchronisation to the L/R clock - * from the codec. May only be needed for slave mode. - */ -static int s3c2412_snd_lrsync(struct s3c_i2sv2_info *i2s) -{ - u32 iiscon; - unsigned long loops = msecs_to_loops(5); - - pr_debug("Entered %s\n", __func__); - - while (--loops) { - iiscon = readl(i2s->regs + S3C2412_IISCON); - if (iiscon & S3C2412_IISCON_LRINDEX) - break; - - cpu_relax(); - } - - if (!loops) { - printk(KERN_ERR "%s: timeout\n", __func__); - return -ETIMEDOUT; - } - - return 0; -} - -/* - * Set S3C2412 I2S DAI format - */ -static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - iismod = readl(i2s->regs + S3C2412_IISMOD); - pr_debug("hw_params r: IISMOD: %x \n", iismod); - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - i2s->master = 0; - iismod |= S3C2412_IISMOD_SLAVE; - break; - case SND_SOC_DAIFMT_CBS_CFS: - i2s->master = 1; - iismod &= ~S3C2412_IISMOD_SLAVE; - break; - default: - pr_err("unknwon master/slave format\n"); - return -EINVAL; - } - - iismod &= ~S3C2412_IISMOD_SDF_MASK; - - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_RIGHT_J: - iismod |= S3C2412_IISMOD_LR_RLOW; - iismod |= S3C2412_IISMOD_SDF_MSB; - break; - case SND_SOC_DAIFMT_LEFT_J: - iismod |= S3C2412_IISMOD_LR_RLOW; - iismod |= S3C2412_IISMOD_SDF_LSB; - break; - case SND_SOC_DAIFMT_I2S: - iismod &= ~S3C2412_IISMOD_LR_RLOW; - iismod |= S3C2412_IISMOD_SDF_IIS; - break; - default: - pr_err("Unknown data format\n"); - return -EINVAL; - } - - writel(iismod, i2s->regs + S3C2412_IISMOD); - pr_debug("hw_params w: IISMOD: %x \n", iismod); - return 0; -} - -static int s3c_i2sv2_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *socdai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai_link *dai = rtd->dai; - struct s3c_i2sv2_info *i2s = to_info(dai->cpu_dai); - struct s3c_dma_params *dma_data; - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = i2s->dma_playback; - else - dma_data = i2s->dma_capture; - - snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data); - - /* Working copies of register */ - iismod = readl(i2s->regs + S3C2412_IISMOD); - pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); - - iismod &= ~S3C64XX_IISMOD_BLC_MASK; - /* Sample size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - iismod |= S3C64XX_IISMOD_BLC_8BIT; - break; - case SNDRV_PCM_FORMAT_S16_LE: - break; - case SNDRV_PCM_FORMAT_S24_LE: - iismod |= S3C64XX_IISMOD_BLC_24BIT; - break; - } - - writel(iismod, i2s->regs + S3C2412_IISMOD); - pr_debug("%s: w: IISMOD: %x\n", __func__, iismod); - - return 0; -} - -static int s3c_i2sv2_set_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - u32 iismod = readl(i2s->regs + S3C2412_IISMOD); - - pr_debug("Entered %s\n", __func__); - pr_debug("%s r: IISMOD: %x\n", __func__, iismod); - - switch (clk_id) { - case S3C_I2SV2_CLKSRC_PCLK: - iismod &= ~S3C2412_IISMOD_IMS_SYSMUX; - break; - - case S3C_I2SV2_CLKSRC_AUDIOBUS: - iismod |= S3C2412_IISMOD_IMS_SYSMUX; - break; - - case S3C_I2SV2_CLKSRC_CDCLK: - /* Error if controller doesn't have the CDCLKCON bit */ - if (!(i2s->feature & S3C_FEATURE_CDCLKCON)) - return -EINVAL; - - switch (dir) { - case SND_SOC_CLOCK_IN: - iismod |= S3C64XX_IISMOD_CDCLKCON; - break; - case SND_SOC_CLOCK_OUT: - iismod &= ~S3C64XX_IISMOD_CDCLKCON; - break; - default: - return -EINVAL; - } - break; - - default: - return -EINVAL; - } - - writel(iismod, i2s->regs + S3C2412_IISMOD); - pr_debug("%s w: IISMOD: %x\n", __func__, iismod); - - return 0; -} - -static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_i2sv2_info *i2s = to_info(rtd->dai->cpu_dai); - int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); - unsigned long irqs; - int ret = 0; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); - - pr_debug("Entered %s\n", __func__); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - /* On start, ensure that the FIFOs are cleared and reset. */ - - writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH, - i2s->regs + S3C2412_IISFIC); - - /* clear again, just in case */ - writel(0x0, i2s->regs + S3C2412_IISFIC); - - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!i2s->master) { - ret = s3c2412_snd_lrsync(i2s); - if (ret) - goto exit_err; - } - - local_irq_save(irqs); - - if (capture) - s3c2412_snd_rxctrl(i2s, 1); - else - s3c2412_snd_txctrl(i2s, 1); - - local_irq_restore(irqs); - - /* - * Load the next buffer to DMA to meet the reqirement - * of the auto reload mechanism of S3C24XX. - * This call won't bother S3C64XX. - */ - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); - - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - local_irq_save(irqs); - - if (capture) - s3c2412_snd_rxctrl(i2s, 0); - else - s3c2412_snd_txctrl(i2s, 0); - - local_irq_restore(irqs); - break; - default: - ret = -EINVAL; - break; - } - -exit_err: - return ret; -} - -/* - * Set S3C2412 Clock dividers - */ -static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - u32 reg; - - pr_debug("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div); - - switch (div_id) { - case S3C_I2SV2_DIV_BCLK: - switch (div) { - case 16: - div = S3C2412_IISMOD_BCLK_16FS; - break; - - case 32: - div = S3C2412_IISMOD_BCLK_32FS; - break; - - case 24: - div = S3C2412_IISMOD_BCLK_24FS; - break; - - case 48: - div = S3C2412_IISMOD_BCLK_48FS; - break; - - default: - return -EINVAL; - } - - reg = readl(i2s->regs + S3C2412_IISMOD); - reg &= ~S3C2412_IISMOD_BCLK_MASK; - writel(reg | div, i2s->regs + S3C2412_IISMOD); - - pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); - break; - - case S3C_I2SV2_DIV_RCLK: - switch (div) { - case 256: - div = S3C2412_IISMOD_RCLK_256FS; - break; - - case 384: - div = S3C2412_IISMOD_RCLK_384FS; - break; - - case 512: - div = S3C2412_IISMOD_RCLK_512FS; - break; - - case 768: - div = S3C2412_IISMOD_RCLK_768FS; - break; - - default: - return -EINVAL; - } - - reg = readl(i2s->regs + S3C2412_IISMOD); - reg &= ~S3C2412_IISMOD_RCLK_MASK; - writel(reg | div, i2s->regs + S3C2412_IISMOD); - pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); - break; - - case S3C_I2SV2_DIV_PRESCALER: - if (div >= 0) { - writel((div << 8) | S3C2412_IISPSR_PSREN, - i2s->regs + S3C2412_IISPSR); - } else { - writel(0x0, i2s->regs + S3C2412_IISPSR); - } - pr_debug("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR)); - break; - - default: - return -EINVAL; - } - - return 0; -} - -static snd_pcm_sframes_t s3c2412_i2s_delay(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - u32 reg = readl(i2s->regs + S3C2412_IISFIC); - snd_pcm_sframes_t delay; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - delay = S3C2412_IISFIC_TXCOUNT(reg); - else - delay = S3C2412_IISFIC_RXCOUNT(reg); - - return delay; -} - -struct clk *s3c_i2sv2_get_clock(struct snd_soc_dai *cpu_dai) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - u32 iismod = readl(i2s->regs + S3C2412_IISMOD); - - if (iismod & S3C2412_IISMOD_IMS_SYSMUX) - return i2s->iis_cclk; - else - return i2s->iis_pclk; -} -EXPORT_SYMBOL_GPL(s3c_i2sv2_get_clock); - -/* default table of all avaialable root fs divisors */ -static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 }; - -int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, - unsigned int *fstab, - unsigned int rate, struct clk *clk) -{ - unsigned long clkrate = clk_get_rate(clk); - unsigned int div; - unsigned int fsclk; - unsigned int actual; - unsigned int fs; - unsigned int fsdiv; - signed int deviation = 0; - unsigned int best_fs = 0; - unsigned int best_div = 0; - unsigned int best_rate = 0; - unsigned int best_deviation = INT_MAX; - - pr_debug("Input clock rate %ldHz\n", clkrate); - - if (fstab == NULL) - fstab = iis_fs_tab; - - for (fs = 0; fs < ARRAY_SIZE(iis_fs_tab); fs++) { - fsdiv = iis_fs_tab[fs]; - - fsclk = clkrate / fsdiv; - div = fsclk / rate; - - if ((fsclk % rate) > (rate / 2)) - div++; - - if (div <= 1) - continue; - - actual = clkrate / (fsdiv * div); - deviation = actual - rate; - - printk(KERN_DEBUG "%ufs: div %u => result %u, deviation %d\n", - fsdiv, div, actual, deviation); - - deviation = abs(deviation); - - if (deviation < best_deviation) { - best_fs = fsdiv; - best_div = div; - best_rate = actual; - best_deviation = deviation; - } - - if (deviation == 0) - break; - } - - printk(KERN_DEBUG "best: fs=%u, div=%u, rate=%u\n", - best_fs, best_div, best_rate); - - info->fs_div = best_fs; - info->clk_div = best_div; - - return 0; -} -EXPORT_SYMBOL_GPL(s3c_i2sv2_iis_calc_rate); - -int s3c_i2sv2_probe(struct platform_device *pdev, - struct snd_soc_dai *dai, - struct s3c_i2sv2_info *i2s, - unsigned long base) -{ - struct device *dev = &pdev->dev; - unsigned int iismod; - - i2s->dev = dev; - - /* record our i2s structure for later use in the callbacks */ - dai->private_data = i2s; - - if (!base) { - struct resource *res = platform_get_resource(pdev, - IORESOURCE_MEM, - 0); - if (!res) { - dev_err(dev, "Unable to get register resource\n"); - return -ENXIO; - } - - if (!request_mem_region(res->start, resource_size(res), - "s3c64xx-i2s-v4")) { - dev_err(dev, "Unable to request register region\n"); - return -EBUSY; - } - - base = res->start; - } - - i2s->regs = ioremap(base, 0x100); - if (i2s->regs == NULL) { - dev_err(dev, "cannot ioremap registers\n"); - return -ENXIO; - } - - i2s->iis_pclk = clk_get(dev, "iis"); - if (IS_ERR(i2s->iis_pclk)) { - dev_err(dev, "failed to get iis_clock\n"); - iounmap(i2s->regs); - return -ENOENT; - } - - clk_enable(i2s->iis_pclk); - - /* Mark ourselves as in TXRX mode so we can run through our cleanup - * process without warnings. */ - iismod = readl(i2s->regs + S3C2412_IISMOD); - iismod |= S3C2412_IISMOD_MODE_TXRX; - writel(iismod, i2s->regs + S3C2412_IISMOD); - s3c2412_snd_txctrl(i2s, 0); - s3c2412_snd_rxctrl(i2s, 0); - - return 0; -} -EXPORT_SYMBOL_GPL(s3c_i2sv2_probe); - -#ifdef CONFIG_PM -static int s3c2412_i2s_suspend(struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - u32 iismod; - - if (dai->active) { - i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); - i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); - i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); - - /* some basic suspend checks */ - - iismod = readl(i2s->regs + S3C2412_IISMOD); - - if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) - pr_warning("%s: RXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) - pr_warning("%s: TXDMA active?\n", __func__); - - if (iismod & S3C2412_IISCON_IIS_ACTIVE) - pr_warning("%s: IIS active\n", __func__); - } - - return 0; -} - -static int s3c2412_i2s_resume(struct snd_soc_dai *dai) -{ - struct s3c_i2sv2_info *i2s = to_info(dai); - - pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n", - dai->active, i2s->suspend_iismod, i2s->suspend_iiscon); - - if (dai->active) { - writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); - writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); - writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); - - writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, - i2s->regs + S3C2412_IISFIC); - - ndelay(250); - writel(0x0, i2s->regs + S3C2412_IISFIC); - } - - return 0; -} -#else -#define s3c2412_i2s_suspend NULL -#define s3c2412_i2s_resume NULL -#endif - -int s3c_i2sv2_register_dai(struct snd_soc_dai *dai) -{ - struct snd_soc_dai_ops *ops = dai->ops; - - ops->trigger = s3c2412_i2s_trigger; - if (!ops->hw_params) - ops->hw_params = s3c_i2sv2_hw_params; - ops->set_fmt = s3c2412_i2s_set_fmt; - ops->set_clkdiv = s3c2412_i2s_set_clkdiv; - ops->set_sysclk = s3c_i2sv2_set_sysclk; - - /* Allow overriding by (for example) IISv4 */ - if (!ops->delay) - ops->delay = s3c2412_i2s_delay; - - dai->suspend = s3c2412_i2s_suspend; - dai->resume = s3c2412_i2s_resume; - - return snd_soc_register_dai(dai); -} -EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai); - -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h deleted file mode 100644 index 766f43a..0000000 --- a/sound/soc/s3c24xx/s3c-i2s-v2.h +++ /dev/null @@ -1,103 +0,0 @@ -/* sound/soc/s3c24xx/s3c-i2s-v2.h - * - * ALSA Soc Audio Layer - S3C_I2SV2 I2S driver - * - * Copyright (c) 2007 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. -*/ - -/* This code is the core support for the I2S block found in a number of - * Samsung SoC devices which is unofficially named I2S-V2. Currently the - * S3C2412 and the S3C64XX series use this block to provide 1 or 2 I2S - * channels via configurable GPIO. - */ - -#ifndef __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H -#define __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H __FILE__ - -#define S3C_I2SV2_DIV_BCLK (1) -#define S3C_I2SV2_DIV_RCLK (2) -#define S3C_I2SV2_DIV_PRESCALER (3) - -#define S3C_I2SV2_CLKSRC_PCLK 0 -#define S3C_I2SV2_CLKSRC_AUDIOBUS 1 -#define S3C_I2SV2_CLKSRC_CDCLK 2 - -/* Set this flag for I2S controllers that have the bit IISMOD[12] - * bridge/break RCLK signal and external Xi2sCDCLK pin. - */ -#define S3C_FEATURE_CDCLKCON (1 << 0) - -/** - * struct s3c_i2sv2_info - S3C I2S-V2 information - * @dev: The parent device passed to use from the probe. - * @regs: The pointer to the device registe block. - * @feature: Set of bit-flags indicating features of the controller. - * @master: True if the I2S core is the I2S bit clock master. - * @dma_playback: DMA information for playback channel. - * @dma_capture: DMA information for capture channel. - * @suspend_iismod: PM save for the IISMOD register. - * @suspend_iiscon: PM save for the IISCON register. - * @suspend_iispsr: PM save for the IISPSR register. - * - * This is the private codec state for the hardware associated with an - * I2S channel such as the register mappings and clock sources. - */ -struct s3c_i2sv2_info { - struct device *dev; - void __iomem *regs; - - u32 feature; - - struct clk *iis_pclk; - struct clk *iis_cclk; - - unsigned char master; - - struct s3c_dma_params *dma_playback; - struct s3c_dma_params *dma_capture; - - u32 suspend_iismod; - u32 suspend_iiscon; - u32 suspend_iispsr; -}; - -extern struct clk *s3c_i2sv2_get_clock(struct snd_soc_dai *cpu_dai); - -struct s3c_i2sv2_rate_calc { - unsigned int clk_div; /* for prescaler */ - unsigned int fs_div; /* for root frame clock */ -}; - -extern int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, - unsigned int *fstab, - unsigned int rate, struct clk *clk); - -/** - * s3c_i2sv2_probe - probe for i2s device helper - * @pdev: The platform device supplied to the original probe. - * @dai: The ASoC DAI structure supplied to the original probe. - * @i2s: Our local i2s structure to fill in. - * @base: The base address for the registers. - */ -extern int s3c_i2sv2_probe(struct platform_device *pdev, - struct snd_soc_dai *dai, - struct s3c_i2sv2_info *i2s, - unsigned long base); - -/** - * s3c_i2sv2_register_dai - register dai with soc core - * @dai: The snd_soc_dai structure to register - * - * Fill in any missing fields and then register the given dai with the - * soc core. - */ -extern int s3c_i2sv2_register_dai(struct snd_soc_dai *dai); - -#endif /* __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H */ diff --git a/sound/soc/s3c24xx/s3c-pcm.c b/sound/soc/s3c24xx/s3c-pcm.c deleted file mode 100644 index 326f0a9..0000000 --- a/sound/soc/s3c24xx/s3c-pcm.c +++ /dev/null @@ -1,554 +0,0 @@ -/* sound/soc/s3c24xx/s3c-pcm.c - * - * ALSA SoC Audio Layer - S3C PCM-Controller driver - * - * Copyright (c) 2009 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh <jassi.brar(a)samsung.com> - * based upon I2S drivers by Ben Dooks. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/kernel.h> -#include <linux/gpio.h> -#include <linux/io.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include <plat/audio.h> -#include <plat/dma.h> - -#include "s3c-dma.h" -#include "s3c-pcm.h" - -static struct s3c2410_dma_client s3c_pcm_dma_client_out = { - .name = "PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c_pcm_dma_client_in = { - .name = "PCM Stereo in" -}; - -static struct s3c_dma_params s3c_pcm_stereo_out[] = { - [0] = { - .client = &s3c_pcm_dma_client_out, - .dma_size = 4, - }, - [1] = { - .client = &s3c_pcm_dma_client_out, - .dma_size = 4, - }, -}; - -static struct s3c_dma_params s3c_pcm_stereo_in[] = { - [0] = { - .client = &s3c_pcm_dma_client_in, - .dma_size = 4, - }, - [1] = { - .client = &s3c_pcm_dma_client_in, - .dma_size = 4, - }, -}; - -static struct s3c_pcm_info s3c_pcm[2]; - -static inline struct s3c_pcm_info *to_info(struct snd_soc_dai *cpu_dai) -{ - return cpu_dai->private_data; -} - -static void s3c_pcm_snd_txctrl(struct s3c_pcm_info *pcm, int on) -{ - void __iomem *regs = pcm->regs; - u32 ctl, clkctl; - - clkctl = readl(regs + S3C_PCM_CLKCTL); - ctl = readl(regs + S3C_PCM_CTL); - ctl &= ~(S3C_PCM_CTL_TXDIPSTICK_MASK - << S3C_PCM_CTL_TXDIPSTICK_SHIFT); - - if (on) { - ctl |= S3C_PCM_CTL_TXDMA_EN; - ctl |= S3C_PCM_CTL_TXFIFO_EN; - ctl |= S3C_PCM_CTL_ENABLE; - ctl |= (0x20<<S3C_PCM_CTL_TXDIPSTICK_SHIFT); - clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; - } else { - ctl &= ~S3C_PCM_CTL_TXDMA_EN; - ctl &= ~S3C_PCM_CTL_TXFIFO_EN; - - if (!(ctl & S3C_PCM_CTL_RXFIFO_EN)) { - ctl &= ~S3C_PCM_CTL_ENABLE; - if (!pcm->idleclk) - clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; - } - } - - writel(clkctl, regs + S3C_PCM_CLKCTL); - writel(ctl, regs + S3C_PCM_CTL); -} - -static void s3c_pcm_snd_rxctrl(struct s3c_pcm_info *pcm, int on) -{ - void __iomem *regs = pcm->regs; - u32 ctl, clkctl; - - ctl = readl(regs + S3C_PCM_CTL); - clkctl = readl(regs + S3C_PCM_CLKCTL); - - if (on) { - ctl |= S3C_PCM_CTL_RXDMA_EN; - ctl |= S3C_PCM_CTL_RXFIFO_EN; - ctl |= S3C_PCM_CTL_ENABLE; - clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; - } else { - ctl &= ~S3C_PCM_CTL_RXDMA_EN; - ctl &= ~S3C_PCM_CTL_RXFIFO_EN; - - if (!(ctl & S3C_PCM_CTL_TXFIFO_EN)) { - ctl &= ~S3C_PCM_CTL_ENABLE; - if (!pcm->idleclk) - clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; - } - } - - writel(clkctl, regs + S3C_PCM_CLKCTL); - writel(ctl, regs + S3C_PCM_CTL); -} - -static int s3c_pcm_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_pcm_info *pcm = to_info(rtd->dai->cpu_dai); - unsigned long flags; - - dev_dbg(pcm->dev, "Entered %s\n", __func__); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - spin_lock_irqsave(&pcm->lock, flags); - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - s3c_pcm_snd_rxctrl(pcm, 1); - else - s3c_pcm_snd_txctrl(pcm, 1); - - spin_unlock_irqrestore(&pcm->lock, flags); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - spin_lock_irqsave(&pcm->lock, flags); - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - s3c_pcm_snd_rxctrl(pcm, 0); - else - s3c_pcm_snd_txctrl(pcm, 0); - - spin_unlock_irqrestore(&pcm->lock, flags); - break; - - default: - return -EINVAL; - } - - return 0; -} - -static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *socdai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai_link *dai = rtd->dai; - struct s3c_pcm_info *pcm = to_info(dai->cpu_dai); - struct s3c_dma_params *dma_data; - void __iomem *regs = pcm->regs; - struct clk *clk; - int sclk_div, sync_div; - unsigned long flags; - u32 clkctl; - - dev_dbg(pcm->dev, "Entered %s\n", __func__); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = pcm->dma_playback; - else - dma_data = pcm->dma_capture; - - snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data); - - /* Strictly check for sample size */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - break; - default: - return -EINVAL; - } - - spin_lock_irqsave(&pcm->lock, flags); - - /* Get hold of the PCMSOURCE_CLK */ - clkctl = readl(regs + S3C_PCM_CLKCTL); - if (clkctl & S3C_PCM_CLKCTL_SERCLKSEL_PCLK) - clk = pcm->pclk; - else - clk = pcm->cclk; - - /* Set the SCLK divider */ - sclk_div = clk_get_rate(clk) / pcm->sclk_per_fs / - params_rate(params) / 2 - 1; - - clkctl &= ~(S3C_PCM_CLKCTL_SCLKDIV_MASK - << S3C_PCM_CLKCTL_SCLKDIV_SHIFT); - clkctl |= ((sclk_div & S3C_PCM_CLKCTL_SCLKDIV_MASK) - << S3C_PCM_CLKCTL_SCLKDIV_SHIFT); - - /* Set the SYNC divider */ - sync_div = pcm->sclk_per_fs - 1; - - clkctl &= ~(S3C_PCM_CLKCTL_SYNCDIV_MASK - << S3C_PCM_CLKCTL_SYNCDIV_SHIFT); - clkctl |= ((sync_div & S3C_PCM_CLKCTL_SYNCDIV_MASK) - << S3C_PCM_CLKCTL_SYNCDIV_SHIFT); - - writel(clkctl, regs + S3C_PCM_CLKCTL); - - spin_unlock_irqrestore(&pcm->lock, flags); - - dev_dbg(pcm->dev, "PCMSOURCE_CLK-%lu SCLK=%ufs SCLK_DIV=%d SYNC_DIV=%d\n", - clk_get_rate(clk), pcm->sclk_per_fs, - sclk_div, sync_div); - - return 0; -} - -static int s3c_pcm_set_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct s3c_pcm_info *pcm = to_info(cpu_dai); - void __iomem *regs = pcm->regs; - unsigned long flags; - int ret = 0; - u32 ctl; - - dev_dbg(pcm->dev, "Entered %s\n", __func__); - - spin_lock_irqsave(&pcm->lock, flags); - - ctl = readl(regs + S3C_PCM_CTL); - - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - /* Nothing to do, NB_NF by default */ - break; - default: - dev_err(pcm->dev, "Unsupported clock inversion!\n"); - ret = -EINVAL; - goto exit; - } - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - /* Nothing to do, Master by default */ - break; - default: - dev_err(pcm->dev, "Unsupported master/slave format!\n"); - ret = -EINVAL; - goto exit; - } - - switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { - case SND_SOC_DAIFMT_CONT: - pcm->idleclk = 1; - break; - case SND_SOC_DAIFMT_GATED: - pcm->idleclk = 0; - break; - default: - dev_err(pcm->dev, "Invalid Clock gating request!\n"); - ret = -EINVAL; - goto exit; - } - - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_DSP_A: - ctl |= S3C_PCM_CTL_TXMSB_AFTER_FSYNC; - ctl |= S3C_PCM_CTL_RXMSB_AFTER_FSYNC; - break; - case SND_SOC_DAIFMT_DSP_B: - ctl &= ~S3C_PCM_CTL_TXMSB_AFTER_FSYNC; - ctl &= ~S3C_PCM_CTL_RXMSB_AFTER_FSYNC; - break; - default: - dev_err(pcm->dev, "Unsupported data format!\n"); - ret = -EINVAL; - goto exit; - } - - writel(ctl, regs + S3C_PCM_CTL); - -exit: - spin_unlock_irqrestore(&pcm->lock, flags); - - return ret; -} - -static int s3c_pcm_set_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - struct s3c_pcm_info *pcm = to_info(cpu_dai); - - switch (div_id) { - case S3C_PCM_SCLK_PER_FS: - pcm->sclk_per_fs = div; - break; - - default: - return -EINVAL; - } - - return 0; -} - -static int s3c_pcm_set_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - struct s3c_pcm_info *pcm = to_info(cpu_dai); - void __iomem *regs = pcm->regs; - u32 clkctl = readl(regs + S3C_PCM_CLKCTL); - - switch (clk_id) { - case S3C_PCM_CLKSRC_PCLK: - clkctl |= S3C_PCM_CLKCTL_SERCLKSEL_PCLK; - break; - - case S3C_PCM_CLKSRC_MUX: - clkctl &= ~S3C_PCM_CLKCTL_SERCLKSEL_PCLK; - - if (clk_get_rate(pcm->cclk) != freq) - clk_set_rate(pcm->cclk, freq); - - break; - - default: - return -EINVAL; - } - - writel(clkctl, regs + S3C_PCM_CLKCTL); - - return 0; -} - -static struct snd_soc_dai_ops s3c_pcm_dai_ops = { - .set_sysclk = s3c_pcm_set_sysclk, - .set_clkdiv = s3c_pcm_set_clkdiv, - .trigger = s3c_pcm_trigger, - .hw_params = s3c_pcm_hw_params, - .set_fmt = s3c_pcm_set_fmt, -}; - -#define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 - -#define S3C_PCM_DECLARE(n) \ -{ \ - .name = "samsung-pcm", \ - .id = (n), \ - .symmetric_rates = 1, \ - .ops = &s3c_pcm_dai_ops, \ - .playback = { \ - .channels_min = 2, \ - .channels_max = 2, \ - .rates = S3C_PCM_RATES, \ - .formats = SNDRV_PCM_FMTBIT_S16_LE, \ - }, \ - .capture = { \ - .channels_min = 2, \ - .channels_max = 2, \ - .rates = S3C_PCM_RATES, \ - .formats = SNDRV_PCM_FMTBIT_S16_LE, \ - }, \ -} - -struct snd_soc_dai s3c_pcm_dai[] = { - S3C_PCM_DECLARE(0), - S3C_PCM_DECLARE(1), -}; -EXPORT_SYMBOL_GPL(s3c_pcm_dai); - -static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) -{ - struct s3c_pcm_info *pcm; - struct snd_soc_dai *dai; - struct resource *mem_res, *dmatx_res, *dmarx_res; - struct s3c_audio_pdata *pcm_pdata; - int ret; - - /* Check for valid device index */ - if ((pdev->id < 0) || pdev->id >= ARRAY_SIZE(s3c_pcm)) { - dev_err(&pdev->dev, "id %d out of range\n", pdev->id); - return -EINVAL; - } - - pcm_pdata = pdev->dev.platform_data; - - /* Check for availability of necessary resource */ - dmatx_res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!dmatx_res) { - dev_err(&pdev->dev, "Unable to get PCM-TX dma resource\n"); - return -ENXIO; - } - - dmarx_res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!dmarx_res) { - dev_err(&pdev->dev, "Unable to get PCM-RX dma resource\n"); - return -ENXIO; - } - - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem_res) { - dev_err(&pdev->dev, "Unable to get register resource\n"); - return -ENXIO; - } - - if (pcm_pdata && pcm_pdata->cfg_gpio && pcm_pdata->cfg_gpio(pdev)) { - dev_err(&pdev->dev, "Unable to configure gpio\n"); - return -EINVAL; - } - - pcm = &s3c_pcm[pdev->id]; - pcm->dev = &pdev->dev; - - spin_lock_init(&pcm->lock); - - dai = &s3c_pcm_dai[pdev->id]; - dai->dev = &pdev->dev; - - /* Default is 128fs */ - pcm->sclk_per_fs = 128; - - pcm->cclk = clk_get(&pdev->dev, "audio-bus"); - if (IS_ERR(pcm->cclk)) { - dev_err(&pdev->dev, "failed to get audio-bus\n"); - ret = PTR_ERR(pcm->cclk); - goto err1; - } - clk_enable(pcm->cclk); - - /* record our pcm structure for later use in the callbacks */ - dai->private_data = pcm; - - if (!request_mem_region(mem_res->start, - resource_size(mem_res), "samsung-pcm")) { - dev_err(&pdev->dev, "Unable to request register region\n"); - ret = -EBUSY; - goto err2; - } - - pcm->regs = ioremap(mem_res->start, 0x100); - if (pcm->regs == NULL) { - dev_err(&pdev->dev, "cannot ioremap registers\n"); - ret = -ENXIO; - goto err3; - } - - pcm->pclk = clk_get(&pdev->dev, "pcm"); - if (IS_ERR(pcm->pclk)) { - dev_err(&pdev->dev, "failed to get pcm_clock\n"); - ret = -ENOENT; - goto err4; - } - clk_enable(pcm->pclk); - - ret = snd_soc_register_dai(dai); - if (ret != 0) { - dev_err(&pdev->dev, "failed to get pcm_clock\n"); - goto err5; - } - - s3c_pcm_stereo_in[pdev->id].dma_addr = mem_res->start - + S3C_PCM_RXFIFO; - s3c_pcm_stereo_out[pdev->id].dma_addr = mem_res->start - + S3C_PCM_TXFIFO; - - s3c_pcm_stereo_in[pdev->id].channel = dmarx_res->start; - s3c_pcm_stereo_out[pdev->id].channel = dmatx_res->start; - - pcm->dma_capture = &s3c_pcm_stereo_in[pdev->id]; - pcm->dma_playback = &s3c_pcm_stereo_out[pdev->id]; - - return 0; - -err5: - clk_disable(pcm->pclk); - clk_put(pcm->pclk); -err4: - iounmap(pcm->regs); -err3: - release_mem_region(mem_res->start, resource_size(mem_res)); -err2: - clk_disable(pcm->cclk); - clk_put(pcm->cclk); -err1: - return ret; -} - -static __devexit int s3c_pcm_dev_remove(struct platform_device *pdev) -{ - struct s3c_pcm_info *pcm = &s3c_pcm[pdev->id]; - struct resource *mem_res; - - iounmap(pcm->regs); - - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(mem_res->start, resource_size(mem_res)); - - clk_disable(pcm->cclk); - clk_disable(pcm->pclk); - clk_put(pcm->pclk); - clk_put(pcm->cclk); - - return 0; -} - -static struct platform_driver s3c_pcm_driver = { - .probe = s3c_pcm_dev_probe, - .remove = s3c_pcm_dev_remove, - .driver = { - .name = "samsung-pcm", - .owner = THIS_MODULE, - }, -}; - -static int __init s3c_pcm_init(void) -{ - return platform_driver_register(&s3c_pcm_driver); -} -module_init(s3c_pcm_init); - -static void __exit s3c_pcm_exit(void) -{ - platform_driver_unregister(&s3c_pcm_driver); -} -module_exit(s3c_pcm_exit); - -/* Module information */ -MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); -MODULE_DESCRIPTION("S3C PCM Controller Driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-pcm.h b/sound/soc/s3c24xx/s3c-pcm.h deleted file mode 100644 index 69ff997..0000000 --- a/sound/soc/s3c24xx/s3c-pcm.h +++ /dev/null @@ -1,123 +0,0 @@ -/* sound/soc/s3c24xx/s3c-pcm.h - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - */ - -#ifndef __S3C_PCM_H -#define __S3C_PCM_H __FILE__ - -/*Register Offsets */ -#define S3C_PCM_CTL (0x00) -#define S3C_PCM_CLKCTL (0x04) -#define S3C_PCM_TXFIFO (0x08) -#define S3C_PCM_RXFIFO (0x0C) -#define S3C_PCM_IRQCTL (0x10) -#define S3C_PCM_IRQSTAT (0x14) -#define S3C_PCM_FIFOSTAT (0x18) -#define S3C_PCM_CLRINT (0x20) - -/* PCM_CTL Bit-Fields */ -#define S3C_PCM_CTL_TXDIPSTICK_MASK (0x3f) -#define S3C_PCM_CTL_TXDIPSTICK_SHIFT (13) -#define S3C_PCM_CTL_RXDIPSTICK_MSK (0x3f<<7) -#define S3C_PCM_CTL_TXDMA_EN (0x1<<6) -#define S3C_PCM_CTL_RXDMA_EN (0x1<<5) -#define S3C_PCM_CTL_TXMSB_AFTER_FSYNC (0x1<<4) -#define S3C_PCM_CTL_RXMSB_AFTER_FSYNC (0x1<<3) -#define S3C_PCM_CTL_TXFIFO_EN (0x1<<2) -#define S3C_PCM_CTL_RXFIFO_EN (0x1<<1) -#define S3C_PCM_CTL_ENABLE (0x1<<0) - -/* PCM_CLKCTL Bit-Fields */ -#define S3C_PCM_CLKCTL_SERCLK_EN (0x1<<19) -#define S3C_PCM_CLKCTL_SERCLKSEL_PCLK (0x1<<18) -#define S3C_PCM_CLKCTL_SCLKDIV_MASK (0x1ff) -#define S3C_PCM_CLKCTL_SYNCDIV_MASK (0x1ff) -#define S3C_PCM_CLKCTL_SCLKDIV_SHIFT (9) -#define S3C_PCM_CLKCTL_SYNCDIV_SHIFT (0) - -/* PCM_TXFIFO Bit-Fields */ -#define S3C_PCM_TXFIFO_DVALID (0x1<<16) -#define S3C_PCM_TXFIFO_DATA_MSK (0xffff<<0) - -/* PCM_RXFIFO Bit-Fields */ -#define S3C_PCM_RXFIFO_DVALID (0x1<<16) -#define S3C_PCM_RXFIFO_DATA_MSK (0xffff<<0) - -/* PCM_IRQCTL Bit-Fields */ -#define S3C_PCM_IRQCTL_IRQEN (0x1<<14) -#define S3C_PCM_IRQCTL_WRDEN (0x1<<12) -#define S3C_PCM_IRQCTL_TXEMPTYEN (0x1<<11) -#define S3C_PCM_IRQCTL_TXALMSTEMPTYEN (0x1<<10) -#define S3C_PCM_IRQCTL_TXFULLEN (0x1<<9) -#define S3C_PCM_IRQCTL_TXALMSTFULLEN (0x1<<8) -#define S3C_PCM_IRQCTL_TXSTARVEN (0x1<<7) -#define S3C_PCM_IRQCTL_TXERROVRFLEN (0x1<<6) -#define S3C_PCM_IRQCTL_RXEMPTEN (0x1<<5) -#define S3C_PCM_IRQCTL_RXALMSTEMPTEN (0x1<<4) -#define S3C_PCM_IRQCTL_RXFULLEN (0x1<<3) -#define S3C_PCM_IRQCTL_RXALMSTFULLEN (0x1<<2) -#define S3C_PCM_IRQCTL_RXSTARVEN (0x1<<1) -#define S3C_PCM_IRQCTL_RXERROVRFLEN (0x1<<0) - -/* PCM_IRQSTAT Bit-Fields */ -#define S3C_PCM_IRQSTAT_IRQPND (0x1<<13) -#define S3C_PCM_IRQSTAT_WRD_XFER (0x1<<12) -#define S3C_PCM_IRQSTAT_TXEMPTY (0x1<<11) -#define S3C_PCM_IRQSTAT_TXALMSTEMPTY (0x1<<10) -#define S3C_PCM_IRQSTAT_TXFULL (0x1<<9) -#define S3C_PCM_IRQSTAT_TXALMSTFULL (0x1<<8) -#define S3C_PCM_IRQSTAT_TXSTARV (0x1<<7) -#define S3C_PCM_IRQSTAT_TXERROVRFL (0x1<<6) -#define S3C_PCM_IRQSTAT_RXEMPT (0x1<<5) -#define S3C_PCM_IRQSTAT_RXALMSTEMPT (0x1<<4) -#define S3C_PCM_IRQSTAT_RXFULL (0x1<<3) -#define S3C_PCM_IRQSTAT_RXALMSTFULL (0x1<<2) -#define S3C_PCM_IRQSTAT_RXSTARV (0x1<<1) -#define S3C_PCM_IRQSTAT_RXERROVRFL (0x1<<0) - -/* PCM_FIFOSTAT Bit-Fields */ -#define S3C_PCM_FIFOSTAT_TXCNT_MSK (0x3f<<14) -#define S3C_PCM_FIFOSTAT_TXFIFOEMPTY (0x1<<13) -#define S3C_PCM_FIFOSTAT_TXFIFOALMSTEMPTY (0x1<<12) -#define S3C_PCM_FIFOSTAT_TXFIFOFULL (0x1<<11) -#define S3C_PCM_FIFOSTAT_TXFIFOALMSTFULL (0x1<<10) -#define S3C_PCM_FIFOSTAT_RXCNT_MSK (0x3f<<4) -#define S3C_PCM_FIFOSTAT_RXFIFOEMPTY (0x1<<3) -#define S3C_PCM_FIFOSTAT_RXFIFOALMSTEMPTY (0x1<<2) -#define S3C_PCM_FIFOSTAT_RXFIFOFULL (0x1<<1) -#define S3C_PCM_FIFOSTAT_RXFIFOALMSTFULL (0x1<<0) - -#define S3C_PCM_CLKSRC_PCLK 0 -#define S3C_PCM_CLKSRC_MUX 1 - -#define S3C_PCM_SCLK_PER_FS 0 - -/** - * struct s3c_pcm_info - S3C PCM Controller information - * @dev: The parent device passed to use from the probe. - * @regs: The pointer to the device register block. - * @dma_playback: DMA information for playback channel. - * @dma_capture: DMA information for capture channel. - */ -struct s3c_pcm_info { - spinlock_t lock; - struct device *dev; - void __iomem *regs; - - unsigned int sclk_per_fs; - - /* Whether to keep PCMSCLK enabled even when idle(no active xfer) */ - unsigned int idleclk; - - struct clk *pclk; - struct clk *cclk; - - struct s3c_dma_params *dma_playback; - struct s3c_dma_params *dma_capture; -}; - -#endif /* __S3C_PCM_H */ diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c deleted file mode 100644 index 709adef..0000000 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ /dev/null @@ -1,190 +0,0 @@ -/* sound/soc/s3c24xx/s3c2412-i2s.c - * - * ALSA Soc Audio Layer - S3C2412 I2S driver - * - * Copyright (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory(a)wolfsonmicro.com - * linux(a)wolfsonmicro.com - * - * Copyright (c) 2007, 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/gpio.h> -#include <linux/clk.h> -#include <linux/kernel.h> -#include <linux/io.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> -#include <mach/hardware.h> - -#include <mach/regs-gpio.h> -#include <mach/dma.h> - -#include "s3c-dma.h" -#include "regs-i2s-v2.h" -#include "s3c2412-i2s.h" - -#define S3C2412_I2S_DEBUG 0 - -static struct s3c2410_dma_client s3c2412_dma_client_out = { - .name = "I2S PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c2412_dma_client_in = { - .name = "I2S PCM Stereo in" -}; - -static struct s3c_dma_params s3c2412_i2s_pcm_stereo_out = { - .client = &s3c2412_dma_client_out, - .channel = DMACH_I2S_OUT, - .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD, - .dma_size = 4, -}; - -static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = { - .client = &s3c2412_dma_client_in, - .channel = DMACH_I2S_IN, - .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD, - .dma_size = 4, -}; - -static struct s3c_i2sv2_info s3c2412_i2s; - -static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) -{ - return cpu_dai->private_data; -} - -static int s3c2412_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - int ret; - - pr_debug("Entered %s\n", __func__); - - ret = s3c_i2sv2_probe(pdev, dai, &s3c2412_i2s, S3C2410_PA_IIS); - if (ret) - return ret; - - s3c2412_i2s.dma_capture = &s3c2412_i2s_pcm_stereo_in; - s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out; - - s3c2412_i2s.iis_cclk = clk_get(&pdev->dev, "i2sclk"); - if (s3c2412_i2s.iis_cclk == NULL) { - pr_err("failed to get i2sclk clock\n"); - iounmap(s3c2412_i2s.regs); - return -ENODEV; - } - - /* Set MPLL as the source for IIS CLK */ - - clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll")); - clk_enable(s3c2412_i2s.iis_cclk); - - s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk; - - /* Configure the I2S pins in correct mode */ - s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); - s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); - s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); - - return 0; -} - -static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - struct s3c_dma_params *dma_data; - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = i2s->dma_playback; - else - dma_data = i2s->dma_capture; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - iismod = readl(i2s->regs + S3C2412_IISMOD); - pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); - - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - iismod |= S3C2412_IISMOD_8BIT; - break; - case SNDRV_PCM_FORMAT_S16_LE: - iismod &= ~S3C2412_IISMOD_8BIT; - break; - } - - writel(iismod, i2s->regs + S3C2412_IISMOD); - pr_debug("%s: w: IISMOD: %x\n", __func__, iismod); - - return 0; -} - -#define S3C2412_I2S_RATES \ - (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) - -static struct snd_soc_dai_ops s3c2412_i2s_dai_ops = { - .hw_params = s3c2412_i2s_hw_params, -}; - -struct snd_soc_dai s3c2412_i2s_dai = { - .name = "s3c2412-i2s", - .id = 0, - .probe = s3c2412_i2s_probe, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C2412_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C2412_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, - }, - .ops = &s3c2412_i2s_dai_ops, -}; -EXPORT_SYMBOL_GPL(s3c2412_i2s_dai); - -static int __init s3c2412_i2s_init(void) -{ - return s3c_i2sv2_register_dai(&s3c2412_i2s_dai); -} -module_init(s3c2412_i2s_init); - -static void __exit s3c2412_i2s_exit(void) -{ - snd_soc_unregister_dai(&s3c2412_i2s_dai); -} -module_exit(s3c2412_i2s_exit); - -/* Module information */ -MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("S3C2412 I2S SoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c2412-i2s.h b/sound/soc/s3c24xx/s3c2412-i2s.h deleted file mode 100644 index 0b5686b..0000000 --- a/sound/soc/s3c24xx/s3c2412-i2s.h +++ /dev/null @@ -1,29 +0,0 @@ -/* sound/soc/s3c24xx/s3c2412-i2s.c - * - * ALSA Soc Audio Layer - S3C2412 I2S driver - * - * Copyright (c) 2007 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. -*/ - -#ifndef __SND_SOC_S3C24XX_S3C2412_I2S_H -#define __SND_SOC_S3C24XX_S3C2412_I2S_H __FILE__ - -#include "s3c-i2s-v2.h" - -#define S3C2412_DIV_BCLK S3C_I2SV2_DIV_BCLK -#define S3C2412_DIV_RCLK S3C_I2SV2_DIV_RCLK -#define S3C2412_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER - -#define S3C2412_CLKSRC_PCLK S3C_I2SV2_CLKSRC_PCLK -#define S3C2412_CLKSRC_I2SCLK S3C_I2SV2_CLKSRC_AUDIOBUS - -extern struct snd_soc_dai s3c2412_i2s_dai; - -#endif /* __SND_SOC_S3C24XX_S3C2412_I2S_H */ diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c deleted file mode 100644 index c3ac890..0000000 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ /dev/null @@ -1,503 +0,0 @@ -/* - * s3c24xx-i2s.c -- ALSA Soc Audio Layer - * - * (c) 2006 Wolfson Microelectronics PLC. - * Graeme Gregory graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com - * - * Copyright 2004-2005 Simtec Electronics - * http://armlinux.simtec.co.uk/ - * Ben Dooks <ben(a)simtec.co.uk> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/clk.h> -#include <linux/jiffies.h> -#include <linux/io.h> -#include <linux/gpio.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include <mach/hardware.h> -#include <mach/regs-gpio.h> -#include <mach/regs-clock.h> - -#include <asm/dma.h> -#include <mach/dma.h> - -#include <plat/regs-iis.h> - -#include "s3c-dma.h" -#include "s3c24xx-i2s.h" - -static struct s3c2410_dma_client s3c24xx_dma_client_out = { - .name = "I2S PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c24xx_dma_client_in = { - .name = "I2S PCM Stereo in" -}; - -static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_out = { - .client = &s3c24xx_dma_client_out, - .channel = DMACH_I2S_OUT, - .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, - .dma_size = 2, -}; - -static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_in = { - .client = &s3c24xx_dma_client_in, - .channel = DMACH_I2S_IN, - .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, - .dma_size = 2, -}; - -struct s3c24xx_i2s_info { - void __iomem *regs; - struct clk *iis_clk; - u32 iiscon; - u32 iismod; - u32 iisfcon; - u32 iispsr; -}; -static struct s3c24xx_i2s_info s3c24xx_i2s; - -static void s3c24xx_snd_txctrl(int on) -{ - u32 iisfcon; - u32 iiscon; - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); - iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); - iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - - pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); - - if (on) { - iisfcon |= S3C2410_IISFCON_TXDMA | S3C2410_IISFCON_TXENABLE; - iiscon |= S3C2410_IISCON_TXDMAEN | S3C2410_IISCON_IISEN; - iiscon &= ~S3C2410_IISCON_TXIDLE; - iismod |= S3C2410_IISMOD_TXMODE; - - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); - writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); - } else { - /* note, we have to disable the FIFOs otherwise bad things - * seem to happen when the DMA stops. According to the - * Samsung supplied kernel, this should allow the DMA - * engine and FIFOs to reset. If this isn't allowed, the - * DMA engine will simply freeze randomly. - */ - - iisfcon &= ~S3C2410_IISFCON_TXENABLE; - iisfcon &= ~S3C2410_IISFCON_TXDMA; - iiscon |= S3C2410_IISCON_TXIDLE; - iiscon &= ~S3C2410_IISCON_TXDMAEN; - iismod &= ~S3C2410_IISMOD_TXMODE; - - writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); - writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - } - - pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); -} - -static void s3c24xx_snd_rxctrl(int on) -{ - u32 iisfcon; - u32 iiscon; - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); - iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); - iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - - pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); - - if (on) { - iisfcon |= S3C2410_IISFCON_RXDMA | S3C2410_IISFCON_RXENABLE; - iiscon |= S3C2410_IISCON_RXDMAEN | S3C2410_IISCON_IISEN; - iiscon &= ~S3C2410_IISCON_RXIDLE; - iismod |= S3C2410_IISMOD_RXMODE; - - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); - writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); - } else { - /* note, we have to disable the FIFOs otherwise bad things - * seem to happen when the DMA stops. According to the - * Samsung supplied kernel, this should allow the DMA - * engine and FIFOs to reset. If this isn't allowed, the - * DMA engine will simply freeze randomly. - */ - - iisfcon &= ~S3C2410_IISFCON_RXENABLE; - iisfcon &= ~S3C2410_IISFCON_RXDMA; - iiscon |= S3C2410_IISCON_RXIDLE; - iiscon &= ~S3C2410_IISCON_RXDMAEN; - iismod &= ~S3C2410_IISMOD_RXMODE; - - writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); - writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - } - - pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); -} - -/* - * Wait for the LR signal to allow synchronisation to the L/R clock - * from the codec. May only be needed for slave mode. - */ -static int s3c24xx_snd_lrsync(void) -{ - u32 iiscon; - int timeout = 50; /* 5ms */ - - pr_debug("Entered %s\n", __func__); - - while (1) { - iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); - if (iiscon & S3C2410_IISCON_LRINDEX) - break; - - if (!timeout--) - return -ETIMEDOUT; - udelay(100); - } - - return 0; -} - -/* - * Check whether CPU is the master or slave - */ -static inline int s3c24xx_snd_is_clkmaster(void) -{ - pr_debug("Entered %s\n", __func__); - - return (readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & S3C2410_IISMOD_SLAVE) ? 0:1; -} - -/* - * Set S3C24xx I2S DAI format - */ -static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("hw_params r: IISMOD: %x \n", iismod); - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - iismod |= S3C2410_IISMOD_SLAVE; - break; - case SND_SOC_DAIFMT_CBS_CFS: - iismod &= ~S3C2410_IISMOD_SLAVE; - break; - default: - return -EINVAL; - } - - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_LEFT_J: - iismod |= S3C2410_IISMOD_MSB; - break; - case SND_SOC_DAIFMT_I2S: - iismod &= ~S3C2410_IISMOD_MSB; - break; - default: - return -EINVAL; - } - - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("hw_params w: IISMOD: %x \n", iismod); - return 0; -} - -static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data; - u32 iismod; - - pr_debug("Entered %s\n", __func__); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c24xx_i2s_pcm_stereo_out; - else - dma_data = &s3c24xx_i2s_pcm_stereo_in; - - snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_data); - - /* Working copies of register */ - iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("hw_params r: IISMOD: %x\n", iismod); - - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - iismod &= ~S3C2410_IISMOD_16BIT; - dma_data->dma_size = 1; - break; - case SNDRV_PCM_FORMAT_S16_LE: - iismod |= S3C2410_IISMOD_16BIT; - dma_data->dma_size = 2; - break; - default: - return -EINVAL; - } - - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - pr_debug("hw_params w: IISMOD: %x\n", iismod); - return 0; -} - -static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - int ret = 0; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); - - pr_debug("Entered %s\n", __func__); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!s3c24xx_snd_is_clkmaster()) { - ret = s3c24xx_snd_lrsync(); - if (ret) - goto exit_err; - } - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - s3c24xx_snd_rxctrl(1); - else - s3c24xx_snd_txctrl(1); - - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - s3c24xx_snd_rxctrl(0); - else - s3c24xx_snd_txctrl(0); - break; - default: - ret = -EINVAL; - break; - } - -exit_err: - return ret; -} - -/* - * Set S3C24xx Clock source - */ -static int s3c24xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, - int clk_id, unsigned int freq, int dir) -{ - u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - - pr_debug("Entered %s\n", __func__); - - iismod &= ~S3C2440_IISMOD_MPLL; - - switch (clk_id) { - case S3C24XX_CLKSRC_PCLK: - break; - case S3C24XX_CLKSRC_MPLL: - iismod |= S3C2440_IISMOD_MPLL; - break; - default: - return -EINVAL; - } - - writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - return 0; -} - -/* - * Set S3C24xx Clock dividers - */ -static int s3c24xx_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, - int div_id, int div) -{ - u32 reg; - - pr_debug("Entered %s\n", __func__); - - switch (div_id) { - case S3C24XX_DIV_BCLK: - reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~S3C2410_IISMOD_FS_MASK; - writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); - break; - case S3C24XX_DIV_MCLK: - reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~(S3C2410_IISMOD_384FS); - writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); - break; - case S3C24XX_DIV_PRESCALER: - writel(div, s3c24xx_i2s.regs + S3C2410_IISPSR); - reg = readl(s3c24xx_i2s.regs + S3C2410_IISCON); - writel(reg | S3C2410_IISCON_PSCEN, s3c24xx_i2s.regs + S3C2410_IISCON); - break; - default: - return -EINVAL; - } - - return 0; -} - -/* - * To avoid duplicating clock code, allow machine driver to - * get the clockrate from here. - */ -u32 s3c24xx_i2s_get_clockrate(void) -{ - return clk_get_rate(s3c24xx_i2s.iis_clk); -} -EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); - -static int s3c24xx_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - pr_debug("Entered %s\n", __func__); - - s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); - if (s3c24xx_i2s.regs == NULL) - return -ENXIO; - - s3c24xx_i2s.iis_clk = clk_get(&pdev->dev, "iis"); - if (s3c24xx_i2s.iis_clk == NULL) { - pr_err("failed to get iis_clock\n"); - iounmap(s3c24xx_i2s.regs); - return -ENODEV; - } - clk_enable(s3c24xx_i2s.iis_clk); - - /* Configure the I2S pins in correct mode */ - s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); - s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); - s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); - - writel(S3C2410_IISCON_IISEN, s3c24xx_i2s.regs + S3C2410_IISCON); - - s3c24xx_snd_txctrl(0); - s3c24xx_snd_rxctrl(0); - - return 0; -} - -#ifdef CONFIG_PM -static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai) -{ - pr_debug("Entered %s\n", __func__); - - s3c24xx_i2s.iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); - s3c24xx_i2s.iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - s3c24xx_i2s.iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); - s3c24xx_i2s.iispsr = readl(s3c24xx_i2s.regs + S3C2410_IISPSR); - - clk_disable(s3c24xx_i2s.iis_clk); - - return 0; -} - -static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai) -{ - pr_debug("Entered %s\n", __func__); - clk_enable(s3c24xx_i2s.iis_clk); - - writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); - writel(s3c24xx_i2s.iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); - writel(s3c24xx_i2s.iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); - writel(s3c24xx_i2s.iispsr, s3c24xx_i2s.regs + S3C2410_IISPSR); - - return 0; -} -#else -#define s3c24xx_i2s_suspend NULL -#define s3c24xx_i2s_resume NULL -#endif - - -#define S3C24XX_I2S_RATES \ - (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) - -static struct snd_soc_dai_ops s3c24xx_i2s_dai_ops = { - .trigger = s3c24xx_i2s_trigger, - .hw_params = s3c24xx_i2s_hw_params, - .set_fmt = s3c24xx_i2s_set_fmt, - .set_clkdiv = s3c24xx_i2s_set_clkdiv, - .set_sysclk = s3c24xx_i2s_set_sysclk, -}; - -struct snd_soc_dai s3c24xx_i2s_dai = { - .name = "s3c24xx-i2s", - .id = 0, - .probe = s3c24xx_i2s_probe, - .suspend = s3c24xx_i2s_suspend, - .resume = s3c24xx_i2s_resume, - .playback = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C24XX_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,}, - .capture = { - .channels_min = 2, - .channels_max = 2, - .rates = S3C24XX_I2S_RATES, - .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &s3c24xx_i2s_dai_ops, -}; -EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai); - -static int __init s3c24xx_i2s_init(void) -{ - return snd_soc_register_dai(&s3c24xx_i2s_dai); -} -module_init(s3c24xx_i2s_init); - -static void __exit s3c24xx_i2s_exit(void) -{ - snd_soc_unregister_dai(&s3c24xx_i2s_dai); -} -module_exit(s3c24xx_i2s_exit); - -/* Module information */ -MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("s3c24xx I2S SoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.h b/sound/soc/s3c24xx/s3c24xx-i2s.h deleted file mode 100644 index 726d91c..0000000 --- a/sound/soc/s3c24xx/s3c24xx-i2s.h +++ /dev/null @@ -1,37 +0,0 @@ -/* - * s3c24xx-i2s.c -- ALSA Soc Audio Layer - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - * Revision history - * 10th Nov 2006 Initial version. - */ - -#ifndef S3C24XXI2S_H_ -#define S3C24XXI2S_H_ - -/* clock sources */ -#define S3C24XX_CLKSRC_PCLK 0 -#define S3C24XX_CLKSRC_MPLL 1 - -/* Clock dividers */ -#define S3C24XX_DIV_MCLK 0 -#define S3C24XX_DIV_BCLK 1 -#define S3C24XX_DIV_PRESCALER 2 - -/* prescaler */ -#define S3C24XX_PRESCALE(a,b) \ - (((a - 1) << S3C2410_IISPSR_INTSHIFT) | ((b - 1) << S3C2410_IISPSR_EXTSHFIT)) - -u32 s3c24xx_i2s_get_clockrate(void); - -extern struct snd_soc_dai s3c24xx_i2s_dai; - -#endif /*S3C24XXI2S_H_*/ diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c deleted file mode 100644 index 4984754..0000000 --- a/sound/soc/s3c24xx/s3c24xx_simtec.c +++ /dev/null @@ -1,394 +0,0 @@ -/* sound/soc/s3c24xx/s3c24xx_simtec.c - * - * Copyright 2009 Simtec Electronics - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. -*/ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/platform_device.h> -#include <linux/gpio.h> -#include <linux/clk.h> -#include <linux/i2c.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <plat/audio-simtec.h> - -#include "s3c-dma.h" -#include "s3c24xx-i2s.h" -#include "s3c24xx_simtec.h" - -static struct s3c24xx_audio_simtec_pdata *pdata; -static struct clk *xtal_clk; - -static int spk_gain; -static int spk_unmute; - -/** - * speaker_gain_get - read the speaker gain setting. - * @kcontrol: The control for the speaker gain. - * @ucontrol: The value that needs to be updated. - * - * Read the value for the AMP gain control. - */ -static int speaker_gain_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = spk_gain; - return 0; -} - -/** - * speaker_gain_set - set the value of the speaker amp gain - * @value: The value to write. - */ -static void speaker_gain_set(int value) -{ - gpio_set_value_cansleep(pdata->amp_gain[0], value & 1); - gpio_set_value_cansleep(pdata->amp_gain[1], value >> 1); -} - -/** - * speaker_gain_put - set the speaker gain setting. - * @kcontrol: The control for the speaker gain. - * @ucontrol: The value that needs to be set. - * - * Set the value of the speaker gain from the specified - * @ucontrol setting. - * - * Note, if the speaker amp is muted, then we do not set a gain value - * as at-least one of the ICs that is fitted will try and power up even - * if the main control is set to off. - */ -static int speaker_gain_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - int value = ucontrol->value.integer.value[0]; - - spk_gain = value; - - if (!spk_unmute) - speaker_gain_set(value); - - return 0; -} - -static const struct snd_kcontrol_new amp_gain_controls[] = { - SOC_SINGLE_EXT("Speaker Gain", 0, 0, 3, 0, - speaker_gain_get, speaker_gain_put), -}; - -/** - * spk_unmute_state - set the unmute state of the speaker - * @to: zero to unmute, non-zero to ununmute. - */ -static void spk_unmute_state(int to) -{ - pr_debug("%s: to=%d\n", __func__, to); - - spk_unmute = to; - gpio_set_value(pdata->amp_gpio, to); - - /* if we're umuting, also re-set the gain */ - if (to && pdata->amp_gain[0] > 0) - speaker_gain_set(spk_gain); -} - -/** - * speaker_unmute_get - read the speaker unmute setting. - * @kcontrol: The control for the speaker gain. - * @ucontrol: The value that needs to be updated. - * - * Read the value for the AMP gain control. - */ -static int speaker_unmute_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = spk_unmute; - return 0; -} - -/** - * speaker_unmute_put - set the speaker unmute setting. - * @kcontrol: The control for the speaker gain. - * @ucontrol: The value that needs to be set. - * - * Set the value of the speaker gain from the specified - * @ucontrol setting. - */ -static int speaker_unmute_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - spk_unmute_state(ucontrol->value.integer.value[0]); - return 0; -} - -/* This is added as a manual control as the speaker amps create clicks - * when their power state is changed, which are far more noticeable than - * anything produced by the CODEC itself. - */ -static const struct snd_kcontrol_new amp_unmute_controls[] = { - SOC_SINGLE_EXT("Speaker Switch", 0, 0, 1, 0, - speaker_unmute_get, speaker_unmute_put), -}; - -void simtec_audio_init(struct snd_soc_codec *codec) -{ - if (pdata->amp_gpio > 0) { - pr_debug("%s: adding amp routes\n", __func__); - - snd_soc_add_controls(codec, amp_unmute_controls, - ARRAY_SIZE(amp_unmute_controls)); - } - - if (pdata->amp_gain[0] > 0) { - pr_debug("%s: adding amp controls\n", __func__); - snd_soc_add_controls(codec, amp_gain_controls, - ARRAY_SIZE(amp_gain_controls)); - } -} -EXPORT_SYMBOL_GPL(simtec_audio_init); - -#define CODEC_CLOCK 12000000 - -/** - * simtec_hw_params - update hardware parameters - * @substream: The audio substream instance. - * @params: The parameters requested. - * - * Update the codec data routing and configuration settings - * from the supplied data. - */ -static int simtec_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - int ret; - - /* Set the CODEC as the bus clock master, I2S */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - pr_err("%s: failed set cpu dai format\n", __func__); - return ret; - } - - /* Set the CODEC as the bus clock master */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM); - if (ret) { - pr_err("%s: failed set codec dai format\n", __func__); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, 0, - CODEC_CLOCK, SND_SOC_CLOCK_IN); - if (ret) { - pr_err( "%s: failed setting codec sysclk\n", __func__); - return ret; - } - - if (pdata->use_mpllin) { - ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_MPLL, - 0, SND_SOC_CLOCK_OUT); - - if (ret) { - pr_err("%s: failed to set MPLLin as clksrc\n", - __func__); - return ret; - } - } - - if (pdata->output_cdclk) { - int cdclk_scale; - - cdclk_scale = clk_get_rate(xtal_clk) / CODEC_CLOCK; - cdclk_scale--; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, - cdclk_scale); - } - - return 0; -} - -static int simtec_call_startup(struct s3c24xx_audio_simtec_pdata *pd) -{ - /* call any board supplied startup code, this currently only - * covers the bast/vr1000 which have a CPLD in the way of the - * LRCLK */ - if (pd->startup) - pd->startup(); - - return 0; -} - -static struct snd_soc_ops simtec_snd_ops = { - .hw_params = simtec_hw_params, -}; - -/** - * attach_gpio_amp - get and configure the necessary gpios - * @dev: The device we're probing. - * @pd: The platform data supplied by the board. - * - * If there is a GPIO based amplifier attached to the board, claim - * the necessary GPIO lines for it, and set default values. - */ -static int attach_gpio_amp(struct device *dev, - struct s3c24xx_audio_simtec_pdata *pd) -{ - int ret; - - /* attach gpio amp gain (if any) */ - if (pdata->amp_gain[0] > 0) { - ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain0"); - if (ret) { - dev_err(dev, "cannot get amp gpio gain0\n"); - return ret; - } - - ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain1"); - if (ret) { - dev_err(dev, "cannot get amp gpio gain1\n"); - gpio_free(pdata->amp_gain[0]); - return ret; - } - - gpio_direction_output(pd->amp_gain[0], 0); - gpio_direction_output(pd->amp_gain[1], 0); - } - - /* note, currently we assume GPA0 isn't valid amp */ - if (pdata->amp_gpio > 0) { - ret = gpio_request(pd->amp_gpio, "gpio-amp"); - if (ret) { - dev_err(dev, "cannot get amp gpio %d (%d)\n", - pd->amp_gpio, ret); - goto err_amp; - } - - /* set the amp off at startup */ - spk_unmute_state(0); - } - - return 0; - -err_amp: - if (pd->amp_gain[0] > 0) { - gpio_free(pd->amp_gain[0]); - gpio_free(pd->amp_gain[1]); - } - - return ret; -} - -static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd) -{ - if (pd->amp_gain[0] > 0) { - gpio_free(pd->amp_gain[0]); - gpio_free(pd->amp_gain[1]); - } - - if (pd->amp_gpio > 0) - gpio_free(pd->amp_gpio); -} - -#ifdef CONFIG_PM -int simtec_audio_resume(struct device *dev) -{ - simtec_call_startup(pdata); - return 0; -} - -const struct dev_pm_ops simtec_audio_pmops = { - .resume = simtec_audio_resume, -}; -EXPORT_SYMBOL_GPL(simtec_audio_pmops); -#endif - -int __devinit simtec_audio_core_probe(struct platform_device *pdev, - struct snd_soc_device *socdev) -{ - struct platform_device *snd_dev; - int ret; - - socdev->card->dai_link->ops = &simtec_snd_ops; - - pdata = pdev->dev.platform_data; - if (!pdata) { - dev_err(&pdev->dev, "no platform data supplied\n"); - return -EINVAL; - } - - simtec_call_startup(pdata); - - xtal_clk = clk_get(&pdev->dev, "xtal"); - if (IS_ERR(xtal_clk)) { - dev_err(&pdev->dev, "could not get clkout0\n"); - return -EINVAL; - } - - dev_info(&pdev->dev, "xtal rate is %ld\n", clk_get_rate(xtal_clk)); - - ret = attach_gpio_amp(&pdev->dev, pdata); - if (ret) - goto err_clk; - - snd_dev = platform_device_alloc("soc-audio", -1); - if (!snd_dev) { - dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n"); - ret = -ENOMEM; - goto err_gpio; - } - - platform_set_drvdata(snd_dev, socdev); - socdev->dev = &snd_dev->dev; - - ret = platform_device_add(snd_dev); - if (ret) { - dev_err(&pdev->dev, "failed to add soc-audio dev\n"); - goto err_pdev; - } - - platform_set_drvdata(pdev, snd_dev); - return 0; - -err_pdev: - platform_device_put(snd_dev); - -err_gpio: - detach_gpio_amp(pdata); - -err_clk: - clk_put(xtal_clk); - return ret; -} -EXPORT_SYMBOL_GPL(simtec_audio_core_probe); - -int __devexit simtec_audio_remove(struct platform_device *pdev) -{ - struct platform_device *snd_dev = platform_get_drvdata(pdev); - - platform_device_unregister(snd_dev); - - detach_gpio_amp(pdata); - clk_put(xtal_clk); - return 0; -} -EXPORT_SYMBOL_GPL(simtec_audio_remove); - -MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("ALSA SoC Simtec Audio common support"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.h b/sound/soc/s3c24xx/s3c24xx_simtec.h deleted file mode 100644 index e18faee..0000000 --- a/sound/soc/s3c24xx/s3c24xx_simtec.h +++ /dev/null @@ -1,22 +0,0 @@ -/* sound/soc/s3c24xx/s3c24xx_simtec.h - * - * Copyright 2009 Simtec Electronics - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. -*/ - -extern void simtec_audio_init(struct snd_soc_codec *codec); - -extern int simtec_audio_core_probe(struct platform_device *pdev, - struct snd_soc_device *socdev); - -extern int simtec_audio_remove(struct platform_device *pdev); - -#ifdef CONFIG_PM -extern const struct dev_pm_ops simtec_audio_pmops; -#define simtec_audio_pm &simtec_audio_pmops -#else -#define simtec_audio_pm NULL -#endif diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c deleted file mode 100644 index bdf8951..0000000 --- a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c +++ /dev/null @@ -1,153 +0,0 @@ -/* sound/soc/s3c24xx/s3c24xx_simtec_hermes.c - * - * Copyright 2009 Simtec Electronics - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. -*/ - -#include <linux/module.h> -#include <linux/clk.h> -#include <linux/platform_device.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <plat/audio-simtec.h> - -#include "s3c-dma.h" -#include "s3c24xx-i2s.h" -#include "s3c24xx_simtec.h" - -#include "../codecs/tlv320aic3x.h" - -static const struct snd_soc_dapm_widget dapm_widgets[] = { - SND_SOC_DAPM_LINE("GSM Out", NULL), - SND_SOC_DAPM_LINE("GSM In", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), - SND_SOC_DAPM_LINE("Line Out", NULL), - SND_SOC_DAPM_LINE("ZV", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), - SND_SOC_DAPM_HP("Headphone Jack", NULL), -}; - -static const struct snd_soc_dapm_route base_map[] = { - /* Headphone connected to HP{L,R}OUT and HP{L,R}COM */ - - { "Headphone Jack", NULL, "HPLOUT" }, - { "Headphone Jack", NULL, "HPLCOM" }, - { "Headphone Jack", NULL, "HPROUT" }, - { "Headphone Jack", NULL, "HPRCOM" }, - - /* ZV connected to Line1 */ - - { "LINE1L", NULL, "ZV" }, - { "LINE1R", NULL, "ZV" }, - - /* Line In connected to Line2 */ - - { "LINE2L", NULL, "Line In" }, - { "LINE2R", NULL, "Line In" }, - - /* Microphone connected to MIC3R and MIC_BIAS */ - - { "MIC3L", NULL, "Mic Jack" }, - - /* GSM connected to MONO_LOUT and MIC3L (in) */ - - { "GSM Out", NULL, "MONO_LOUT" }, - { "MIC3L", NULL, "GSM In" }, - - /* Speaker is connected to LINEOUT{LN,LP,RN,RP}, however we are - * not using the DAPM to power it up and down as there it makes - * a click when powering up. */ -}; - -/** - * simtec_hermes_init - initialise and add controls - * @codec; The codec instance to attach to. - * - * Attach our controls and configure the necessary codec - * mappings for our sound card instance. -*/ -static int simtec_hermes_init(struct snd_soc_codec *codec) -{ - snd_soc_dapm_new_controls(codec, dapm_widgets, - ARRAY_SIZE(dapm_widgets)); - - snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); - - snd_soc_dapm_enable_pin(codec, "Headphone Jack"); - snd_soc_dapm_enable_pin(codec, "Line In"); - snd_soc_dapm_enable_pin(codec, "Line Out"); - snd_soc_dapm_enable_pin(codec, "Mic Jack"); - - simtec_audio_init(codec); - snd_soc_dapm_sync(codec); - - return 0; -} - -static struct aic3x_setup_data codec_setup = { -}; - -static struct snd_soc_dai_link simtec_dai_aic33 = { - .name = "tlv320aic33", - .stream_name = "TLV320AIC33", - .cpu_dai = &s3c24xx_i2s_dai, - .codec_dai = &aic3x_dai, - .init = simtec_hermes_init, -}; - -/* simtec audio machine driver */ -static struct snd_soc_card snd_soc_machine_simtec_aic33 = { - .name = "Simtec-Hermes", - .platform = &s3c24xx_soc_platform, - .dai_link = &simtec_dai_aic33, - .num_links = 1, -}; - -/* simtec audio subsystem */ -static struct snd_soc_device simtec_snd_devdata_aic33 = { - .card = &snd_soc_machine_simtec_aic33, - .codec_dev = &soc_codec_dev_aic3x, - .codec_data = &codec_setup, -}; - -static int __devinit simtec_audio_hermes_probe(struct platform_device *pd) -{ - dev_info(&pd->dev, "probing....\n"); - return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic33); -} - -static struct platform_driver simtec_audio_hermes_platdrv = { - .driver = { - .owner = THIS_MODULE, - .name = "s3c24xx-simtec-hermes-snd", - .pm = simtec_audio_pm, - }, - .probe = simtec_audio_hermes_probe, - .remove = __devexit_p(simtec_audio_remove), -}; - -MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd"); - -static int __init simtec_hermes_modinit(void) -{ - return platform_driver_register(&simtec_audio_hermes_platdrv); -} - -static void __exit simtec_hermes_modexit(void) -{ - platform_driver_unregister(&simtec_audio_hermes_platdrv); -} - -module_init(simtec_hermes_modinit); -module_exit(simtec_hermes_modexit); - -MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c deleted file mode 100644 index 185c0ac..0000000 --- a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c +++ /dev/null @@ -1,137 +0,0 @@ -/* sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c - * - * Copyright 2009 Simtec Electronics - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. -*/ - -#include <linux/module.h> -#include <linux/clk.h> -#include <linux/platform_device.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include <plat/audio-simtec.h> - -#include "s3c-dma.h" -#include "s3c24xx-i2s.h" -#include "s3c24xx_simtec.h" - -#include "../codecs/tlv320aic23.h" - -/* supported machines: - * - * Machine Connections AMP - * ------- ----------- --- - * BAST MIC, HPOUT, LOUT, LIN TPA2001D1 (HPOUTL,R) (gain hardwired) - * VR1000 HPOUT, LIN None - * VR2000 LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) - * DePicture LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) - * Anubis LIN, LOUT, MIC, HP TPA2001D1 (HPOUTL,R) - */ - -static const struct snd_soc_dapm_widget dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), - SND_SOC_DAPM_LINE("Line Out", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), -}; - -static const struct snd_soc_dapm_route base_map[] = { - { "Headphone Jack", NULL, "LHPOUT"}, - { "Headphone Jack", NULL, "RHPOUT"}, - - { "Line Out", NULL, "LOUT" }, - { "Line Out", NULL, "ROUT" }, - - { "LLINEIN", NULL, "Line In"}, - { "RLINEIN", NULL, "Line In"}, - - { "MICIN", NULL, "Mic Jack"}, -}; - -/** - * simtec_tlv320aic23_init - initialise and add controls - * @codec; The codec instance to attach to. - * - * Attach our controls and configure the necessary codec - * mappings for our sound card instance. -*/ -static int simtec_tlv320aic23_init(struct snd_soc_codec *codec) -{ - snd_soc_dapm_new_controls(codec, dapm_widgets, - ARRAY_SIZE(dapm_widgets)); - - snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); - - snd_soc_dapm_enable_pin(codec, "Headphone Jack"); - snd_soc_dapm_enable_pin(codec, "Line In"); - snd_soc_dapm_enable_pin(codec, "Line Out"); - snd_soc_dapm_enable_pin(codec, "Mic Jack"); - - simtec_audio_init(codec); - snd_soc_dapm_sync(codec); - - return 0; -} - -static struct snd_soc_dai_link simtec_dai_aic23 = { - .name = "tlv320aic23", - .stream_name = "TLV320AIC23", - .cpu_dai = &s3c24xx_i2s_dai, - .codec_dai = &tlv320aic23_dai, - .init = simtec_tlv320aic23_init, -}; - -/* simtec audio machine driver */ -static struct snd_soc_card snd_soc_machine_simtec_aic23 = { - .name = "Simtec", - .platform = &s3c24xx_soc_platform, - .dai_link = &simtec_dai_aic23, - .num_links = 1, -}; - -/* simtec audio subsystem */ -static struct snd_soc_device simtec_snd_devdata_aic23 = { - .card = &snd_soc_machine_simtec_aic23, - .codec_dev = &soc_codec_dev_tlv320aic23, -}; - -static int __devinit simtec_audio_tlv320aic23_probe(struct platform_device *pd) -{ - return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic23); -} - -static struct platform_driver simtec_audio_tlv320aic23_platdrv = { - .driver = { - .owner = THIS_MODULE, - .name = "s3c24xx-simtec-tlv320aic23", - .pm = simtec_audio_pm, - }, - .probe = simtec_audio_tlv320aic23_probe, - .remove = __devexit_p(simtec_audio_remove), -}; - -MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23"); - -static int __init simtec_tlv320aic23_modinit(void) -{ - return platform_driver_register(&simtec_audio_tlv320aic23_platdrv); -} - -static void __exit simtec_tlv320aic23_modexit(void) -{ - platform_driver_unregister(&simtec_audio_tlv320aic23_platdrv); -} - -module_init(simtec_tlv320aic23_modinit); -module_exit(simtec_tlv320aic23_modexit); - -MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c deleted file mode 100644 index 052d596..0000000 --- a/sound/soc/s3c24xx/s3c24xx_uda134x.c +++ /dev/null @@ -1,373 +0,0 @@ -/* - * Modifications by Christian Pellegrin <chripell(a)evolware.org> - * - * s3c24xx_uda134x.c -- S3C24XX_UDA134X ALSA SoC Audio board driver - * - * Copyright 2007 Dension Audio Systems Ltd. - * Author: Zoltan Devai - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/module.h> -#include <linux/clk.h> -#include <linux/mutex.h> -#include <linux/gpio.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> -#include <sound/s3c24xx_uda134x.h> -#include <sound/uda134x.h> - -#include <plat/regs-iis.h> - -#include "s3c-dma.h" -#include "s3c24xx-i2s.h" -#include "../codecs/uda134x.h" - - -/* #define ENFORCE_RATES 1 */ -/* - Unfortunately the S3C24XX in master mode has a limited capacity of - generating the clock for the codec. If you define this only rates - that are really available will be enforced. But be careful, most - user level application just want the usual sampling frequencies (8, - 11.025, 22.050, 44.1 kHz) and anyway resampling is a costly - operation for embedded systems. So if you aren't very lucky or your - hardware engineer wasn't very forward-looking it's better to leave - this undefined. If you do so an approximate value for the requested - sampling rate in the range -/+ 5% will be chosen. If this in not - possible an error will be returned. -*/ - -static struct clk *xtal; -static struct clk *pclk; -/* this is need because we don't have a place where to keep the - * pointers to the clocks in each substream. We get the clocks only - * when we are actually using them so we don't block stuff like - * frequency change or oscillator power-off */ -static int clk_users; -static DEFINE_MUTEX(clk_lock); - -static unsigned int rates[33 * 2]; -#ifdef ENFORCE_RATES -static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; -#endif - -static struct platform_device *s3c24xx_uda134x_snd_device; - -static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) -{ - int ret = 0; -#ifdef ENFORCE_RATES - struct snd_pcm_runtime *runtime = substream->runtime; -#endif - - mutex_lock(&clk_lock); - pr_debug("%s %d\n", __func__, clk_users); - if (clk_users == 0) { - xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal"); - if (!xtal) { - printk(KERN_ERR "%s cannot get xtal\n", __func__); - ret = -EBUSY; - } else { - pclk = clk_get(&s3c24xx_uda134x_snd_device->dev, - "pclk"); - if (!pclk) { - printk(KERN_ERR "%s cannot get pclk\n", - __func__); - clk_put(xtal); - ret = -EBUSY; - } - } - if (!ret) { - int i, j; - - for (i = 0; i < 2; i++) { - int fs = i ? 256 : 384; - - rates[i*33] = clk_get_rate(xtal) / fs; - for (j = 1; j < 33; j++) - rates[i*33 + j] = clk_get_rate(pclk) / - (j * fs); - } - } - } - clk_users += 1; - mutex_unlock(&clk_lock); - if (!ret) { -#ifdef ENFORCE_RATES - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &hw_constraints_rates); - if (ret < 0) - printk(KERN_ERR "%s cannot set constraints\n", - __func__); -#endif - } - return ret; -} - -static void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream) -{ - mutex_lock(&clk_lock); - pr_debug("%s %d\n", __func__, clk_users); - clk_users -= 1; - if (clk_users == 0) { - clk_put(xtal); - xtal = NULL; - clk_put(pclk); - pclk = NULL; - } - mutex_unlock(&clk_lock); -} - -static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - unsigned int clk = 0; - int ret = 0; - int clk_source, fs_mode; - unsigned long rate = params_rate(params); - long err, cerr; - unsigned int div; - int i, bi; - - err = 999999; - bi = 0; - for (i = 0; i < 2*33; i++) { - cerr = rates[i] - rate; - if (cerr < 0) - cerr = -cerr; - if (cerr < err) { - err = cerr; - bi = i; - } - } - if (bi / 33 == 1) - fs_mode = S3C2410_IISMOD_256FS; - else - fs_mode = S3C2410_IISMOD_384FS; - if (bi % 33 == 0) { - clk_source = S3C24XX_CLKSRC_MPLL; - div = 1; - } else { - clk_source = S3C24XX_CLKSRC_PCLK; - div = bi % 33; - } - pr_debug("%s desired rate %lu, %d\n", __func__, rate, bi); - - clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate; - pr_debug("%s will use: %s %s %d sysclk %d err %ld\n", __func__, - fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS", - clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK", - div, clk, err); - - if ((err * 100 / rate) > 5) { - printk(KERN_ERR "S3C24XX_UDA134X: effective frequency " - "too different from desired (%ld%%)\n", - err * 100 / rate); - return -EINVAL; - } - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source , clk, - SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, fs_mode); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, - S3C2410_IISMOD_32FS); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, - S3C24XX_PRESCALE(div, div)); - if (ret < 0) - return ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, - SND_SOC_CLOCK_OUT); - if (ret < 0) - return ret; - - return 0; -} - -static struct snd_soc_ops s3c24xx_uda134x_ops = { - .startup = s3c24xx_uda134x_startup, - .shutdown = s3c24xx_uda134x_shutdown, - .hw_params = s3c24xx_uda134x_hw_params, -}; - -static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { - .name = "UDA134X", - .stream_name = "UDA134X", - .codec_dai = &uda134x_dai, - .cpu_dai = &s3c24xx_i2s_dai, - .ops = &s3c24xx_uda134x_ops, -}; - -static struct snd_soc_card snd_soc_s3c24xx_uda134x = { - .name = "S3C24XX_UDA134X", - .platform = &s3c24xx_soc_platform, - .dai_link = &s3c24xx_uda134x_dai_link, - .num_links = 1, -}; - -static struct s3c24xx_uda134x_platform_data *s3c24xx_uda134x_l3_pins; - -static void setdat(int v) -{ - gpio_set_value(s3c24xx_uda134x_l3_pins->l3_data, v > 0); -} - -static void setclk(int v) -{ - gpio_set_value(s3c24xx_uda134x_l3_pins->l3_clk, v > 0); -} - -static void setmode(int v) -{ - gpio_set_value(s3c24xx_uda134x_l3_pins->l3_mode, v > 0); -} - -static struct uda134x_platform_data s3c24xx_uda134x = { - .l3 = { - .setdat = setdat, - .setclk = setclk, - .setmode = setmode, - .data_hold = 1, - .data_setup = 1, - .clock_high = 1, - .mode_hold = 1, - .mode = 1, - .mode_setup = 1, - }, -}; - -static struct snd_soc_device s3c24xx_uda134x_snd_devdata = { - .card = &snd_soc_s3c24xx_uda134x, - .codec_dev = &soc_codec_dev_uda134x, - .codec_data = &s3c24xx_uda134x, -}; - -static int s3c24xx_uda134x_setup_pin(int pin, char *fun) -{ - if (gpio_request(pin, "s3c24xx_uda134x") < 0) { - printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " - "l3 %s pin already in use", fun); - return -EBUSY; - } - gpio_direction_output(pin, 0); - return 0; -} - -static int s3c24xx_uda134x_probe(struct platform_device *pdev) -{ - int ret; - - printk(KERN_INFO "S3C24XX_UDA134X SoC Audio driver\n"); - - s3c24xx_uda134x_l3_pins = pdev->dev.platform_data; - if (s3c24xx_uda134x_l3_pins == NULL) { - printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " - "unable to find platform data\n"); - return -ENODEV; - } - s3c24xx_uda134x.power = s3c24xx_uda134x_l3_pins->power; - s3c24xx_uda134x.model = s3c24xx_uda134x_l3_pins->model; - - if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_data, - "data") < 0) - return -EBUSY; - if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_clk, - "clk") < 0) { - gpio_free(s3c24xx_uda134x_l3_pins->l3_data); - return -EBUSY; - } - if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_mode, - "mode") < 0) { - gpio_free(s3c24xx_uda134x_l3_pins->l3_data); - gpio_free(s3c24xx_uda134x_l3_pins->l3_clk); - return -EBUSY; - } - - s3c24xx_uda134x_snd_device = platform_device_alloc("soc-audio", -1); - if (!s3c24xx_uda134x_snd_device) { - printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " - "Unable to register\n"); - return -ENOMEM; - } - - platform_set_drvdata(s3c24xx_uda134x_snd_device, - &s3c24xx_uda134x_snd_devdata); - s3c24xx_uda134x_snd_devdata.dev = &s3c24xx_uda134x_snd_device->dev; - ret = platform_device_add(s3c24xx_uda134x_snd_device); - if (ret) { - printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n"); - platform_device_put(s3c24xx_uda134x_snd_device); - } - - return ret; -} - -static int s3c24xx_uda134x_remove(struct platform_device *pdev) -{ - platform_device_unregister(s3c24xx_uda134x_snd_device); - gpio_free(s3c24xx_uda134x_l3_pins->l3_data); - gpio_free(s3c24xx_uda134x_l3_pins->l3_clk); - gpio_free(s3c24xx_uda134x_l3_pins->l3_mode); - return 0; -} - -static struct platform_driver s3c24xx_uda134x_driver = { - .probe = s3c24xx_uda134x_probe, - .remove = s3c24xx_uda134x_remove, - .driver = { - .name = "s3c24xx_uda134x", - .owner = THIS_MODULE, - }, -}; - -static int __init s3c24xx_uda134x_init(void) -{ - return platform_driver_register(&s3c24xx_uda134x_driver); -} - -static void __exit s3c24xx_uda134x_exit(void) -{ - platform_driver_unregister(&s3c24xx_uda134x_driver); -} - - -module_init(s3c24xx_uda134x_init); -module_exit(s3c24xx_uda134x_exit); - -MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell(a)evolware.org>"); -MODULE_DESCRIPTION("S3C24XX_UDA134X ALSA SoC audio driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c64xx-i2s-v4.c b/sound/soc/s3c24xx/s3c64xx-i2s-v4.c deleted file mode 100644 index 06db130..0000000 --- a/sound/soc/s3c24xx/s3c64xx-i2s-v4.c +++ /dev/null @@ -1,209 +0,0 @@ -/* sound/soc/s3c24xx/s3c64xx-i2s-v4.c - * - * ALSA SoC Audio Layer - S3C64XX I2Sv4 driver - * Copyright (c) 2010 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh <jassi.brar(a)samsung.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/clk.h> -#include <linux/gpio.h> -#include <linux/io.h> - -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include <mach/gpio-bank-c.h> -#include <mach/gpio-bank-h.h> -#include <plat/gpio-cfg.h> - -#include <mach/map.h> -#include <mach/dma.h> - -#include "s3c-dma.h" -#include "regs-i2s-v2.h" -#include "s3c64xx-i2s.h" - -static struct s3c2410_dma_client s3c64xx_dma_client_out = { - .name = "I2Sv4 PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c64xx_dma_client_in = { - .name = "I2Sv4 PCM Stereo in" -}; - -static struct s3c_dma_params s3c64xx_i2sv4_pcm_stereo_out; -static struct s3c_dma_params s3c64xx_i2sv4_pcm_stereo_in; -static struct s3c_i2sv2_info s3c64xx_i2sv4; - -struct snd_soc_dai s3c64xx_i2s_v4_dai; -EXPORT_SYMBOL_GPL(s3c64xx_i2s_v4_dai); - -static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) -{ - return cpu_dai->private_data; -} - -static int s3c64xx_i2sv4_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - /* configure GPIO for i2s port */ - s3c_gpio_cfgpin(S3C64XX_GPC(4), S3C64XX_GPC4_I2S_V40_DO0); - s3c_gpio_cfgpin(S3C64XX_GPC(5), S3C64XX_GPC5_I2S_V40_DO1); - s3c_gpio_cfgpin(S3C64XX_GPC(7), S3C64XX_GPC7_I2S_V40_DO2); - s3c_gpio_cfgpin(S3C64XX_GPH(6), S3C64XX_GPH6_I2S_V40_BCLK); - s3c_gpio_cfgpin(S3C64XX_GPH(7), S3C64XX_GPH7_I2S_V40_CDCLK); - s3c_gpio_cfgpin(S3C64XX_GPH(8), S3C64XX_GPH8_I2S_V40_LRCLK); - s3c_gpio_cfgpin(S3C64XX_GPH(9), S3C64XX_GPH9_I2S_V40_DI); - - return 0; -} - -static int s3c_i2sv4_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) -{ - struct s3c_i2sv2_info *i2s = to_info(cpu_dai); - struct s3c_dma_params *dma_data; - u32 iismod; - - dev_dbg(cpu_dai->dev, "Entered %s\n", __func__); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = i2s->dma_playback; - else - dma_data = i2s->dma_capture; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - iismod = readl(i2s->regs + S3C2412_IISMOD); - dev_dbg(cpu_dai->dev, "%s: r: IISMOD: %x\n", __func__, iismod); - - iismod &= ~S3C64XX_IISMOD_BLC_MASK; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - iismod |= S3C64XX_IISMOD_BLC_8BIT; - break; - case SNDRV_PCM_FORMAT_S16_LE: - break; - case SNDRV_PCM_FORMAT_S24_LE: - iismod |= S3C64XX_IISMOD_BLC_24BIT; - break; - } - - writel(iismod, i2s->regs + S3C2412_IISMOD); - dev_dbg(cpu_dai->dev, "%s: w: IISMOD: %x\n", __func__, iismod); - - return 0; -} - -static struct snd_soc_dai_ops s3c64xx_i2sv4_dai_ops = { - .hw_params = s3c_i2sv4_hw_params, -}; - -static __devinit int s3c64xx_i2sv4_dev_probe(struct platform_device *pdev) -{ - struct s3c_i2sv2_info *i2s; - struct snd_soc_dai *dai; - int ret; - - i2s = &s3c64xx_i2sv4; - dai = &s3c64xx_i2s_v4_dai; - - if (dai->dev) { - dev_dbg(dai->dev, "%s: \ - I2Sv4 instance already registered!\n", __func__); - return -EBUSY; - } - - dai->dev = &pdev->dev; - dai->name = "s3c64xx-i2s-v4"; - dai->id = 0; - dai->symmetric_rates = 1; - dai->playback.channels_min = 2; - dai->playback.channels_max = 2; - dai->playback.rates = S3C64XX_I2S_RATES; - dai->playback.formats = S3C64XX_I2S_FMTS; - dai->capture.channels_min = 2; - dai->capture.channels_max = 2; - dai->capture.rates = S3C64XX_I2S_RATES; - dai->capture.formats = S3C64XX_I2S_FMTS; - dai->probe = s3c64xx_i2sv4_probe; - dai->ops = &s3c64xx_i2sv4_dai_ops; - - i2s->feature |= S3C_FEATURE_CDCLKCON; - - i2s->dma_capture = &s3c64xx_i2sv4_pcm_stereo_in; - i2s->dma_playback = &s3c64xx_i2sv4_pcm_stereo_out; - - i2s->dma_capture->channel = DMACH_HSI_I2SV40_RX; - i2s->dma_capture->dma_addr = S3C64XX_PA_IISV4 + S3C2412_IISRXD; - i2s->dma_playback->channel = DMACH_HSI_I2SV40_TX; - i2s->dma_playback->dma_addr = S3C64XX_PA_IISV4 + S3C2412_IISTXD; - - i2s->dma_capture->client = &s3c64xx_dma_client_in; - i2s->dma_capture->dma_size = 4; - i2s->dma_playback->client = &s3c64xx_dma_client_out; - i2s->dma_playback->dma_size = 4; - - i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus"); - if (IS_ERR(i2s->iis_cclk)) { - dev_err(&pdev->dev, "failed to get audio-bus\n"); - ret = PTR_ERR(i2s->iis_cclk); - goto err; - } - - clk_enable(i2s->iis_cclk); - - ret = s3c_i2sv2_probe(pdev, dai, i2s, 0); - if (ret) - goto err_clk; - - ret = s3c_i2sv2_register_dai(dai); - if (ret != 0) - goto err_i2sv2; - - return 0; - -err_i2sv2: - /* Not implemented for I2Sv2 core yet */ -err_clk: - clk_put(i2s->iis_cclk); -err: - return ret; -} - -static __devexit int s3c64xx_i2sv4_dev_remove(struct platform_device *pdev) -{ - dev_err(&pdev->dev, "Device removal not yet supported\n"); - return 0; -} - -static struct platform_driver s3c64xx_i2sv4_driver = { - .probe = s3c64xx_i2sv4_dev_probe, - .remove = s3c64xx_i2sv4_dev_remove, - .driver = { - .name = "s3c64xx-iis-v4", - .owner = THIS_MODULE, - }, -}; - -static int __init s3c64xx_i2sv4_init(void) -{ - return platform_driver_register(&s3c64xx_i2sv4_driver); -} -module_init(s3c64xx_i2sv4_init); - -static void __exit s3c64xx_i2sv4_exit(void) -{ - platform_driver_unregister(&s3c64xx_i2sv4_driver); -} -module_exit(s3c64xx_i2sv4_exit); - -/* Module information */ -MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); -MODULE_DESCRIPTION("S3C64XX I2Sv4 SoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c deleted file mode 100644 index 1d85cb8..0000000 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ /dev/null @@ -1,190 +0,0 @@ -/* sound/soc/s3c24xx/s3c64xx-i2s.c - * - * ALSA SoC Audio Layer - S3C64XX I2S driver - * - * Copyright 2008 Openmoko, Inc. - * Copyright 2008 Simtec Electronics - * Ben Dooks <ben(a)simtec.co.uk> - * http://armlinux.simtec.co.uk/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/clk.h> -#include <linux/gpio.h> -#include <linux/io.h> - -#include <sound/soc.h> - -#include <mach/gpio-bank-d.h> -#include <mach/gpio-bank-e.h> -#include <plat/gpio-cfg.h> - -#include <mach/map.h> -#include <mach/dma.h> - -#include "s3c-dma.h" -#include "regs-i2s-v2.h" -#include "s3c64xx-i2s.h" - -/* The value should be set to maximum of the total number - * of I2Sv3 controllers that any supported SoC has. - */ -#define MAX_I2SV3 2 - -static struct s3c2410_dma_client s3c64xx_dma_client_out = { - .name = "I2S PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c64xx_dma_client_in = { - .name = "I2S PCM Stereo in" -}; - -static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_out[MAX_I2SV3]; -static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_in[MAX_I2SV3]; -static struct s3c_i2sv2_info s3c64xx_i2s[MAX_I2SV3]; - -struct snd_soc_dai s3c64xx_i2s_dai[MAX_I2SV3]; -EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai); - -static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) -{ - return cpu_dai->private_data; -} - -static int s3c64xx_i2s_probe(struct platform_device *pdev, - struct snd_soc_dai *dai) -{ - /* configure GPIO for i2s port */ - switch (dai->id) { - case 0: - s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_I2S0_CLK); - s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_I2S0_CDCLK); - s3c_gpio_cfgpin(S3C64XX_GPD(2), S3C64XX_GPD2_I2S0_LRCLK); - s3c_gpio_cfgpin(S3C64XX_GPD(3), S3C64XX_GPD3_I2S0_DI); - s3c_gpio_cfgpin(S3C64XX_GPD(4), S3C64XX_GPD4_I2S0_D0); - break; - case 1: - s3c_gpio_cfgpin(S3C64XX_GPE(0), S3C64XX_GPE0_I2S1_CLK); - s3c_gpio_cfgpin(S3C64XX_GPE(1), S3C64XX_GPE1_I2S1_CDCLK); - s3c_gpio_cfgpin(S3C64XX_GPE(2), S3C64XX_GPE2_I2S1_LRCLK); - s3c_gpio_cfgpin(S3C64XX_GPE(3), S3C64XX_GPE3_I2S1_DI); - s3c_gpio_cfgpin(S3C64XX_GPE(4), S3C64XX_GPE4_I2S1_D0); - } - - return 0; -} - - -static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops; - -static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) -{ - struct s3c_i2sv2_info *i2s; - struct snd_soc_dai *dai; - int ret; - - if (pdev->id >= MAX_I2SV3) { - dev_err(&pdev->dev, "id %d out of range\n", pdev->id); - return -EINVAL; - } - - i2s = &s3c64xx_i2s[pdev->id]; - dai = &s3c64xx_i2s_dai[pdev->id]; - dai->dev = &pdev->dev; - dai->name = "s3c64xx-i2s"; - dai->id = pdev->id; - dai->symmetric_rates = 1; - dai->playback.channels_min = 2; - dai->playback.channels_max = 2; - dai->playback.rates = S3C64XX_I2S_RATES; - dai->playback.formats = S3C64XX_I2S_FMTS; - dai->capture.channels_min = 2; - dai->capture.channels_max = 2; - dai->capture.rates = S3C64XX_I2S_RATES; - dai->capture.formats = S3C64XX_I2S_FMTS; - dai->probe = s3c64xx_i2s_probe; - dai->ops = &s3c64xx_i2s_dai_ops; - - i2s->feature |= S3C_FEATURE_CDCLKCON; - - i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id]; - i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id]; - - if (pdev->id == 0) { - i2s->dma_capture->channel = DMACH_I2S0_IN; - i2s->dma_capture->dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISRXD; - i2s->dma_playback->channel = DMACH_I2S0_OUT; - i2s->dma_playback->dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISTXD; - } else { - i2s->dma_capture->channel = DMACH_I2S1_IN; - i2s->dma_capture->dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISRXD; - i2s->dma_playback->channel = DMACH_I2S1_OUT; - i2s->dma_playback->dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISTXD; - } - - i2s->dma_capture->client = &s3c64xx_dma_client_in; - i2s->dma_capture->dma_size = 4; - i2s->dma_playback->client = &s3c64xx_dma_client_out; - i2s->dma_playback->dma_size = 4; - - i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus"); - if (IS_ERR(i2s->iis_cclk)) { - dev_err(&pdev->dev, "failed to get audio-bus\n"); - ret = PTR_ERR(i2s->iis_cclk); - goto err; - } - - clk_enable(i2s->iis_cclk); - - ret = s3c_i2sv2_probe(pdev, dai, i2s, 0); - if (ret) - goto err_clk; - - ret = s3c_i2sv2_register_dai(dai); - if (ret != 0) - goto err_i2sv2; - - return 0; - -err_i2sv2: - /* Not implemented for I2Sv2 core yet */ -err_clk: - clk_put(i2s->iis_cclk); -err: - return ret; -} - -static __devexit int s3c64xx_iis_dev_remove(struct platform_device *pdev) -{ - dev_err(&pdev->dev, "Device removal not yet supported\n"); - return 0; -} - -static struct platform_driver s3c64xx_iis_driver = { - .probe = s3c64xx_iis_dev_probe, - .remove = s3c64xx_iis_dev_remove, - .driver = { - .name = "s3c64xx-iis", - .owner = THIS_MODULE, - }, -}; - -static int __init s3c64xx_i2s_init(void) -{ - return platform_driver_register(&s3c64xx_iis_driver); -} -module_init(s3c64xx_i2s_init); - -static void __exit s3c64xx_i2s_exit(void) -{ - platform_driver_unregister(&s3c64xx_iis_driver); -} -module_exit(s3c64xx_i2s_exit); - -/* Module information */ -MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); -MODULE_DESCRIPTION("S3C64XX I2S SoC Interface"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h deleted file mode 100644 index 7a40f43..0000000 --- a/sound/soc/s3c24xx/s3c64xx-i2s.h +++ /dev/null @@ -1,42 +0,0 @@ -/* sound/soc/s3c24xx/s3c64xx-i2s.h - * - * ALSA SoC Audio Layer - S3C64XX I2S driver - * - * Copyright 2008 Openmoko, Inc. - * Copyright 2008 Simtec Electronics - * Ben Dooks <ben(a)simtec.co.uk> - * http://armlinux.simtec.co.uk/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __SND_SOC_S3C24XX_S3C64XX_I2S_H -#define __SND_SOC_S3C24XX_S3C64XX_I2S_H __FILE__ - -struct clk; - -#include "s3c-i2s-v2.h" - -#define S3C64XX_DIV_BCLK S3C_I2SV2_DIV_BCLK -#define S3C64XX_DIV_RCLK S3C_I2SV2_DIV_RCLK -#define S3C64XX_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER - -#define S3C64XX_CLKSRC_PCLK S3C_I2SV2_CLKSRC_PCLK -#define S3C64XX_CLKSRC_MUX S3C_I2SV2_CLKSRC_AUDIOBUS -#define S3C64XX_CLKSRC_CDCLK S3C_I2SV2_CLKSRC_CDCLK - -#define S3C64XX_I2S_RATES \ - (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) - -#define S3C64XX_I2S_FMTS \ - (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE) - -extern struct snd_soc_dai s3c64xx_i2s_dai[]; -extern struct snd_soc_dai s3c64xx_i2s_v4_dai; - -#endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */ diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c deleted file mode 100644 index 3622588..0000000 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ /dev/null @@ -1,81 +0,0 @@ -/* - * smdk2443_wm9710.c -- SoC audio for smdk2443 - * - * Copyright 2007 Wolfson Microelectronics PLC. - * Author: Graeme Gregory - * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - * - */ - -#include <linux/module.h> -#include <linux/device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include "../codecs/ac97.h" -#include "s3c-dma.h" -#include "s3c-ac97.h" - -static struct snd_soc_card smdk2443; - -static struct snd_soc_dai_link smdk2443_dai[] = { -{ - .name = "AC97", - .stream_name = "AC97 HiFi", - .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], - .codec_dai = &ac97_dai, -}, -}; - -static struct snd_soc_card smdk2443 = { - .name = "SMDK2443", - .platform = &s3c24xx_soc_platform, - .dai_link = smdk2443_dai, - .num_links = ARRAY_SIZE(smdk2443_dai), -}; - -static struct snd_soc_device smdk2443_snd_ac97_devdata = { - .card = &smdk2443, - .codec_dev = &soc_codec_dev_ac97, -}; - -static struct platform_device *smdk2443_snd_ac97_device; - -static int __init smdk2443_init(void) -{ - int ret; - - smdk2443_snd_ac97_device = platform_device_alloc("soc-audio", -1); - if (!smdk2443_snd_ac97_device) - return -ENOMEM; - - platform_set_drvdata(smdk2443_snd_ac97_device, - &smdk2443_snd_ac97_devdata); - smdk2443_snd_ac97_devdata.dev = &smdk2443_snd_ac97_device->dev; - ret = platform_device_add(smdk2443_snd_ac97_device); - - if (ret) - platform_device_put(smdk2443_snd_ac97_device); - - return ret; -} - -static void __exit smdk2443_exit(void) -{ - platform_device_unregister(smdk2443_snd_ac97_device); -} - -module_init(smdk2443_init); -module_exit(smdk2443_exit); - -/* Module information */ -MODULE_AUTHOR("Graeme Gregory, graeme.gregory(a)wolfsonmicro.com, www.wolfsonmicro.com"); -MODULE_DESCRIPTION("ALSA SoC WM9710 SMDK2443"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c deleted file mode 100644 index 07e8e51..0000000 --- a/sound/soc/s3c24xx/smdk64xx_wm8580.c +++ /dev/null @@ -1,266 +0,0 @@ -/* - * smdk64xx_wm8580.c - * - * Copyright (c) 2009 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh <jassi.brar(a)samsung.com> - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ - -#include <linux/platform_device.h> -#include <linux/clk.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> - -#include "../codecs/wm8580.h" -#include "s3c-dma.h" -#include "s3c64xx-i2s.h" - -/* SMDK64XX has a 12MHZ crystal attached to WM8580 */ -#define SMDK64XX_WM8580_FREQ 12000000 - -static int smdk64xx_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - unsigned int pll_out; - int bfs, rfs, ret; - - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_U8: - case SNDRV_PCM_FORMAT_S8: - bfs = 16; - break; - case SNDRV_PCM_FORMAT_U16_LE: - case SNDRV_PCM_FORMAT_S16_LE: - bfs = 32; - break; - default: - return -EINVAL; - } - - /* The Fvco for WM8580 PLLs must fall within [90,100]MHz. - * This criterion can't be met if we request PLL output - * as {8000x256, 64000x256, 11025x256}Hz. - * As a wayout, we rather change rfs to a minimum value that - * results in (params_rate(params) * rfs), and itself, acceptable - * to both - the CODEC and the CPU. - */ - switch (params_rate(params)) { - case 16000: - case 22050: - case 32000: - case 44100: - case 48000: - case 88200: - case 96000: - rfs = 256; - break; - case 64000: - rfs = 384; - break; - case 8000: - case 11025: - rfs = 512; - break; - default: - return -EINVAL; - } - pll_out = params_rate(params) * rfs; - - /* Set the Codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* Set the AP DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S - | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_CDCLK, - 0, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* We use PCLK for basic ops in SoC-Slave mode */ - ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, - 0, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* Set WM8580 to drive MCLK from its PLLA */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, - WM8580_CLKSRC_PLLA); - if (ret < 0) - return ret; - - /* Explicitly set WM8580-DAC to source from MCLK */ - ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_DAC_CLKSEL, - WM8580_CLKSRC_MCLK); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, - SMDK64XX_WM8580_FREQ, pll_out); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_BCLK, bfs); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, rfs); - if (ret < 0) - return ret; - - return 0; -} - -/* - * SMDK64XX WM8580 DAI operations. - */ -static struct snd_soc_ops smdk64xx_ops = { - .hw_params = smdk64xx_hw_params, -}; - -/* SMDK64xx Playback widgets */ -static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = { - SND_SOC_DAPM_HP("Front-L/R", NULL), - SND_SOC_DAPM_HP("Center/Sub", NULL), - SND_SOC_DAPM_HP("Rear-L/R", NULL), -}; - -/* SMDK64xx Capture widgets */ -static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = { - SND_SOC_DAPM_MIC("MicIn", NULL), - SND_SOC_DAPM_LINE("LineIn", NULL), -}; - -/* SMDK-PAIFTX connections */ -static const struct snd_soc_dapm_route audio_map_tx[] = { - /* MicIn feeds AINL */ - {"AINL", NULL, "MicIn"}, - - /* LineIn feeds AINL/R */ - {"AINL", NULL, "LineIn"}, - {"AINR", NULL, "LineIn"}, -}; - -/* SMDK-PAIFRX connections */ -static const struct snd_soc_dapm_route audio_map_rx[] = { - /* Front Left/Right are fed VOUT1L/R */ - {"Front-L/R", NULL, "VOUT1L"}, - {"Front-L/R", NULL, "VOUT1R"}, - - /* Center/Sub are fed VOUT2L/R */ - {"Center/Sub", NULL, "VOUT2L"}, - {"Center/Sub", NULL, "VOUT2R"}, - - /* Rear Left/Right are fed VOUT3L/R */ - {"Rear-L/R", NULL, "VOUT3L"}, - {"Rear-L/R", NULL, "VOUT3R"}, -}; - -static int smdk64xx_wm8580_init_paiftx(struct snd_soc_codec *codec) -{ - /* Add smdk64xx specific Capture widgets */ - snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt, - ARRAY_SIZE(wm8580_dapm_widgets_cpt)); - - /* Set up PAIFTX audio path */ - snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx)); - - /* Enabling the microphone requires the fitting of a 0R - * resistor to connect the line from the microphone jack. - */ - snd_soc_dapm_disable_pin(codec, "MicIn"); - - /* signal a DAPM event */ - snd_soc_dapm_sync(codec); - - return 0; -} - -static int smdk64xx_wm8580_init_paifrx(struct snd_soc_codec *codec) -{ - /* Add smdk64xx specific Playback widgets */ - snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk, - ARRAY_SIZE(wm8580_dapm_widgets_pbk)); - - /* Set up PAIFRX audio path */ - snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx)); - - /* signal a DAPM event */ - snd_soc_dapm_sync(codec); - - return 0; -} - -static struct snd_soc_dai_link smdk64xx_dai[] = { -{ /* Primary Playback i/f */ - .name = "WM8580 PAIF RX", - .stream_name = "Playback", - .cpu_dai = &s3c64xx_i2s_v4_dai, - .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX], - .init = smdk64xx_wm8580_init_paifrx, - .ops = &smdk64xx_ops, -}, -{ /* Primary Capture i/f */ - .name = "WM8580 PAIF TX", - .stream_name = "Capture", - .cpu_dai = &s3c64xx_i2s_v4_dai, - .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX], - .init = smdk64xx_wm8580_init_paiftx, - .ops = &smdk64xx_ops, -}, -}; - -static struct snd_soc_card smdk64xx = { - .name = "smdk64xx", - .platform = &s3c24xx_soc_platform, - .dai_link = smdk64xx_dai, - .num_links = ARRAY_SIZE(smdk64xx_dai), -}; - -static struct snd_soc_device smdk64xx_snd_devdata = { - .card = &smdk64xx, - .codec_dev = &soc_codec_dev_wm8580, -}; - -static struct platform_device *smdk64xx_snd_device; - -static int __init smdk64xx_audio_init(void) -{ - int ret; - - smdk64xx_snd_device = platform_device_alloc("soc-audio", -1); - if (!smdk64xx_snd_device) - return -ENOMEM; - - platform_set_drvdata(smdk64xx_snd_device, &smdk64xx_snd_devdata); - smdk64xx_snd_devdata.dev = &smdk64xx_snd_device->dev; - ret = platform_device_add(smdk64xx_snd_device); - - if (ret) - platform_device_put(smdk64xx_snd_device); - - return ret; -} -module_init(smdk64xx_audio_init); - -MODULE_AUTHOR("Jaswinder Singh, jassi.brar(a)samsung.com"); -MODULE_DESCRIPTION("ALSA SoC SMDK64XX WM8580"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/smdk_wm9713.c b/sound/soc/s3c24xx/smdk_wm9713.c deleted file mode 100644 index 24fd39f..0000000 --- a/sound/soc/s3c24xx/smdk_wm9713.c +++ /dev/null @@ -1,94 +0,0 @@ -/* - * smdk_wm9713.c -- SoC audio for SMDK - * - * Copyright 2010 Samsung Electronics Co. Ltd. - * Author: Jaswinder Singh Brar <jassi.brar(a)samsung.com> - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License as - * published by the Free Software Foundation; either version 2 of the - * License, or (at your option) any later version. - * - */ - -#include <linux/module.h> -#include <linux/device.h> -#include <sound/soc.h> - -#include "../codecs/wm9713.h" -#include "s3c-dma.h" -#include "s3c-ac97.h" - -static struct snd_soc_card smdk; - -/* - * Default CFG switch settings to use this driver: - * - * SMDK6410: Set CFG1 1-3 On, CFG2 1-4 Off - */ - -/* - Playback (HeadPhone):- - $ amixer sset 'Headphone' unmute - $ amixer sset 'Right Headphone Out Mux' 'Headphone' - $ amixer sset 'Left Headphone Out Mux' 'Headphone' - $ amixer sset 'Right HP Mixer PCM' unmute - $ amixer sset 'Left HP Mixer PCM' unmute - - Capture (LineIn):- - $ amixer sset 'Right Capture Source' 'Line' - $ amixer sset 'Left Capture Source' 'Line' -*/ - -static struct snd_soc_dai_link smdk_dai = { - .name = "AC97", - .stream_name = "AC97 PCM", - .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], - .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI], -}; - -static struct snd_soc_card smdk = { - .name = "SMDK", - .platform = &s3c24xx_soc_platform, - .dai_link = &smdk_dai, - .num_links = 1, -}; - -static struct snd_soc_device smdk_snd_ac97_devdata = { - .card = &smdk, - .codec_dev = &soc_codec_dev_wm9713, -}; - -static struct platform_device *smdk_snd_ac97_device; - -static int __init smdk_init(void) -{ - int ret; - - smdk_snd_ac97_device = platform_device_alloc("soc-audio", -1); - if (!smdk_snd_ac97_device) - return -ENOMEM; - - platform_set_drvdata(smdk_snd_ac97_device, - &smdk_snd_ac97_devdata); - smdk_snd_ac97_devdata.dev = &smdk_snd_ac97_device->dev; - - ret = platform_device_add(smdk_snd_ac97_device); - if (ret) - platform_device_put(smdk_snd_ac97_device); - - return ret; -} - -static void __exit smdk_exit(void) -{ - platform_device_unregister(smdk_snd_ac97_device); -} - -module_init(smdk_init); -module_exit(smdk_exit); - -/* Module information */ -MODULE_AUTHOR("Jaswinder Singh Brar, jassi.brar(a)samsung.com"); -MODULE_DESCRIPTION("ALSA SoC SMDK+WM9713"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig new file mode 100644 index 0000000..2a7cc22 --- /dev/null +++ b/sound/soc/samsung/Kconfig @@ -0,0 +1,127 @@ +config SND_S3C24XX_SOC + tristate "SoC Audio for the Samsung S3CXXXX chips" + depends on ARCH_S3C2410 || ARCH_S3C64XX + select S3C64XX_DMA if ARCH_S3C64XX + help + Say Y or M if you want to add support for codecs attached to + the S3C24XX AC97 or I2S interfaces. You will also need to + select the audio interfaces to support below. + +config SND_S3C24XX_SOC_I2S + tristate + select S3C2410_DMA + +config SND_S3C_I2SV2_SOC + tristate + +config SND_S3C2412_SOC_I2S + tristate + select SND_S3C_I2SV2_SOC + select S3C2410_DMA + +config SND_S3C64XX_SOC_I2S + tristate + select SND_S3C_I2SV2_SOC + select S3C64XX_DMA + +config SND_S3C64XX_SOC_I2S_V4 + tristate + select SND_S3C_I2SV2_SOC + select S3C64XX_DMA + +config SND_S3C_SOC_PCM + tristate + +config SND_S3C_SOC_AC97 + tristate + select SND_SOC_AC97_BUS + +config SND_S3C24XX_SOC_NEO1973_WM8753 + tristate "SoC I2S Audio support for NEO1973 - WM8753" + depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA01 + select SND_S3C24XX_SOC_I2S + select SND_SOC_WM8753 + help + Say Y if you want to add support for SoC audio on smdk2440 + with the WM8753. + +config SND_S3C24XX_SOC_NEO1973_GTA02_WM8753 + tristate "Audio support for the Openmoko Neo FreeRunner (GTA02)" + depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA02 + select SND_S3C24XX_SOC_I2S + select SND_SOC_WM8753 + help + This driver provides audio support for the Openmoko Neo FreeRunner + smartphone. + +config SND_S3C24XX_SOC_JIVE_WM8750 + tristate "SoC I2S Audio support for Jive" + depends on SND_S3C24XX_SOC && MACH_JIVE + select SND_SOC_WM8750 + select SND_S3C2412_SOC_I2S + help + Sat Y if you want to add support for SoC audio on the Jive. + +config SND_S3C64XX_SOC_WM8580 + tristate "SoC I2S Audio support for WM8580 on SMDK64XX" + depends on SND_S3C24XX_SOC && MACH_SMDK6410 + select SND_SOC_WM8580 + select SND_S3C64XX_SOC_I2S_V4 + help + Say Y if you want to add support for SoC audio on the SMDK6410. + +config SND_S3C24XX_SOC_SMDK2443_WM9710 + tristate "SoC AC97 Audio support for SMDK2443 - WM9710" + depends on SND_S3C24XX_SOC && MACH_SMDK2443 + select S3C2410_DMA + select AC97_BUS + select SND_SOC_AC97_CODEC + select SND_S3C_SOC_AC97 + help + Say Y if you want to add support for SoC audio on smdk2443 + with the WM9710. + +config SND_S3C24XX_SOC_LN2440SBC_ALC650 + tristate "SoC AC97 Audio support for LN2440SBC - ALC650" + depends on SND_S3C24XX_SOC && ARCH_S3C2410 + select S3C2410_DMA + select AC97_BUS + select SND_SOC_AC97_CODEC + select SND_S3C_SOC_AC97 + help + Say Y if you want to add support for SoC audio on ln2440sbc + with the ALC650. + +config SND_S3C24XX_SOC_S3C24XX_UDA134X + tristate "SoC I2S Audio support UDA134X wired to a S3C24XX" + depends on SND_S3C24XX_SOC && ARCH_S3C2410 + select SND_S3C24XX_SOC_I2S + select SND_SOC_L3 + select SND_SOC_UDA134X + +config SND_S3C24XX_SOC_SIMTEC + tristate + help + Internal node for common S3C24XX/Simtec suppor + +config SND_S3C24XX_SOC_SIMTEC_TLV320AIC23 + tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards" + depends on SND_S3C24XX_SOC && ARCH_S3C2410 + select SND_S3C24XX_SOC_I2S + select SND_SOC_TLV320AIC23 + select SND_S3C24XX_SOC_SIMTEC + +config SND_S3C24XX_SOC_SIMTEC_HERMES + tristate "SoC I2S Audio support for Simtec Hermes board" + depends on SND_S3C24XX_SOC && ARCH_S3C2410 + select SND_S3C24XX_SOC_I2S + select SND_SOC_TLV320AIC3X + select SND_S3C24XX_SOC_SIMTEC + +config SND_SOC_SMDK_WM9713 + tristate "SoC AC97 Audio support for SMDK with WM9713" + depends on SND_S3C24XX_SOC && MACH_SMDK6410 + select SND_SOC_WM9713 + select SND_S3C_SOC_AC97 + help + Sat Y if you want to add support for SoC audio on the SMDK. diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile new file mode 100644 index 0000000..81d8dc5 --- /dev/null +++ b/sound/soc/samsung/Makefile @@ -0,0 +1,43 @@ +# S3c24XX Platform Support +snd-soc-s3c24xx-objs := s3c-dma.o +snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o +snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o +snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o +snd-soc-s3c-ac97-objs := s3c-ac97.o +snd-soc-s3c64xx-i2s-v4-objs := s3c64xx-i2s-v4.o +snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o +snd-soc-s3c-pcm-objs := s3c-pcm.o + +obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o +obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o +obj-$(CONFIG_SND_S3C_SOC_AC97) += snd-soc-s3c-ac97.o +obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o +obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o +obj-$(CONFIG_SND_S3C64XX_SOC_I2S_V4) += snd-soc-s3c64xx-i2s-v4.o +obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o +obj-$(CONFIG_SND_S3C_SOC_PCM) += snd-soc-s3c-pcm.o + +# S3C24XX Machine Support +snd-soc-jive-wm8750-objs := jive_wm8750.o +snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o +snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o +snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o +snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o +snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o +snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o +snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o +snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o +snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o +snd-soc-smdk-wm9713-objs := smdk_wm9713.o + +obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o +obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o +obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o +obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o +obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o +obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o +obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o +obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o +obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o +obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o +obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c new file mode 100644 index 0000000..8c108b1 --- /dev/null +++ b/sound/soc/samsung/jive_wm8750.c @@ -0,0 +1,196 @@ +/* sound/soc/s3c24xx/jive_wm8750.c + * + * Copyright 2007,2008 Simtec Electronics + * + * Based on sound/soc/pxa/spitz.c + * Copyright 2005 Wolfson Microelectronics PLC. + * Copyright 2005 Openedhand Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include "s3c-dma.h" +#include "s3c2412-i2s.h" + +#include "../codecs/wm8750.h" + +static const struct snd_soc_dapm_route audio_map[] = { + { "Headphone Jack", NULL, "LOUT1" }, + { "Headphone Jack", NULL, "ROUT1" }, + { "Internal Speaker", NULL, "LOUT2" }, + { "Internal Speaker", NULL, "ROUT2" }, + { "LINPUT1", NULL, "Line Input" }, + { "RINPUT1", NULL, "Line Input" }, +}; + +static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Internal Speaker", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +static int jive_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct s3c_i2sv2_rate_calc div; + unsigned int clk = 0; + int ret = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + clk = 11289600; + break; + } + + s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params), + s3c_i2sv2_get_clock(cpu_dai)); + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_RCLK, div.fs_div); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_PRESCALER, + div.clk_div - 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops jive_ops = { + .hw_params = jive_hw_params, +}; + +static int jive_wm8750_init(struct snd_soc_codec *codec) +{ + int err; + + /* These endpoints are not being used. */ + snd_soc_dapm_nc_pin(codec, "LINPUT2"); + snd_soc_dapm_nc_pin(codec, "RINPUT2"); + snd_soc_dapm_nc_pin(codec, "LINPUT3"); + snd_soc_dapm_nc_pin(codec, "RINPUT3"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "MONO"); + + /* Add jive specific widgets */ + err = snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, + ARRAY_SIZE(wm8750_dapm_widgets)); + if (err) { + printk(KERN_ERR "%s: failed to add widgets (%d)\n", + __func__, err); + return err; + } + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link jive_dai = { + .name = "wm8750", + .stream_name = "WM8750", + .cpu_dai = &s3c2412_i2s_dai, + .codec_dai = &wm8750_dai, + .init = jive_wm8750_init, + .ops = &jive_ops, +}; + +/* jive audio machine driver */ +static struct snd_soc_card snd_soc_machine_jive = { + .name = "Jive", + .platform = &s3c24xx_soc_platform, + .dai_link = &jive_dai, + .num_links = 1, +}; + +/* jive audio subsystem */ +static struct snd_soc_device jive_snd_devdata = { + .card = &snd_soc_machine_jive, + .codec_dev = &soc_codec_dev_wm8750, +}; + +static struct platform_device *jive_snd_device; + +static int __init jive_init(void) +{ + int ret; + + if (!machine_is_jive()) + return 0; + + printk("JIVE WM8750 Audio support\n"); + + jive_snd_device = platform_device_alloc("soc-audio", -1); + if (!jive_snd_device) + return -ENOMEM; + + platform_set_drvdata(jive_snd_device, &jive_snd_devdata); + jive_snd_devdata.dev = &jive_snd_device->dev; + ret = platform_device_add(jive_snd_device); + + if (ret) + platform_device_put(jive_snd_device); + + return ret; +} + +static void __exit jive_exit(void) +{ + platform_device_unregister(jive_snd_device); +} + +module_init(jive_init); +module_exit(jive_exit); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Jive Audio support"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/lm4857.h b/sound/soc/samsung/lm4857.h new file mode 100644 index 0000000..0cf5b70 --- /dev/null +++ b/sound/soc/samsung/lm4857.h @@ -0,0 +1,32 @@ +/* + * lm4857.h -- ALSA Soc Audio Layer + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * Revision history + * 18th Jun 2007 Initial version. + */ + +#ifndef LM4857_H_ +#define LM4857_H_ + +/* The register offsets in the cache array */ +#define LM4857_MVOL 0 +#define LM4857_LVOL 1 +#define LM4857_RVOL 2 +#define LM4857_CTRL 3 + +/* the shifts required to set these bits */ +#define LM4857_3D 5 +#define LM4857_WAKEUP 5 +#define LM4857_EPGAIN 4 + +#endif /*LM4857_H_*/ + diff --git a/sound/soc/samsung/ln2440sbc_alc650.c b/sound/soc/samsung/ln2440sbc_alc650.c new file mode 100644 index 0000000..ffa954f --- /dev/null +++ b/sound/soc/samsung/ln2440sbc_alc650.c @@ -0,0 +1,85 @@ +/* + * SoC audio for ln2440sbc + * + * Copyright 2007 KonekTel, a.s. + * Author: Ivan Kuten + * ivan.kuten(a)promwad.com + * + * Heavily based on smdk2443_wm9710.c + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/ac97.h" +#include "s3c-dma.h" +#include "s3c-ac97.h" + +static struct snd_soc_card ln2440sbc; + +static struct snd_soc_dai_link ln2440sbc_dai[] = { +{ + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], + .codec_dai = &ac97_dai, +}, +}; + +static struct snd_soc_card ln2440sbc = { + .name = "LN2440SBC", + .platform = &s3c24xx_soc_platform, + .dai_link = ln2440sbc_dai, + .num_links = ARRAY_SIZE(ln2440sbc_dai), +}; + +static struct snd_soc_device ln2440sbc_snd_ac97_devdata = { + .card = &ln2440sbc, + .codec_dev = &soc_codec_dev_ac97, +}; + +static struct platform_device *ln2440sbc_snd_ac97_device; + +static int __init ln2440sbc_init(void) +{ + int ret; + + ln2440sbc_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!ln2440sbc_snd_ac97_device) + return -ENOMEM; + + platform_set_drvdata(ln2440sbc_snd_ac97_device, + &ln2440sbc_snd_ac97_devdata); + ln2440sbc_snd_ac97_devdata.dev = &ln2440sbc_snd_ac97_device->dev; + ret = platform_device_add(ln2440sbc_snd_ac97_device); + + if (ret) + platform_device_put(ln2440sbc_snd_ac97_device); + + return ret; +} + +static void __exit ln2440sbc_exit(void) +{ + platform_device_unregister(ln2440sbc_snd_ac97_device); +} + +module_init(ln2440sbc_init); +module_exit(ln2440sbc_exit); + +/* Module information */ +MODULE_AUTHOR("Ivan Kuten"); +MODULE_DESCRIPTION("ALSA SoC ALC650 LN2440SBC"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c new file mode 100644 index 0000000..209c259 --- /dev/null +++ b/sound/soc/samsung/neo1973_gta02_wm8753.c @@ -0,0 +1,506 @@ +/* + * neo1973_gta02_wm8753.c -- SoC audio for Openmoko Freerunner(GTA02) + * + * Copyright 2007 Openmoko Inc + * Author: Graeme Gregory <graeme(a)openmoko.org> + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory <linux(a)wolfsonmicro.com> + * Copyright 2009 Wolfson Microelectronics + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include <plat/regs-iis.h> + +#include <mach/regs-clock.h> +#include <asm/io.h> +#include <mach/gta02.h> +#include "../codecs/wm8753.h" +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" + +static struct snd_soc_card neo1973_gta02; + +static int neo1973_gta02_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int pll_out = 0, bclk = 0; + int ret = 0; + unsigned long iis_clkrate; + + iis_clkrate = s3c24xx_i2s_get_clockrate(); + + switch (params_rate(params)) { + case 8000: + case 16000: + pll_out = 12288000; + break; + case 48000: + bclk = WM8753_BCLK_DIV_4; + pll_out = 12288000; + break; + case 96000: + bclk = WM8753_BCLK_DIV_2; + pll_out = 12288000; + break; + case 11025: + bclk = WM8753_BCLK_DIV_16; + pll_out = 11289600; + break; + case 22050: + bclk = WM8753_BCLK_DIV_8; + pll_out = 11289600; + break; + case 44100: + bclk = WM8753_BCLK_DIV_4; + pll_out = 11289600; + break; + case 88200: + bclk = WM8753_BCLK_DIV_2; + pll_out = 11289600; + break; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + /* set codec BCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(codec_dai, + WM8753_BCLKDIV, bclk); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(4, 4)); + if (ret < 0) + return ret; + + /* codec PLL input is PCLK/4 */ + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, + iis_clkrate / 4, pll_out); + if (ret < 0) + return ret; + + return 0; +} + +static int neo1973_gta02_hifi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); +} + +/* + * Neo1973 WM8753 HiFi DAI opserations. + */ +static struct snd_soc_ops neo1973_gta02_hifi_ops = { + .hw_params = neo1973_gta02_hifi_hw_params, + .hw_free = neo1973_gta02_hifi_hw_free, +}; + +static int neo1973_gta02_voice_hw_params( + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int pcmdiv = 0; + int ret = 0; + unsigned long iis_clkrate; + + iis_clkrate = s3c24xx_i2s_get_clockrate(); + + if (params_rate(params) != 8000) + return -EINVAL; + if (params_channels(params) != 1) + return -EINVAL; + + pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */ + + /* todo: gg check mode (DSP_B) against CSR datasheet */ + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, + 12288000, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set codec PCM division for sample rate */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, + pcmdiv); + if (ret < 0) + return ret; + + /* configue and enable PLL for 12.288MHz output */ + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, + iis_clkrate / 4, 12288000); + if (ret < 0) + return ret; + + return 0; +} + +static int neo1973_gta02_voice_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); +} + +static struct snd_soc_ops neo1973_gta02_voice_ops = { + .hw_params = neo1973_gta02_voice_hw_params, + .hw_free = neo1973_gta02_voice_hw_free, +}; + +#define LM4853_AMP 1 +#define LM4853_SPK 2 + +static u8 lm4853_state; + +/* This has no effect, it exists only to maintain compatibility with + * existing ALSA state files. + */ +static int lm4853_set_state(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int val = ucontrol->value.integer.value[0]; + + if (val) + lm4853_state |= LM4853_AMP; + else + lm4853_state &= ~LM4853_AMP; + + return 0; +} + +static int lm4853_get_state(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = lm4853_state & LM4853_AMP; + + return 0; +} + +static int lm4853_set_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int val = ucontrol->value.integer.value[0]; + + if (val) { + lm4853_state |= LM4853_SPK; + gpio_set_value(GTA02_GPIO_HP_IN, 0); + } else { + lm4853_state &= ~LM4853_SPK; + gpio_set_value(GTA02_GPIO_HP_IN, 1); + } + + return 0; +} + +static int lm4853_get_spk(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = (lm4853_state & LM4853_SPK) >> 1; + + return 0; +} + +static int lm4853_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, + int event) +{ + gpio_set_value(GTA02_GPIO_AMP_SHUT, SND_SOC_DAPM_EVENT_OFF(value)); + + return 0; +} + +static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Stereo Out", lm4853_event), + SND_SOC_DAPM_LINE("GSM Line Out", NULL), + SND_SOC_DAPM_LINE("GSM Line In", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Handset Mic", NULL), + SND_SOC_DAPM_SPK("Handset Spk", NULL), +}; + + +/* example machine audio_mapnections */ +static const struct snd_soc_dapm_route audio_map[] = { + + /* Connections to the lm4853 amp */ + {"Stereo Out", NULL, "LOUT1"}, + {"Stereo Out", NULL, "ROUT1"}, + + /* Connections to the GSM Module */ + {"GSM Line Out", NULL, "MONO1"}, + {"GSM Line Out", NULL, "MONO2"}, + {"RXP", NULL, "GSM Line In"}, + {"RXN", NULL, "GSM Line In"}, + + /* Connections to Headset */ + {"MIC1", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Headset Mic"}, + + /* Call Mic */ + {"MIC2", NULL, "Mic Bias"}, + {"MIC2N", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Handset Mic"}, + + /* Call Speaker */ + {"Handset Spk", NULL, "LOUT2"}, + {"Handset Spk", NULL, "ROUT2"}, + + /* Connect the ALC pins */ + {"ACIN", NULL, "ACOP"}, +}; + +static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = { + SOC_DAPM_PIN_SWITCH("Stereo Out"), + SOC_DAPM_PIN_SWITCH("GSM Line Out"), + SOC_DAPM_PIN_SWITCH("GSM Line In"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Handset Mic"), + SOC_DAPM_PIN_SWITCH("Handset Spk"), + + /* This has no effect, it exists only to maintain compatibility with + * existing ALSA state files. + */ + SOC_SINGLE_EXT("Amp State Switch", 6, 0, 1, 0, + lm4853_get_state, + lm4853_set_state), + SOC_SINGLE_EXT("Amp Spk Switch", 7, 0, 1, 0, + lm4853_get_spk, + lm4853_set_spk), +}; + +/* + * This is an example machine initialisation for a wm8753 connected to a + * neo1973 GTA02. + */ +static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec) +{ + int err; + + /* set up NC codec pins */ + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "OUT4"); + snd_soc_dapm_nc_pin(codec, "LINE1"); + snd_soc_dapm_nc_pin(codec, "LINE2"); + + /* Add neo1973 gta02 specific widgets */ + snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, + ARRAY_SIZE(wm8753_dapm_widgets)); + + /* add neo1973 gta02 specific controls */ + err = snd_soc_add_controls(codec, wm8753_neo1973_gta02_controls, + ARRAY_SIZE(wm8753_neo1973_gta02_controls)); + + if (err < 0) + return err; + + /* set up neo1973 gta02 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* set endpoints to default off mode */ + snd_soc_dapm_disable_pin(codec, "Stereo Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Handset Mic"); + snd_soc_dapm_disable_pin(codec, "Handset Spk"); + + /* allow audio paths from the GSM modem to run during suspend */ + snd_soc_dapm_ignore_suspend(codec, "Stereo Out"); + snd_soc_dapm_ignore_suspend(codec, "GSM Line Out"); + snd_soc_dapm_ignore_suspend(codec, "GSM Line In"); + snd_soc_dapm_ignore_suspend(codec, "Headset Mic"); + snd_soc_dapm_ignore_suspend(codec, "Handset Mic"); + snd_soc_dapm_ignore_suspend(codec, "Handset Spk"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* + * BT Codec DAI + */ +static struct snd_soc_dai bt_dai = { + .name = "Bluetooth", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_dai_link neo1973_gta02_dai[] = { +{ /* Hifi Playback - for similatious use with voice below */ + .name = "WM8753", + .stream_name = "WM8753 HiFi", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &wm8753_dai[WM8753_DAI_HIFI], + .init = neo1973_gta02_wm8753_init, + .ops = &neo1973_gta02_hifi_ops, +}, +{ /* Voice via BT */ + .name = "Bluetooth", + .stream_name = "Voice", + .cpu_dai = &bt_dai, + .codec_dai = &wm8753_dai[WM8753_DAI_VOICE], + .ops = &neo1973_gta02_voice_ops, +}, +}; + +static struct snd_soc_card neo1973_gta02 = { + .name = "neo1973-gta02", + .platform = &s3c24xx_soc_platform, + .dai_link = neo1973_gta02_dai, + .num_links = ARRAY_SIZE(neo1973_gta02_dai), +}; + +static struct snd_soc_device neo1973_gta02_snd_devdata = { + .card = &neo1973_gta02, + .codec_dev = &soc_codec_dev_wm8753, +}; + +static struct platform_device *neo1973_gta02_snd_device; + +static int __init neo1973_gta02_init(void) +{ + int ret; + + if (!machine_is_neo1973_gta02()) { + printk(KERN_INFO + "Only GTA02 is supported by this ASoC driver\n"); + return -ENODEV; + } + + /* register bluetooth DAI here */ + ret = snd_soc_register_dai(&bt_dai); + if (ret) + return ret; + + neo1973_gta02_snd_device = platform_device_alloc("soc-audio", -1); + if (!neo1973_gta02_snd_device) + return -ENOMEM; + + platform_set_drvdata(neo1973_gta02_snd_device, + &neo1973_gta02_snd_devdata); + neo1973_gta02_snd_devdata.dev = &neo1973_gta02_snd_device->dev; + ret = platform_device_add(neo1973_gta02_snd_device); + + if (ret) { + platform_device_put(neo1973_gta02_snd_device); + return ret; + } + + /* Initialise GPIOs used by amp */ + ret = gpio_request(GTA02_GPIO_HP_IN, "GTA02_HP_IN"); + if (ret) { + pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_HP_IN); + goto err_unregister_device; + } + + ret = gpio_direction_output(GTA02_GPIO_AMP_HP_IN, 1); + if (ret) { + pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_HP_IN); + goto err_free_gpio_hp_in; + } + + ret = gpio_request(GTA02_GPIO_AMP_SHUT, "GTA02_AMP_SHUT"); + if (ret) { + pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_AMP_SHUT); + goto err_free_gpio_hp_in; + } + + ret = gpio_direction_output(GTA02_GPIO_AMP_SHUT, 1); + if (ret) { + pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_AMP_SHUT); + goto err_free_gpio_amp_shut; + } + + return 0; + +err_free_gpio_amp_shut: + gpio_free(GTA02_GPIO_AMP_SHUT); +err_free_gpio_hp_in: + gpio_free(GTA02_GPIO_HP_IN); +err_unregister_device: + platform_device_unregister(neo1973_gta02_snd_device); + return ret; +} +module_init(neo1973_gta02_init); + +static void __exit neo1973_gta02_exit(void) +{ + snd_soc_unregister_dai(&bt_dai); + platform_device_unregister(neo1973_gta02_snd_device); + gpio_free(GTA02_GPIO_HP_IN); + gpio_free(GTA02_GPIO_AMP_SHUT); +} +module_exit(neo1973_gta02_exit); + +/* Module information */ +MODULE_AUTHOR("Graeme Gregory, graeme(a)openmoko.org"); +MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 GTA02"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c new file mode 100644 index 0000000..0cb4f86 --- /dev/null +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -0,0 +1,707 @@ +/* + * neo1973_wm8753.c -- SoC audio for Neo1973 + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include <asm/mach-types.h> +#include <asm/hardware/scoop.h> +#include <mach/regs-clock.h> +#include <mach/regs-gpio.h> +#include <mach/hardware.h> +#include <linux/io.h> +#include <mach/spi-gpio.h> + +#include <plat/regs-iis.h> + +#include "../codecs/wm8753.h" +#include "lm4857.h" +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" + +/* define the scenarios */ +#define NEO_AUDIO_OFF 0 +#define NEO_GSM_CALL_AUDIO_HANDSET 1 +#define NEO_GSM_CALL_AUDIO_HEADSET 2 +#define NEO_GSM_CALL_AUDIO_BLUETOOTH 3 +#define NEO_STEREO_TO_SPEAKERS 4 +#define NEO_STEREO_TO_HEADPHONES 5 +#define NEO_CAPTURE_HANDSET 6 +#define NEO_CAPTURE_HEADSET 7 +#define NEO_CAPTURE_BLUETOOTH 8 + +static struct snd_soc_card neo1973; +static struct i2c_client *i2c; + +static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int pll_out = 0, bclk = 0; + int ret = 0; + unsigned long iis_clkrate; + + pr_debug("Entered %s\n", __func__); + + iis_clkrate = s3c24xx_i2s_get_clockrate(); + + switch (params_rate(params)) { + case 8000: + case 16000: + pll_out = 12288000; + break; + case 48000: + bclk = WM8753_BCLK_DIV_4; + pll_out = 12288000; + break; + case 96000: + bclk = WM8753_BCLK_DIV_2; + pll_out = 12288000; + break; + case 11025: + bclk = WM8753_BCLK_DIV_16; + pll_out = 11289600; + break; + case 22050: + bclk = WM8753_BCLK_DIV_8; + pll_out = 11289600; + break; + case 44100: + bclk = WM8753_BCLK_DIV_4; + pll_out = 11289600; + break; + case 88200: + bclk = WM8753_BCLK_DIV_2; + pll_out = 11289600; + break; + } + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set MCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + /* set codec BCLK division for sample rate */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk); + if (ret < 0) + return ret; + + /* set prescaler division for sample rate */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(4, 4)); + if (ret < 0) + return ret; + + /* codec PLL input is PCLK/4 */ + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, + iis_clkrate / 4, pll_out); + if (ret < 0) + return ret; + + return 0; +} + +static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + pr_debug("Entered %s\n", __func__); + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); +} + +/* + * Neo1973 WM8753 HiFi DAI opserations. + */ +static struct snd_soc_ops neo1973_hifi_ops = { + .hw_params = neo1973_hifi_hw_params, + .hw_free = neo1973_hifi_hw_free, +}; + +static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int pcmdiv = 0; + int ret = 0; + unsigned long iis_clkrate; + + pr_debug("Entered %s\n", __func__); + + iis_clkrate = s3c24xx_i2s_get_clockrate(); + + if (params_rate(params) != 8000) + return -EINVAL; + if (params_channels(params) != 1) + return -EINVAL; + + pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */ + + /* todo: gg check mode (DSP_B) against CSR datasheet */ + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set codec PCM division for sample rate */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv); + if (ret < 0) + return ret; + + /* configue and enable PLL for 12.288MHz output */ + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, + iis_clkrate / 4, 12288000); + if (ret < 0) + return ret; + + return 0; +} + +static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + pr_debug("Entered %s\n", __func__); + + /* disable the PLL */ + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); +} + +static struct snd_soc_ops neo1973_voice_ops = { + .hw_params = neo1973_voice_hw_params, + .hw_free = neo1973_voice_hw_free, +}; + +static int neo1973_scenario; + +static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = neo1973_scenario; + return 0; +} + +static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) +{ + pr_debug("Entered %s\n", __func__); + + switch (neo1973_scenario) { + case NEO_AUDIO_OFF: + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + break; + case NEO_GSM_CALL_AUDIO_HANDSET: + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); + break; + case NEO_GSM_CALL_AUDIO_HEADSET: + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + break; + case NEO_GSM_CALL_AUDIO_BLUETOOTH: + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + break; + case NEO_STEREO_TO_SPEAKERS: + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + break; + case NEO_STEREO_TO_HEADPHONES: + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + break; + case NEO_CAPTURE_HANDSET: + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); + break; + case NEO_CAPTURE_HEADSET: + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + break; + case NEO_CAPTURE_BLUETOOTH: + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + break; + default: + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); + } + + snd_soc_dapm_sync(codec); + + return 0; +} + +static int neo1973_set_scenario(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + pr_debug("Entered %s\n", __func__); + + if (neo1973_scenario == ucontrol->value.integer.value[0]) + return 0; + + neo1973_scenario = ucontrol->value.integer.value[0]; + set_scenario_endpoints(codec, neo1973_scenario); + return 1; +} + +static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0}; + +static void lm4857_write_regs(void) +{ + pr_debug("Entered %s\n", __func__); + + if (i2c_master_send(i2c, lm4857_regs, 4) != 4) + printk(KERN_ERR "lm4857: i2c write failed\n"); +} + +static int lm4857_get_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int reg = mc->reg; + int shift = mc->shift; + int mask = mc->max; + + pr_debug("Entered %s\n", __func__); + + ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask; + return 0; +} + +static int lm4857_set_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int reg = mc->reg; + int shift = mc->shift; + int mask = mc->max; + + if (((lm4857_regs[reg] >> shift) & mask) == + ucontrol->value.integer.value[0]) + return 0; + + lm4857_regs[reg] &= ~(mask << shift); + lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift; + lm4857_write_regs(); + return 1; +} + +static int lm4857_get_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = lm4857_regs[LM4857_CTRL] & 0x0F; + + pr_debug("Entered %s\n", __func__); + + if (value) + value -= 5; + + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int lm4857_set_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 value = ucontrol->value.integer.value[0]; + + pr_debug("Entered %s\n", __func__); + + if (value) + value += 5; + + if ((lm4857_regs[LM4857_CTRL] & 0x0F) == value) + return 0; + + lm4857_regs[LM4857_CTRL] &= 0xF0; + lm4857_regs[LM4857_CTRL] |= value; + lm4857_write_regs(); + return 1; +} + +static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { + SND_SOC_DAPM_LINE("Audio Out", NULL), + SND_SOC_DAPM_LINE("GSM Line Out", NULL), + SND_SOC_DAPM_LINE("GSM Line In", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Call Mic", NULL), +}; + + +static const struct snd_soc_dapm_route dapm_routes[] = { + + /* Connections to the lm4857 amp */ + {"Audio Out", NULL, "LOUT1"}, + {"Audio Out", NULL, "ROUT1"}, + + /* Connections to the GSM Module */ + {"GSM Line Out", NULL, "MONO1"}, + {"GSM Line Out", NULL, "MONO2"}, + {"RXP", NULL, "GSM Line In"}, + {"RXN", NULL, "GSM Line In"}, + + /* Connections to Headset */ + {"MIC1", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Headset Mic"}, + + /* Call Mic */ + {"MIC2", NULL, "Mic Bias"}, + {"MIC2N", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Call Mic"}, + + /* Connect the ALC pins */ + {"ACIN", NULL, "ACOP"}, +}; + +static const char *lm4857_mode[] = { + "Off", + "Call Speaker", + "Stereo Speakers", + "Stereo Speakers + Headphones", + "Headphones" +}; + +static const struct soc_enum lm4857_mode_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode), +}; + +static const char *neo_scenarios[] = { + "Off", + "GSM Handset", + "GSM Headset", + "GSM Bluetooth", + "Speakers", + "Headphones", + "Capture Handset", + "Capture Headset", + "Capture Bluetooth" +}; + +static const struct soc_enum neo_scenario_enum[] = { + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios), +}; + +static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0); +static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0); + +static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { + SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, stereo_tlv), + SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, stereo_tlv), + SOC_SINGLE_EXT_TLV("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, mono_tlv), + SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0], + lm4857_get_mode, lm4857_set_mode), + SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0], + neo1973_get_scenario, neo1973_set_scenario), + SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0, + lm4857_get_reg, lm4857_set_reg), + SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0, + lm4857_get_reg, lm4857_set_reg), + SOC_SINGLE_EXT("Amp Fast Wakeup Playback Switch", LM4857_CTRL, 5, 1, 0, + lm4857_get_reg, lm4857_set_reg), + SOC_SINGLE_EXT("Amp Earpiece 6dB Playback Switch", LM4857_CTRL, 4, 1, 0, + lm4857_get_reg, lm4857_set_reg), +}; + +/* + * This is an example machine initialisation for a wm8753 connected to a + * neo1973 II. It is missing logic to detect hp/mic insertions and logic + * to re-route the audio in such an event. + */ +static int neo1973_wm8753_init(struct snd_soc_codec *codec) +{ + int err; + + pr_debug("Entered %s\n", __func__); + + /* set up NC codec pins */ + snd_soc_dapm_nc_pin(codec, "LOUT2"); + snd_soc_dapm_nc_pin(codec, "ROUT2"); + snd_soc_dapm_nc_pin(codec, "OUT3"); + snd_soc_dapm_nc_pin(codec, "OUT4"); + snd_soc_dapm_nc_pin(codec, "LINE1"); + snd_soc_dapm_nc_pin(codec, "LINE2"); + + /* Add neo1973 specific widgets */ + snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, + ARRAY_SIZE(wm8753_dapm_widgets)); + + /* set endpoints to default mode */ + set_scenario_endpoints(codec, NEO_AUDIO_OFF); + + /* add neo1973 specific controls */ + err = snd_soc_add_controls(codec, wm8753_neo1973_controls, + ARRAY_SIZE(8753_neo1973_controls)); + if (err < 0) + return err; + + /* set up neo1973 specific audio routes */ + err = snd_soc_dapm_add_routes(codec, dapm_routes, + ARRAY_SIZE(dapm_routes)); + + snd_soc_dapm_sync(codec); + return 0; +} + +/* + * BT Codec DAI + */ +static struct snd_soc_dai bt_dai = { + .name = "Bluetooth", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_dai_link neo1973_dai[] = { +{ /* Hifi Playback - for similatious use with voice below */ + .name = "WM8753", + .stream_name = "WM8753 HiFi", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &wm8753_dai[WM8753_DAI_HIFI], + .init = neo1973_wm8753_init, + .ops = &neo1973_hifi_ops, +}, +{ /* Voice via BT */ + .name = "Bluetooth", + .stream_name = "Voice", + .cpu_dai = &bt_dai, + .codec_dai = &wm8753_dai[WM8753_DAI_VOICE], + .ops = &neo1973_voice_ops, +}, +}; + +static struct snd_soc_card neo1973 = { + .name = "neo1973", + .platform = &s3c24xx_soc_platform, + .dai_link = neo1973_dai, + .num_links = ARRAY_SIZE(neo1973_dai), +}; + +static struct snd_soc_device neo1973_snd_devdata = { + .card = &neo1973, + .codec_dev = &soc_codec_dev_wm8753, +}; + +static int lm4857_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + pr_debug("Entered %s\n", __func__); + + i2c = client; + + lm4857_write_regs(); + return 0; +} + +static int lm4857_i2c_remove(struct i2c_client *client) +{ + pr_debug("Entered %s\n", __func__); + + i2c = NULL; + + return 0; +} + +static u8 lm4857_state; + +static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) +{ + pr_debug("Entered %s\n", __func__); + + dev_dbg(&dev->dev, "lm4857_suspend\n"); + lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf; + if (lm4857_state) { + lm4857_regs[LM4857_CTRL] &= 0xf0; + lm4857_write_regs(); + } + return 0; +} + +static int lm4857_resume(struct i2c_client *dev) +{ + pr_debug("Entered %s\n", __func__); + + if (lm4857_state) { + lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f); + lm4857_write_regs(); + } + return 0; +} + +static void lm4857_shutdown(struct i2c_client *dev) +{ + pr_debug("Entered %s\n", __func__); + + dev_dbg(&dev->dev, "lm4857_shutdown\n"); + lm4857_regs[LM4857_CTRL] &= 0xf0; + lm4857_write_regs(); +} + +static const struct i2c_device_id lm4857_i2c_id[] = { + { "neo1973_lm4857", 0 }, + { } +}; + +static struct i2c_driver lm4857_i2c_driver = { + .driver = { + .name = "LM4857 I2C Amp", + .owner = THIS_MODULE, + }, + .suspend = lm4857_suspend, + .resume = lm4857_resume, + .shutdown = lm4857_shutdown, + .probe = lm4857_i2c_probe, + .remove = lm4857_i2c_remove, + .id_table = lm4857_i2c_id, +}; + +static struct platform_device *neo1973_snd_device; + +static int __init neo1973_init(void) +{ + int ret; + + pr_debug("Entered %s\n", __func__); + + if (!machine_is_neo1973_gta01()) { + printk(KERN_INFO + "Only GTA01 hardware supported by ASoC driver\n"); + return -ENODEV; + } + + neo1973_snd_device = platform_device_alloc("soc-audio", -1); + if (!neo1973_snd_device) + return -ENOMEM; + + platform_set_drvdata(neo1973_snd_device, &neo1973_snd_devdata); + neo1973_snd_devdata.dev = &neo1973_snd_device->dev; + ret = platform_device_add(neo1973_snd_device); + + if (ret) { + platform_device_put(neo1973_snd_device); + return ret; + } + + ret = i2c_add_driver(&lm4857_i2c_driver); + + if (ret != 0) + platform_device_unregister(neo1973_snd_device); + + return ret; +} + +static void __exit neo1973_exit(void) +{ + pr_debug("Entered %s\n", __func__); + + i2c_del_driver(&lm4857_i2c_driver); + platform_device_unregister(neo1973_snd_device); +} + +module_init(neo1973_init); +module_exit(neo1973_exit); + +/* Module information */ +MODULE_AUTHOR("Graeme Gregory, graeme(a)openmoko.org, www.openmoko.org"); +MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/regs-i2s-v2.h b/sound/soc/samsung/regs-i2s-v2.h new file mode 100644 index 0000000..5e5e568 --- /dev/null +++ b/sound/soc/samsung/regs-i2s-v2.h @@ -0,0 +1,115 @@ +/* linux/include/asm-arm/plat-s3c24xx/regs-s3c2412-iis.h + * + * Copyright 2007 Simtec Electronics <linux(a)simtec.co.uk> + * http://armlinux.simtec.co.uk/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * S3C2412 IIS register definition +*/ + +#ifndef __ASM_ARCH_REGS_S3C2412_IIS_H +#define __ASM_ARCH_REGS_S3C2412_IIS_H + +#define S3C2412_IISCON (0x00) +#define S3C2412_IISMOD (0x04) +#define S3C2412_IISFIC (0x08) +#define S3C2412_IISPSR (0x0C) +#define S3C2412_IISTXD (0x10) +#define S3C2412_IISRXD (0x14) + +#define S5PC1XX_IISFICS 0x18 +#define S5PC1XX_IISTXDS 0x1C + +#define S5PC1XX_IISCON_SW_RST (1 << 31) +#define S5PC1XX_IISCON_FRXOFSTATUS (1 << 26) +#define S5PC1XX_IISCON_FRXORINTEN (1 << 25) +#define S5PC1XX_IISCON_FTXSURSTAT (1 << 24) +#define S5PC1XX_IISCON_FTXSURINTEN (1 << 23) +#define S5PC1XX_IISCON_TXSDMAPAUSE (1 << 20) +#define S5PC1XX_IISCON_TXSDMACTIVE (1 << 18) + +#define S3C64XX_IISCON_FTXURSTATUS (1 << 17) +#define S3C64XX_IISCON_FTXURINTEN (1 << 16) +#define S3C64XX_IISCON_TXFIFO2_EMPTY (1 << 15) +#define S3C64XX_IISCON_TXFIFO1_EMPTY (1 << 14) +#define S3C64XX_IISCON_TXFIFO2_FULL (1 << 13) +#define S3C64XX_IISCON_TXFIFO1_FULL (1 << 12) + +#define S3C2412_IISCON_LRINDEX (1 << 11) +#define S3C2412_IISCON_TXFIFO_EMPTY (1 << 10) +#define S3C2412_IISCON_RXFIFO_EMPTY (1 << 9) +#define S3C2412_IISCON_TXFIFO_FULL (1 << 8) +#define S3C2412_IISCON_RXFIFO_FULL (1 << 7) +#define S3C2412_IISCON_TXDMA_PAUSE (1 << 6) +#define S3C2412_IISCON_RXDMA_PAUSE (1 << 5) +#define S3C2412_IISCON_TXCH_PAUSE (1 << 4) +#define S3C2412_IISCON_RXCH_PAUSE (1 << 3) +#define S3C2412_IISCON_TXDMA_ACTIVE (1 << 2) +#define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1) +#define S3C2412_IISCON_IIS_ACTIVE (1 << 0) + +#define S5PC1XX_IISMOD_OPCLK_CDCLK_OUT (0 << 30) +#define S5PC1XX_IISMOD_OPCLK_CDCLK_IN (1 << 30) +#define S5PC1XX_IISMOD_OPCLK_BCLK_OUT (2 << 30) +#define S5PC1XX_IISMOD_OPCLK_PCLK (3 << 30) +#define S5PC1XX_IISMOD_OPCLK_MASK (3 << 30) +#define S5PC1XX_IISMOD_TXS_IDMA (1 << 28) /* Sec_TXFIFO use I-DMA */ +#define S5PC1XX_IISMOD_BLCS_MASK 0x3 +#define S5PC1XX_IISMOD_BLCS_SHIFT 26 +#define S5PC1XX_IISMOD_BLCP_MASK 0x3 +#define S5PC1XX_IISMOD_BLCP_SHIFT 24 + +#define S3C64XX_IISMOD_C2DD_HHALF (1 << 21) /* Discard Higher-half */ +#define S3C64XX_IISMOD_C2DD_LHALF (1 << 20) /* Discard Lower-half */ +#define S3C64XX_IISMOD_C1DD_HHALF (1 << 19) +#define S3C64XX_IISMOD_C1DD_LHALF (1 << 18) +#define S3C64XX_IISMOD_DC2_EN (1 << 17) +#define S3C64XX_IISMOD_DC1_EN (1 << 16) +#define S3C64XX_IISMOD_BLC_16BIT (0 << 13) +#define S3C64XX_IISMOD_BLC_8BIT (1 << 13) +#define S3C64XX_IISMOD_BLC_24BIT (2 << 13) +#define S3C64XX_IISMOD_BLC_MASK (3 << 13) + +#define S3C2412_IISMOD_IMS_SYSMUX (1 << 10) +#define S3C2412_IISMOD_SLAVE (1 << 11) +#define S3C2412_IISMOD_MODE_TXONLY (0 << 8) +#define S3C2412_IISMOD_MODE_RXONLY (1 << 8) +#define S3C2412_IISMOD_MODE_TXRX (2 << 8) +#define S3C2412_IISMOD_MODE_MASK (3 << 8) +#define S3C2412_IISMOD_LR_LLOW (0 << 7) +#define S3C2412_IISMOD_LR_RLOW (1 << 7) +#define S3C2412_IISMOD_SDF_IIS (0 << 5) +#define S3C2412_IISMOD_SDF_MSB (1 << 5) +#define S3C2412_IISMOD_SDF_LSB (2 << 5) +#define S3C2412_IISMOD_SDF_MASK (3 << 5) +#define S3C2412_IISMOD_RCLK_256FS (0 << 3) +#define S3C2412_IISMOD_RCLK_512FS (1 << 3) +#define S3C2412_IISMOD_RCLK_384FS (2 << 3) +#define S3C2412_IISMOD_RCLK_768FS (3 << 3) +#define S3C2412_IISMOD_RCLK_MASK (3 << 3) +#define S3C2412_IISMOD_BCLK_32FS (0 << 1) +#define S3C2412_IISMOD_BCLK_48FS (1 << 1) +#define S3C2412_IISMOD_BCLK_16FS (2 << 1) +#define S3C2412_IISMOD_BCLK_24FS (3 << 1) +#define S3C2412_IISMOD_BCLK_MASK (3 << 1) +#define S3C2412_IISMOD_8BIT (1 << 0) + +#define S3C64XX_IISMOD_CDCLKCON (1 << 12) + +#define S3C2412_IISPSR_PSREN (1 << 15) + +#define S3C64XX_IISFIC_TX2COUNT(x) (((x) >> 24) & 0xf) +#define S3C64XX_IISFIC_TX1COUNT(x) (((x) >> 16) & 0xf) + +#define S3C2412_IISFIC_TXFLUSH (1 << 15) +#define S3C2412_IISFIC_RXFLUSH (1 << 7) +#define S3C2412_IISFIC_TXCOUNT(x) (((x) >> 8) & 0xf) +#define S3C2412_IISFIC_RXCOUNT(x) (((x) >> 0) & 0xf) + +#define S5PC1XX_IISFICS_TXFLUSH (1 << 15) +#define S5PC1XX_IISFICS_TXCOUNT(x) (((x) >> 8) & 0x7f) + +#endif /* __ASM_ARCH_REGS_S3C2412_IIS_H */ diff --git a/sound/soc/samsung/s3c-ac97.c b/sound/soc/samsung/s3c-ac97.c new file mode 100644 index 0000000..ecf4fd0 --- /dev/null +++ b/sound/soc/samsung/s3c-ac97.c @@ -0,0 +1,521 @@ +/* sound/soc/s3c24xx/s3c-ac97.c + * + * ALSA SoC Audio Layer - S3C AC97 Controller driver + * Evolved from s3c2443-ac97.c + * + * Copyright (c) 2010 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * Credits: Graeme Gregory, Sean Choi + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/clk.h> + +#include <sound/soc.h> + +#include <plat/regs-ac97.h> +#include <mach/dma.h> +#include <plat/audio.h> + +#include "s3c-dma.h" +#include "s3c-ac97.h" + +#define AC_CMD_ADDR(x) (x << 16) +#define AC_CMD_DATA(x) (x & 0xffff) + +struct s3c_ac97_info { + unsigned state; + struct clk *ac97_clk; + void __iomem *regs; + struct mutex lock; + struct completion done; +}; +static struct s3c_ac97_info s3c_ac97; + +static struct s3c2410_dma_client s3c_dma_client_out = { + .name = "AC97 PCMOut" +}; + +static struct s3c2410_dma_client s3c_dma_client_in = { + .name = "AC97 PCMIn" +}; + +static struct s3c2410_dma_client s3c_dma_client_micin = { + .name = "AC97 MicIn" +}; + +static struct s3c_dma_params s3c_ac97_pcm_out = { + .client = &s3c_dma_client_out, + .dma_size = 4, +}; + +static struct s3c_dma_params s3c_ac97_pcm_in = { + .client = &s3c_dma_client_in, + .dma_size = 4, +}; + +static struct s3c_dma_params s3c_ac97_mic_in = { + .client = &s3c_dma_client_micin, + .dma_size = 4, +}; + +static void s3c_ac97_activate(struct snd_ac97 *ac97) +{ + u32 ac_glbctrl, stat; + + stat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT) & 0x7; + if (stat == S3C_AC97_GLBSTAT_MAINSTATE_ACTIVE) + return; /* Return if already active */ + + INIT_COMPLETION(s3c_ac97.done); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl = S3C_AC97_GLBCTRL_ACLINKON; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) + printk(KERN_ERR "AC97: Unable to activate!"); +} + +static unsigned short s3c_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + u32 ac_glbctrl, ac_codec_cmd; + u32 stat, addr, data; + + mutex_lock(&s3c_ac97.lock); + + s3c_ac97_activate(ac97); + + INIT_COMPLETION(s3c_ac97.done); + + ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); + ac_codec_cmd = S3C_AC97_CODEC_CMD_READ | AC_CMD_ADDR(reg); + writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); + + udelay(50); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) + printk(KERN_ERR "AC97: Unable to read!"); + + stat = readl(s3c_ac97.regs + S3C_AC97_STAT); + addr = (stat >> 16) & 0x7f; + data = (stat & 0xffff); + + if (addr != reg) + printk(KERN_ERR "s3c-ac97: req addr = %02x, rep addr = %02x\n", reg, addr); + + mutex_unlock(&s3c_ac97.lock); + + return (unsigned short)data; +} + +static void s3c_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + u32 ac_glbctrl, ac_codec_cmd; + + mutex_lock(&s3c_ac97.lock); + + s3c_ac97_activate(ac97); + + INIT_COMPLETION(s3c_ac97.done); + + ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); + ac_codec_cmd = AC_CMD_ADDR(reg) | AC_CMD_DATA(val); + writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); + + udelay(50); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + if (!wait_for_completion_timeout(&s3c_ac97.done, HZ)) + printk(KERN_ERR "AC97: Unable to write!"); + + ac_codec_cmd = readl(s3c_ac97.regs + S3C_AC97_CODEC_CMD); + ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ; + writel(ac_codec_cmd, s3c_ac97.regs + S3C_AC97_CODEC_CMD); + + mutex_unlock(&s3c_ac97.lock); +} + +static void s3c_ac97_cold_reset(struct snd_ac97 *ac97) +{ + writel(S3C_AC97_GLBCTRL_COLDRESET, + s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + writel(0, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); +} + +static void s3c_ac97_warm_reset(struct snd_ac97 *ac97) +{ + u32 stat; + + stat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT) & 0x7; + if (stat == S3C_AC97_GLBSTAT_MAINSTATE_ACTIVE) + return; /* Return if already active */ + + writel(S3C_AC97_GLBCTRL_WARMRESET, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + writel(0, s3c_ac97.regs + S3C_AC97_GLBCTRL); + msleep(1); + + s3c_ac97_activate(ac97); +} + +static irqreturn_t s3c_ac97_irq(int irq, void *dev_id) +{ + u32 ac_glbctrl, ac_glbstat; + + ac_glbstat = readl(s3c_ac97.regs + S3C_AC97_GLBSTAT); + + if (ac_glbstat & S3C_AC97_GLBSTAT_CODECREADY) { + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl &= ~S3C_AC97_GLBCTRL_CODECREADYIE; + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + complete(&s3c_ac97.done); + } + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl |= (1<<30); /* Clear interrupt */ + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + return IRQ_HANDLED; +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = s3c_ac97_read, + .write = s3c_ac97_write, + .warm_reset = s3c_ac97_warm_reset, + .reset = s3c_ac97_cold_reset, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct s3c_dma_params *dma_data; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = &s3c_ac97_pcm_out; + else + dma_data = &s3c_ac97_pcm_in; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + + return 0; +} + +static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + u32 ac_glbctrl; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMINTM_MASK; + else + ac_glbctrl &= ~S3C_AC97_GLBCTRL_PCMOUTTM_MASK; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + ac_glbctrl |= S3C_AC97_GLBCTRL_PCMINTM_DMA; + else + ac_glbctrl |= S3C_AC97_GLBCTRL_PCMOUTTM_DMA; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + break; + } + + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); + + return 0; +} + +static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return -ENODEV; + else + snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); + + return 0; +} + +static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + u32 ac_glbctrl; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + + ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); + ac_glbctrl &= ~S3C_AC97_GLBCTRL_MICINTM_MASK; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ac_glbctrl |= S3C_AC97_GLBCTRL_MICINTM_DMA; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + break; + } + + writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); + + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); + + return 0; +} + +static struct snd_soc_dai_ops s3c_ac97_dai_ops = { + .hw_params = s3c_ac97_hw_params, + .trigger = s3c_ac97_trigger, +}; + +static struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { + .hw_params = s3c_ac97_hw_mic_params, + .trigger = s3c_ac97_mic_trigger, +}; + +struct snd_soc_dai s3c_ac97_dai[] = { + [S3C_AC97_DAI_PCM] = { + .name = "s3c-ac97", + .id = S3C_AC97_DAI_PCM, + .ac97_control = 1, + .playback = { + .stream_name = "AC97 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "AC97 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &s3c_ac97_dai_ops, + }, + [S3C_AC97_DAI_MIC] = { + .name = "s3c-ac97-mic", + .id = S3C_AC97_DAI_MIC, + .ac97_control = 1, + .capture = { + .stream_name = "AC97 Mic Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &s3c_ac97_mic_dai_ops, + }, +}; +EXPORT_SYMBOL_GPL(s3c_ac97_dai); + +static __devinit int s3c_ac97_probe(struct platform_device *pdev) +{ + struct resource *mem_res, *dmatx_res, *dmarx_res, *dmamic_res, *irq_res; + struct s3c_audio_pdata *ac97_pdata; + int ret; + + ac97_pdata = pdev->dev.platform_data; + if (!ac97_pdata || !ac97_pdata->cfg_gpio) { + dev_err(&pdev->dev, "cfg_gpio callback not provided!\n"); + return -EINVAL; + } + + /* Check for availability of necessary resource */ + dmatx_res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmatx_res) { + dev_err(&pdev->dev, "Unable to get AC97-TX dma resource\n"); + return -ENXIO; + } + + dmarx_res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmarx_res) { + dev_err(&pdev->dev, "Unable to get AC97-RX dma resource\n"); + return -ENXIO; + } + + dmamic_res = platform_get_resource(pdev, IORESOURCE_DMA, 2); + if (!dmamic_res) { + dev_err(&pdev->dev, "Unable to get AC97-MIC dma resource\n"); + return -ENXIO; + } + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem_res) { + dev_err(&pdev->dev, "Unable to get register resource\n"); + return -ENXIO; + } + + irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!irq_res) { + dev_err(&pdev->dev, "AC97 IRQ not provided!\n"); + return -ENXIO; + } + + if (!request_mem_region(mem_res->start, + resource_size(mem_res), "s3c-ac97")) { + dev_err(&pdev->dev, "Unable to request register region\n"); + return -EBUSY; + } + + s3c_ac97_pcm_out.channel = dmatx_res->start; + s3c_ac97_pcm_out.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; + s3c_ac97_pcm_in.channel = dmarx_res->start; + s3c_ac97_pcm_in.dma_addr = mem_res->start + S3C_AC97_PCM_DATA; + s3c_ac97_mic_in.channel = dmamic_res->start; + s3c_ac97_mic_in.dma_addr = mem_res->start + S3C_AC97_MIC_DATA; + + init_completion(&s3c_ac97.done); + mutex_init(&s3c_ac97.lock); + + s3c_ac97.regs = ioremap(mem_res->start, resource_size(mem_res)); + if (s3c_ac97.regs == NULL) { + dev_err(&pdev->dev, "Unable to ioremap register region\n"); + ret = -ENXIO; + goto err1; + } + + s3c_ac97.ac97_clk = clk_get(&pdev->dev, "ac97"); + if (IS_ERR(s3c_ac97.ac97_clk)) { + dev_err(&pdev->dev, "s3c-ac97 failed to get ac97_clock\n"); + ret = -ENODEV; + goto err2; + } + clk_enable(s3c_ac97.ac97_clk); + + if (ac97_pdata->cfg_gpio(pdev)) { + dev_err(&pdev->dev, "Unable to configure gpio\n"); + ret = -EINVAL; + goto err3; + } + + ret = request_irq(irq_res->start, s3c_ac97_irq, + IRQF_DISABLED, "AC97", NULL); + if (ret < 0) { + printk(KERN_ERR "s3c-ac97: interrupt request failed.\n"); + goto err4; + } + + s3c_ac97_dai[S3C_AC97_DAI_PCM].dev = &pdev->dev; + s3c_ac97_dai[S3C_AC97_DAI_MIC].dev = &pdev->dev; + + ret = snd_soc_register_dais(s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); + if (ret) + goto err5; + + return 0; + +err5: + free_irq(irq_res->start, NULL); +err4: +err3: + clk_disable(s3c_ac97.ac97_clk); + clk_put(s3c_ac97.ac97_clk); +err2: + iounmap(s3c_ac97.regs); +err1: + release_mem_region(mem_res->start, resource_size(mem_res)); + + return ret; +} + +static __devexit int s3c_ac97_remove(struct platform_device *pdev) +{ + struct resource *mem_res, *irq_res; + + snd_soc_unregister_dais(s3c_ac97_dai, ARRAY_SIZE(s3c_ac97_dai)); + + irq_res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (irq_res) + free_irq(irq_res->start, NULL); + + clk_disable(s3c_ac97.ac97_clk); + clk_put(s3c_ac97.ac97_clk); + + iounmap(s3c_ac97.regs); + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (mem_res) + release_mem_region(mem_res->start, resource_size(mem_res)); + + return 0; +} + +static struct platform_driver s3c_ac97_driver = { + .probe = s3c_ac97_probe, + .remove = s3c_ac97_remove, + .driver = { + .name = "s3c-ac97", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c_ac97_init(void) +{ + return platform_driver_register(&s3c_ac97_driver); +} +module_init(s3c_ac97_init); + +static void __exit s3c_ac97_exit(void) +{ + platform_driver_unregister(&s3c_ac97_driver); +} +module_exit(s3c_ac97_exit); + +MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); +MODULE_DESCRIPTION("AC97 driver for the Samsung SoC"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c-ac97.h b/sound/soc/samsung/s3c-ac97.h new file mode 100644 index 0000000..2781983 --- /dev/null +++ b/sound/soc/samsung/s3c-ac97.h @@ -0,0 +1,23 @@ +/* sound/soc/s3c24xx/s3c-ac97.h + * + * ALSA SoC Audio Layer - S3C AC97 Controller driver + * Evolved from s3c2443-ac97.h + * + * Copyright (c) 2010 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * Credits: Graeme Gregory, Sean Choi + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __S3C_AC97_H_ +#define __S3C_AC97_H_ + +#define S3C_AC97_DAI_PCM 0 +#define S3C_AC97_DAI_MIC 1 + +extern struct snd_soc_dai s3c_ac97_dai[]; + +#endif /* __S3C_AC97_H_ */ diff --git a/sound/soc/samsung/s3c-dma.c b/sound/soc/samsung/s3c-dma.c new file mode 100644 index 0000000..1b61c23 --- /dev/null +++ b/sound/soc/samsung/s3c-dma.c @@ -0,0 +1,483 @@ +/* + * s3c-dma.c -- ALSA Soc Audio Layer + * + * (c) 2006 Wolfson Microelectronics PLC. + * Graeme Gregory graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * + * Copyright 2004-2005 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/dma.h> +#include <mach/hardware.h> +#include <mach/dma.h> + +#include "s3c-dma.h" + +static const struct snd_pcm_hardware s3c_dma_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 128*1024, + .period_bytes_min = PAGE_SIZE, + .period_bytes_max = PAGE_SIZE*2, + .periods_min = 2, + .periods_max = 128, + .fifo_size = 32, +}; + +struct s3c24xx_runtime_data { + spinlock_t lock; + int state; + unsigned int dma_loaded; + unsigned int dma_limit; + unsigned int dma_period; + dma_addr_t dma_start; + dma_addr_t dma_pos; + dma_addr_t dma_end; + struct s3c_dma_params *params; +}; + +/* s3c_dma_enqueue + * + * place a dma buffer onto the queue for the dma system + * to handle. +*/ +static void s3c_dma_enqueue(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + dma_addr_t pos = prtd->dma_pos; + unsigned int limit; + int ret; + + pr_debug("Entered %s\n", __func__); + + if (s3c_dma_has_circular()) + limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; + else + limit = prtd->dma_limit; + + pr_debug("%s: loaded %d, limit %d\n", + __func__, prtd->dma_loaded, limit); + + while (prtd->dma_loaded < limit) { + unsigned long len = prtd->dma_period; + + pr_debug("dma_loaded: %d\n", prtd->dma_loaded); + + if ((pos + len) > prtd->dma_end) { + len = prtd->dma_end - pos; + pr_debug(KERN_DEBUG "%s: corrected dma len %ld\n", + __func__, len); + } + + ret = s3c2410_dma_enqueue(prtd->params->channel, + substream, pos, len); + + if (ret == 0) { + prtd->dma_loaded++; + pos += prtd->dma_period; + if (pos >= prtd->dma_end) + pos = prtd->dma_start; + } else + break; + } + + prtd->dma_pos = pos; +} + +static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, + void *dev_id, int size, + enum s3c2410_dma_buffresult result) +{ + struct snd_pcm_substream *substream = dev_id; + struct s3c24xx_runtime_data *prtd; + + pr_debug("Entered %s\n", __func__); + + if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR) + return; + + prtd = substream->runtime->private_data; + + if (substream) + snd_pcm_period_elapsed(substream); + + spin_lock(&prtd->lock); + if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) { + prtd->dma_loaded--; + s3c_dma_enqueue(substream); + } + + spin_unlock(&prtd->lock); +} + +static int s3c_dma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + unsigned long totbytes = params_buffer_bytes(params); + struct s3c_dma_params *dma = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + int ret = 0; + + + pr_debug("Entered %s\n", __func__); + + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ + if (!dma) + return 0; + + /* this may get called several times by oss emulation + * with different params -HW */ + if (prtd->params == NULL) { + /* prepare DMA */ + prtd->params = dma; + + pr_debug("params %p, client %p, channel %d\n", prtd->params, + prtd->params->client, prtd->params->channel); + + ret = s3c2410_dma_request(prtd->params->channel, + prtd->params->client, NULL); + + if (ret < 0) { + printk(KERN_ERR "failed to get dma channel\n"); + return ret; + } + + /* use the circular buffering if we have it available. */ + if (s3c_dma_has_circular()) + s3c2410_dma_setflags(prtd->params->channel, + S3C2410_DMAF_CIRCULAR); + } + + s3c2410_dma_set_buffdone_fn(prtd->params->channel, + s3c24xx_audio_buffdone); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + runtime->dma_bytes = totbytes; + + spin_lock_irq(&prtd->lock); + prtd->dma_loaded = 0; + prtd->dma_limit = runtime->hw.periods_min; + prtd->dma_period = params_period_bytes(params); + prtd->dma_start = runtime->dma_addr; + prtd->dma_pos = prtd->dma_start; + prtd->dma_end = prtd->dma_start + totbytes; + spin_unlock_irq(&prtd->lock); + + return 0; +} + +static int s3c_dma_hw_free(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + + pr_debug("Entered %s\n", __func__); + + /* TODO - do we need to ensure DMA flushed */ + snd_pcm_set_runtime_buffer(substream, NULL); + + if (prtd->params) { + s3c2410_dma_free(prtd->params->channel, prtd->params->client); + prtd->params = NULL; + } + + return 0; +} + +static int s3c_dma_prepare(struct snd_pcm_substream *substream) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + /* return if this is a bufferless transfer e.g. + * codec <--> BT codec or GSM modem -- lg FIXME */ + if (!prtd->params) + return 0; + + /* channel needs configuring for mem=>device, increment memory addr, + * sync to pclk, half-word transfers to the IIS-FIFO. */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + s3c2410_dma_devconfig(prtd->params->channel, + S3C2410_DMASRC_MEM, + prtd->params->dma_addr); + } else { + s3c2410_dma_devconfig(prtd->params->channel, + S3C2410_DMASRC_HW, + prtd->params->dma_addr); + } + + s3c2410_dma_config(prtd->params->channel, + prtd->params->dma_size); + + /* flush the DMA channel */ + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_FLUSH); + prtd->dma_loaded = 0; + prtd->dma_pos = prtd->dma_start; + + /* enqueue dma buffers */ + s3c_dma_enqueue(substream); + + return ret; +} + +static int s3c_dma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + prtd->state |= ST_RUNNING; + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + prtd->state &= ~ST_RUNNING; + s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STOP); + break; + + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} + +static snd_pcm_uframes_t +s3c_dma_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd = runtime->private_data; + unsigned long res; + dma_addr_t src, dst; + + pr_debug("Entered %s\n", __func__); + + spin_lock(&prtd->lock); + s3c2410_dma_getposition(prtd->params->channel, &src, &dst); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + res = dst - prtd->dma_start; + else + res = src - prtd->dma_start; + + spin_unlock(&prtd->lock); + + pr_debug("Pointer %x %x\n", src, dst); + + /* we seem to be getting the odd error from the pcm library due + * to out-of-bounds pointers. this is maybe due to the dma engine + * not having loaded the new values for the channel before being + * callled... (todo - fix ) + */ + + if (res >= snd_pcm_lib_buffer_bytes(substream)) { + if (res == snd_pcm_lib_buffer_bytes(substream)) + res = 0; + } + + return bytes_to_frames(substream->runtime, res); +} + +static int s3c_dma_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd; + + pr_debug("Entered %s\n", __func__); + + snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + snd_soc_set_runtime_hwparams(substream, &s3c_dma_hardware); + + prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + return 0; +} + +static int s3c_dma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s3c24xx_runtime_data *prtd = runtime->private_data; + + pr_debug("Entered %s\n", __func__); + + if (!prtd) + pr_debug("s3c_dma_close called with prtd == NULL\n"); + + kfree(prtd); + + return 0; +} + +static int s3c_dma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + pr_debug("Entered %s\n", __func__); + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops s3c_dma_ops = { + .open = s3c_dma_open, + .close = s3c_dma_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = s3c_dma_hw_params, + .hw_free = s3c_dma_hw_free, + .prepare = s3c_dma_prepare, + .trigger = s3c_dma_trigger, + .pointer = s3c_dma_pointer, + .mmap = s3c_dma_mmap, +}; + +static int s3c_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = s3c_dma_hardware.buffer_bytes_max; + + pr_debug("Entered %s\n", __func__); + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + return 0; +} + +static void s3c_dma_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + pr_debug("Entered %s\n", __func__); + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static u64 s3c_dma_mask = DMA_BIT_MASK(32); + +static int s3c_dma_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int ret = 0; + + pr_debug("Entered %s\n", __func__); + + if (!card->dev->dma_mask) + card->dev->dma_mask = &s3c_dma_mask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = s3c_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + ret = s3c_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} + +struct snd_soc_platform s3c24xx_soc_platform = { + .name = "s3c24xx-audio", + .pcm_ops = &s3c_dma_ops, + .pcm_new = s3c_dma_new, + .pcm_free = s3c_dma_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); + +static int __init s3c24xx_soc_platform_init(void) +{ + return snd_soc_register_platform(&s3c24xx_soc_platform); +} +module_init(s3c24xx_soc_platform_init); + +static void __exit s3c24xx_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&s3c24xx_soc_platform); +} +module_exit(s3c24xx_soc_platform_exit); + +MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("Samsung S3C Audio DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c-dma.h b/sound/soc/samsung/s3c-dma.h new file mode 100644 index 0000000..69bb6bf --- /dev/null +++ b/sound/soc/samsung/s3c-dma.h @@ -0,0 +1,31 @@ +/* + * s3c-dma.h -- + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * ALSA PCM interface for the Samsung S3C24xx CPU + */ + +#ifndef _S3C_AUDIO_H +#define _S3C_AUDIO_H + +#define ST_RUNNING (1<<0) +#define ST_OPENED (1<<1) + +struct s3c_dma_params { + struct s3c2410_dma_client *client; /* stream identifier */ + int channel; /* Channel ID */ + dma_addr_t dma_addr; + int dma_size; /* Size of the DMA transfer */ +}; + +#define S3C24XX_DAI_I2S 0 + +/* platform data */ +extern struct snd_soc_platform s3c24xx_soc_platform; +extern struct snd_ac97_bus_ops s3c24xx_ac97_ops; + +#endif diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c new file mode 100644 index 0000000..13311c8 --- /dev/null +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -0,0 +1,776 @@ +/* sound/soc/s3c24xx/s3c-i2c-v2.c + * + * ALSA Soc Audio Layer - I2S core for newer Samsung SoCs. + * + * Copyright (c) 2006 Wolfson Microelectronics PLC. + * Graeme Gregory graeme.gregory(a)wolfsonmicro.com + * linux(a)wolfsonmicro.com + * + * Copyright (c) 2008, 2007, 2004-2005 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/io.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/dma.h> + +#include "regs-i2s-v2.h" +#include "s3c-i2s-v2.h" +#include "s3c-dma.h" + +#undef S3C_IIS_V2_SUPPORTED + +#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413) +#define S3C_IIS_V2_SUPPORTED +#endif + +#ifdef CONFIG_PLAT_S3C64XX +#define S3C_IIS_V2_SUPPORTED +#endif + +#ifndef S3C_IIS_V2_SUPPORTED +#error Unsupported CPU model +#endif + +#define S3C2412_I2S_DEBUG_CON 0 + +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +#define bit_set(v, b) (((v) & (b)) ? 1 : 0) + +#if S3C2412_I2S_DEBUG_CON +static void dbg_showcon(const char *fn, u32 con) +{ + printk(KERN_DEBUG "%s: LRI=%d, TXFEMPT=%d, RXFEMPT=%d, TXFFULL=%d, RXFFULL=%d\n", fn, + bit_set(con, S3C2412_IISCON_LRINDEX), + bit_set(con, S3C2412_IISCON_TXFIFO_EMPTY), + bit_set(con, S3C2412_IISCON_RXFIFO_EMPTY), + bit_set(con, S3C2412_IISCON_TXFIFO_FULL), + bit_set(con, S3C2412_IISCON_RXFIFO_FULL)); + + printk(KERN_DEBUG "%s: PAUSE: TXDMA=%d, RXDMA=%d, TXCH=%d, RXCH=%d\n", + fn, + bit_set(con, S3C2412_IISCON_TXDMA_PAUSE), + bit_set(con, S3C2412_IISCON_RXDMA_PAUSE), + bit_set(con, S3C2412_IISCON_TXCH_PAUSE), + bit_set(con, S3C2412_IISCON_RXCH_PAUSE)); + printk(KERN_DEBUG "%s: ACTIVE: TXDMA=%d, RXDMA=%d, IIS=%d\n", fn, + bit_set(con, S3C2412_IISCON_TXDMA_ACTIVE), + bit_set(con, S3C2412_IISCON_RXDMA_ACTIVE), + bit_set(con, S3C2412_IISCON_IIS_ACTIVE)); +} +#else +static inline void dbg_showcon(const char *fn, u32 con) +{ +} +#endif + + +/* Turn on or off the transmission path. */ +static void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on) +{ + void __iomem *regs = i2s->regs; + u32 fic, con, mod; + + pr_debug("%s(%d)\n", __func__, on); + + fic = readl(regs + S3C2412_IISFIC); + con = readl(regs + S3C2412_IISCON); + mod = readl(regs + S3C2412_IISMOD); + + pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); + + if (on) { + con |= S3C2412_IISCON_TXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; + con &= ~S3C2412_IISCON_TXDMA_PAUSE; + con &= ~S3C2412_IISCON_TXCH_PAUSE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_TXONLY: + case S3C2412_IISMOD_MODE_TXRX: + /* do nothing, we are in the right mode */ + break; + + case S3C2412_IISMOD_MODE_RXONLY: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_TXRX; + break; + + default: + dev_err(i2s->dev, "TXEN: Invalid MODE %x in IISMOD\n", + mod & S3C2412_IISMOD_MODE_MASK); + break; + } + + writel(con, regs + S3C2412_IISCON); + writel(mod, regs + S3C2412_IISMOD); + } else { + /* Note, we do not have any indication that the FIFO problems + * tha the S3C2410/2440 had apply here, so we should be able + * to disable the DMA and TX without resetting the FIFOS. + */ + + con |= S3C2412_IISCON_TXDMA_PAUSE; + con |= S3C2412_IISCON_TXCH_PAUSE; + con &= ~S3C2412_IISCON_TXDMA_ACTIVE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_TXRX: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_RXONLY; + break; + + case S3C2412_IISMOD_MODE_TXONLY: + mod &= ~S3C2412_IISMOD_MODE_MASK; + con &= ~S3C2412_IISCON_IIS_ACTIVE; + break; + + default: + dev_err(i2s->dev, "TXDIS: Invalid MODE %x in IISMOD\n", + mod & S3C2412_IISMOD_MODE_MASK); + break; + } + + writel(mod, regs + S3C2412_IISMOD); + writel(con, regs + S3C2412_IISCON); + } + + fic = readl(regs + S3C2412_IISFIC); + dbg_showcon(__func__, con); + pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); +} + +static void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on) +{ + void __iomem *regs = i2s->regs; + u32 fic, con, mod; + + pr_debug("%s(%d)\n", __func__, on); + + fic = readl(regs + S3C2412_IISFIC); + con = readl(regs + S3C2412_IISCON); + mod = readl(regs + S3C2412_IISMOD); + + pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); + + if (on) { + con |= S3C2412_IISCON_RXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; + con &= ~S3C2412_IISCON_RXDMA_PAUSE; + con &= ~S3C2412_IISCON_RXCH_PAUSE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_TXRX: + case S3C2412_IISMOD_MODE_RXONLY: + /* do nothing, we are in the right mode */ + break; + + case S3C2412_IISMOD_MODE_TXONLY: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_TXRX; + break; + + default: + dev_err(i2s->dev, "RXEN: Invalid MODE %x in IISMOD\n", + mod & S3C2412_IISMOD_MODE_MASK); + } + + writel(mod, regs + S3C2412_IISMOD); + writel(con, regs + S3C2412_IISCON); + } else { + /* See txctrl notes on FIFOs. */ + + con &= ~S3C2412_IISCON_RXDMA_ACTIVE; + con |= S3C2412_IISCON_RXDMA_PAUSE; + con |= S3C2412_IISCON_RXCH_PAUSE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_RXONLY: + con &= ~S3C2412_IISCON_IIS_ACTIVE; + mod &= ~S3C2412_IISMOD_MODE_MASK; + break; + + case S3C2412_IISMOD_MODE_TXRX: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_TXONLY; + break; + + default: + dev_err(i2s->dev, "RXDIS: Invalid MODE %x in IISMOD\n", + mod & S3C2412_IISMOD_MODE_MASK); + } + + writel(con, regs + S3C2412_IISCON); + writel(mod, regs + S3C2412_IISMOD); + } + + fic = readl(regs + S3C2412_IISFIC); + pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); +} + +#define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t) + +/* + * Wait for the LR signal to allow synchronisation to the L/R clock + * from the codec. May only be needed for slave mode. + */ +static int s3c2412_snd_lrsync(struct s3c_i2sv2_info *i2s) +{ + u32 iiscon; + unsigned long loops = msecs_to_loops(5); + + pr_debug("Entered %s\n", __func__); + + while (--loops) { + iiscon = readl(i2s->regs + S3C2412_IISCON); + if (iiscon & S3C2412_IISCON_LRINDEX) + break; + + cpu_relax(); + } + + if (!loops) { + printk(KERN_ERR "%s: timeout\n", __func__); + return -ETIMEDOUT; + } + + return 0; +} + +/* + * Set S3C2412 I2S DAI format + */ +static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + iismod = readl(i2s->regs + S3C2412_IISMOD); + pr_debug("hw_params r: IISMOD: %x \n", iismod); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + i2s->master = 0; + iismod |= S3C2412_IISMOD_SLAVE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + i2s->master = 1; + iismod &= ~S3C2412_IISMOD_SLAVE; + break; + default: + pr_err("unknwon master/slave format\n"); + return -EINVAL; + } + + iismod &= ~S3C2412_IISMOD_SDF_MASK; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + iismod |= S3C2412_IISMOD_LR_RLOW; + iismod |= S3C2412_IISMOD_SDF_MSB; + break; + case SND_SOC_DAIFMT_LEFT_J: + iismod |= S3C2412_IISMOD_LR_RLOW; + iismod |= S3C2412_IISMOD_SDF_LSB; + break; + case SND_SOC_DAIFMT_I2S: + iismod &= ~S3C2412_IISMOD_LR_RLOW; + iismod |= S3C2412_IISMOD_SDF_IIS; + break; + default: + pr_err("Unknown data format\n"); + return -EINVAL; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + pr_debug("hw_params w: IISMOD: %x \n", iismod); + return 0; +} + +static int s3c_i2sv2_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *socdai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_link *dai = rtd->dai; + struct s3c_i2sv2_info *i2s = to_info(dai->cpu_dai); + struct s3c_dma_params *dma_data; + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = i2s->dma_playback; + else + dma_data = i2s->dma_capture; + + snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data); + + /* Working copies of register */ + iismod = readl(i2s->regs + S3C2412_IISMOD); + pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); + + iismod &= ~S3C64XX_IISMOD_BLC_MASK; + /* Sample size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + iismod |= S3C64XX_IISMOD_BLC_8BIT; + break; + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S24_LE: + iismod |= S3C64XX_IISMOD_BLC_24BIT; + break; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + pr_debug("%s: w: IISMOD: %x\n", __func__, iismod); + + return 0; +} + +static int s3c_i2sv2_set_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 iismod = readl(i2s->regs + S3C2412_IISMOD); + + pr_debug("Entered %s\n", __func__); + pr_debug("%s r: IISMOD: %x\n", __func__, iismod); + + switch (clk_id) { + case S3C_I2SV2_CLKSRC_PCLK: + iismod &= ~S3C2412_IISMOD_IMS_SYSMUX; + break; + + case S3C_I2SV2_CLKSRC_AUDIOBUS: + iismod |= S3C2412_IISMOD_IMS_SYSMUX; + break; + + case S3C_I2SV2_CLKSRC_CDCLK: + /* Error if controller doesn't have the CDCLKCON bit */ + if (!(i2s->feature & S3C_FEATURE_CDCLKCON)) + return -EINVAL; + + switch (dir) { + case SND_SOC_CLOCK_IN: + iismod |= S3C64XX_IISMOD_CDCLKCON; + break; + case SND_SOC_CLOCK_OUT: + iismod &= ~S3C64XX_IISMOD_CDCLKCON; + break; + default: + return -EINVAL; + } + break; + + default: + return -EINVAL; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + pr_debug("%s w: IISMOD: %x\n", __func__, iismod); + + return 0; +} + +static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_i2sv2_info *i2s = to_info(rtd->dai->cpu_dai); + int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + unsigned long irqs; + int ret = 0; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + + pr_debug("Entered %s\n", __func__); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* On start, ensure that the FIFOs are cleared and reset. */ + + writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH, + i2s->regs + S3C2412_IISFIC); + + /* clear again, just in case */ + writel(0x0, i2s->regs + S3C2412_IISFIC); + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!i2s->master) { + ret = s3c2412_snd_lrsync(i2s); + if (ret) + goto exit_err; + } + + local_irq_save(irqs); + + if (capture) + s3c2412_snd_rxctrl(i2s, 1); + else + s3c2412_snd_txctrl(i2s, 1); + + local_irq_restore(irqs); + + /* + * Load the next buffer to DMA to meet the reqirement + * of the auto reload mechanism of S3C24XX. + * This call won't bother S3C64XX. + */ + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); + + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + local_irq_save(irqs); + + if (capture) + s3c2412_snd_rxctrl(i2s, 0); + else + s3c2412_snd_txctrl(i2s, 0); + + local_irq_restore(irqs); + break; + default: + ret = -EINVAL; + break; + } + +exit_err: + return ret; +} + +/* + * Set S3C2412 Clock dividers + */ +static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 reg; + + pr_debug("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div); + + switch (div_id) { + case S3C_I2SV2_DIV_BCLK: + switch (div) { + case 16: + div = S3C2412_IISMOD_BCLK_16FS; + break; + + case 32: + div = S3C2412_IISMOD_BCLK_32FS; + break; + + case 24: + div = S3C2412_IISMOD_BCLK_24FS; + break; + + case 48: + div = S3C2412_IISMOD_BCLK_48FS; + break; + + default: + return -EINVAL; + } + + reg = readl(i2s->regs + S3C2412_IISMOD); + reg &= ~S3C2412_IISMOD_BCLK_MASK; + writel(reg | div, i2s->regs + S3C2412_IISMOD); + + pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); + break; + + case S3C_I2SV2_DIV_RCLK: + switch (div) { + case 256: + div = S3C2412_IISMOD_RCLK_256FS; + break; + + case 384: + div = S3C2412_IISMOD_RCLK_384FS; + break; + + case 512: + div = S3C2412_IISMOD_RCLK_512FS; + break; + + case 768: + div = S3C2412_IISMOD_RCLK_768FS; + break; + + default: + return -EINVAL; + } + + reg = readl(i2s->regs + S3C2412_IISMOD); + reg &= ~S3C2412_IISMOD_RCLK_MASK; + writel(reg | div, i2s->regs + S3C2412_IISMOD); + pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); + break; + + case S3C_I2SV2_DIV_PRESCALER: + if (div >= 0) { + writel((div << 8) | S3C2412_IISPSR_PSREN, + i2s->regs + S3C2412_IISPSR); + } else { + writel(0x0, i2s->regs + S3C2412_IISPSR); + } + pr_debug("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR)); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_sframes_t s3c2412_i2s_delay(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + u32 reg = readl(i2s->regs + S3C2412_IISFIC); + snd_pcm_sframes_t delay; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + delay = S3C2412_IISFIC_TXCOUNT(reg); + else + delay = S3C2412_IISFIC_RXCOUNT(reg); + + return delay; +} + +struct clk *s3c_i2sv2_get_clock(struct snd_soc_dai *cpu_dai) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + u32 iismod = readl(i2s->regs + S3C2412_IISMOD); + + if (iismod & S3C2412_IISMOD_IMS_SYSMUX) + return i2s->iis_cclk; + else + return i2s->iis_pclk; +} +EXPORT_SYMBOL_GPL(s3c_i2sv2_get_clock); + +/* default table of all avaialable root fs divisors */ +static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 }; + +int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, + unsigned int *fstab, + unsigned int rate, struct clk *clk) +{ + unsigned long clkrate = clk_get_rate(clk); + unsigned int div; + unsigned int fsclk; + unsigned int actual; + unsigned int fs; + unsigned int fsdiv; + signed int deviation = 0; + unsigned int best_fs = 0; + unsigned int best_div = 0; + unsigned int best_rate = 0; + unsigned int best_deviation = INT_MAX; + + pr_debug("Input clock rate %ldHz\n", clkrate); + + if (fstab == NULL) + fstab = iis_fs_tab; + + for (fs = 0; fs < ARRAY_SIZE(iis_fs_tab); fs++) { + fsdiv = iis_fs_tab[fs]; + + fsclk = clkrate / fsdiv; + div = fsclk / rate; + + if ((fsclk % rate) > (rate / 2)) + div++; + + if (div <= 1) + continue; + + actual = clkrate / (fsdiv * div); + deviation = actual - rate; + + printk(KERN_DEBUG "%ufs: div %u => result %u, deviation %d\n", + fsdiv, div, actual, deviation); + + deviation = abs(deviation); + + if (deviation < best_deviation) { + best_fs = fsdiv; + best_div = div; + best_rate = actual; + best_deviation = deviation; + } + + if (deviation == 0) + break; + } + + printk(KERN_DEBUG "best: fs=%u, div=%u, rate=%u\n", + best_fs, best_div, best_rate); + + info->fs_div = best_fs; + info->clk_div = best_div; + + return 0; +} +EXPORT_SYMBOL_GPL(s3c_i2sv2_iis_calc_rate); + +int s3c_i2sv2_probe(struct platform_device *pdev, + struct snd_soc_dai *dai, + struct s3c_i2sv2_info *i2s, + unsigned long base) +{ + struct device *dev = &pdev->dev; + unsigned int iismod; + + i2s->dev = dev; + + /* record our i2s structure for later use in the callbacks */ + dai->private_data = i2s; + + if (!base) { + struct resource *res = platform_get_resource(pdev, + IORESOURCE_MEM, + 0); + if (!res) { + dev_err(dev, "Unable to get register resource\n"); + return -ENXIO; + } + + if (!request_mem_region(res->start, resource_size(res), + "s3c64xx-i2s-v4")) { + dev_err(dev, "Unable to request register region\n"); + return -EBUSY; + } + + base = res->start; + } + + i2s->regs = ioremap(base, 0x100); + if (i2s->regs == NULL) { + dev_err(dev, "cannot ioremap registers\n"); + return -ENXIO; + } + + i2s->iis_pclk = clk_get(dev, "iis"); + if (IS_ERR(i2s->iis_pclk)) { + dev_err(dev, "failed to get iis_clock\n"); + iounmap(i2s->regs); + return -ENOENT; + } + + clk_enable(i2s->iis_pclk); + + /* Mark ourselves as in TXRX mode so we can run through our cleanup + * process without warnings. */ + iismod = readl(i2s->regs + S3C2412_IISMOD); + iismod |= S3C2412_IISMOD_MODE_TXRX; + writel(iismod, i2s->regs + S3C2412_IISMOD); + s3c2412_snd_txctrl(i2s, 0); + s3c2412_snd_rxctrl(i2s, 0); + + return 0; +} +EXPORT_SYMBOL_GPL(s3c_i2sv2_probe); + +#ifdef CONFIG_PM +static int s3c2412_i2s_suspend(struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + u32 iismod; + + if (dai->active) { + i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); + i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); + i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); + + /* some basic suspend checks */ + + iismod = readl(i2s->regs + S3C2412_IISMOD); + + if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) + pr_warning("%s: RXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) + pr_warning("%s: TXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_IIS_ACTIVE) + pr_warning("%s: IIS active\n", __func__); + } + + return 0; +} + +static int s3c2412_i2s_resume(struct snd_soc_dai *dai) +{ + struct s3c_i2sv2_info *i2s = to_info(dai); + + pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n", + dai->active, i2s->suspend_iismod, i2s->suspend_iiscon); + + if (dai->active) { + writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); + writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); + writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); + + writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, + i2s->regs + S3C2412_IISFIC); + + ndelay(250); + writel(0x0, i2s->regs + S3C2412_IISFIC); + } + + return 0; +} +#else +#define s3c2412_i2s_suspend NULL +#define s3c2412_i2s_resume NULL +#endif + +int s3c_i2sv2_register_dai(struct snd_soc_dai *dai) +{ + struct snd_soc_dai_ops *ops = dai->ops; + + ops->trigger = s3c2412_i2s_trigger; + if (!ops->hw_params) + ops->hw_params = s3c_i2sv2_hw_params; + ops->set_fmt = s3c2412_i2s_set_fmt; + ops->set_clkdiv = s3c2412_i2s_set_clkdiv; + ops->set_sysclk = s3c_i2sv2_set_sysclk; + + /* Allow overriding by (for example) IISv4 */ + if (!ops->delay) + ops->delay = s3c2412_i2s_delay; + + dai->suspend = s3c2412_i2s_suspend; + dai->resume = s3c2412_i2s_resume; + + return snd_soc_register_dai(dai); +} +EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai); + +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c-i2s-v2.h b/sound/soc/samsung/s3c-i2s-v2.h new file mode 100644 index 0000000..766f43a --- /dev/null +++ b/sound/soc/samsung/s3c-i2s-v2.h @@ -0,0 +1,103 @@ +/* sound/soc/s3c24xx/s3c-i2s-v2.h + * + * ALSA Soc Audio Layer - S3C_I2SV2 I2S driver + * + * Copyright (c) 2007 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. +*/ + +/* This code is the core support for the I2S block found in a number of + * Samsung SoC devices which is unofficially named I2S-V2. Currently the + * S3C2412 and the S3C64XX series use this block to provide 1 or 2 I2S + * channels via configurable GPIO. + */ + +#ifndef __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H +#define __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H __FILE__ + +#define S3C_I2SV2_DIV_BCLK (1) +#define S3C_I2SV2_DIV_RCLK (2) +#define S3C_I2SV2_DIV_PRESCALER (3) + +#define S3C_I2SV2_CLKSRC_PCLK 0 +#define S3C_I2SV2_CLKSRC_AUDIOBUS 1 +#define S3C_I2SV2_CLKSRC_CDCLK 2 + +/* Set this flag for I2S controllers that have the bit IISMOD[12] + * bridge/break RCLK signal and external Xi2sCDCLK pin. + */ +#define S3C_FEATURE_CDCLKCON (1 << 0) + +/** + * struct s3c_i2sv2_info - S3C I2S-V2 information + * @dev: The parent device passed to use from the probe. + * @regs: The pointer to the device registe block. + * @feature: Set of bit-flags indicating features of the controller. + * @master: True if the I2S core is the I2S bit clock master. + * @dma_playback: DMA information for playback channel. + * @dma_capture: DMA information for capture channel. + * @suspend_iismod: PM save for the IISMOD register. + * @suspend_iiscon: PM save for the IISCON register. + * @suspend_iispsr: PM save for the IISPSR register. + * + * This is the private codec state for the hardware associated with an + * I2S channel such as the register mappings and clock sources. + */ +struct s3c_i2sv2_info { + struct device *dev; + void __iomem *regs; + + u32 feature; + + struct clk *iis_pclk; + struct clk *iis_cclk; + + unsigned char master; + + struct s3c_dma_params *dma_playback; + struct s3c_dma_params *dma_capture; + + u32 suspend_iismod; + u32 suspend_iiscon; + u32 suspend_iispsr; +}; + +extern struct clk *s3c_i2sv2_get_clock(struct snd_soc_dai *cpu_dai); + +struct s3c_i2sv2_rate_calc { + unsigned int clk_div; /* for prescaler */ + unsigned int fs_div; /* for root frame clock */ +}; + +extern int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, + unsigned int *fstab, + unsigned int rate, struct clk *clk); + +/** + * s3c_i2sv2_probe - probe for i2s device helper + * @pdev: The platform device supplied to the original probe. + * @dai: The ASoC DAI structure supplied to the original probe. + * @i2s: Our local i2s structure to fill in. + * @base: The base address for the registers. + */ +extern int s3c_i2sv2_probe(struct platform_device *pdev, + struct snd_soc_dai *dai, + struct s3c_i2sv2_info *i2s, + unsigned long base); + +/** + * s3c_i2sv2_register_dai - register dai with soc core + * @dai: The snd_soc_dai structure to register + * + * Fill in any missing fields and then register the given dai with the + * soc core. + */ +extern int s3c_i2sv2_register_dai(struct snd_soc_dai *dai); + +#endif /* __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H */ diff --git a/sound/soc/samsung/s3c-pcm.c b/sound/soc/samsung/s3c-pcm.c new file mode 100644 index 0000000..326f0a9 --- /dev/null +++ b/sound/soc/samsung/s3c-pcm.c @@ -0,0 +1,554 @@ +/* sound/soc/s3c24xx/s3c-pcm.c + * + * ALSA SoC Audio Layer - S3C PCM-Controller driver + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * based upon I2S drivers by Ben Dooks. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/kernel.h> +#include <linux/gpio.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <plat/audio.h> +#include <plat/dma.h> + +#include "s3c-dma.h" +#include "s3c-pcm.h" + +static struct s3c2410_dma_client s3c_pcm_dma_client_out = { + .name = "PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c_pcm_dma_client_in = { + .name = "PCM Stereo in" +}; + +static struct s3c_dma_params s3c_pcm_stereo_out[] = { + [0] = { + .client = &s3c_pcm_dma_client_out, + .dma_size = 4, + }, + [1] = { + .client = &s3c_pcm_dma_client_out, + .dma_size = 4, + }, +}; + +static struct s3c_dma_params s3c_pcm_stereo_in[] = { + [0] = { + .client = &s3c_pcm_dma_client_in, + .dma_size = 4, + }, + [1] = { + .client = &s3c_pcm_dma_client_in, + .dma_size = 4, + }, +}; + +static struct s3c_pcm_info s3c_pcm[2]; + +static inline struct s3c_pcm_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static void s3c_pcm_snd_txctrl(struct s3c_pcm_info *pcm, int on) +{ + void __iomem *regs = pcm->regs; + u32 ctl, clkctl; + + clkctl = readl(regs + S3C_PCM_CLKCTL); + ctl = readl(regs + S3C_PCM_CTL); + ctl &= ~(S3C_PCM_CTL_TXDIPSTICK_MASK + << S3C_PCM_CTL_TXDIPSTICK_SHIFT); + + if (on) { + ctl |= S3C_PCM_CTL_TXDMA_EN; + ctl |= S3C_PCM_CTL_TXFIFO_EN; + ctl |= S3C_PCM_CTL_ENABLE; + ctl |= (0x20<<S3C_PCM_CTL_TXDIPSTICK_SHIFT); + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } else { + ctl &= ~S3C_PCM_CTL_TXDMA_EN; + ctl &= ~S3C_PCM_CTL_TXFIFO_EN; + + if (!(ctl & S3C_PCM_CTL_RXFIFO_EN)) { + ctl &= ~S3C_PCM_CTL_ENABLE; + if (!pcm->idleclk) + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } + } + + writel(clkctl, regs + S3C_PCM_CLKCTL); + writel(ctl, regs + S3C_PCM_CTL); +} + +static void s3c_pcm_snd_rxctrl(struct s3c_pcm_info *pcm, int on) +{ + void __iomem *regs = pcm->regs; + u32 ctl, clkctl; + + ctl = readl(regs + S3C_PCM_CTL); + clkctl = readl(regs + S3C_PCM_CLKCTL); + + if (on) { + ctl |= S3C_PCM_CTL_RXDMA_EN; + ctl |= S3C_PCM_CTL_RXFIFO_EN; + ctl |= S3C_PCM_CTL_ENABLE; + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } else { + ctl &= ~S3C_PCM_CTL_RXDMA_EN; + ctl &= ~S3C_PCM_CTL_RXFIFO_EN; + + if (!(ctl & S3C_PCM_CTL_TXFIFO_EN)) { + ctl &= ~S3C_PCM_CTL_ENABLE; + if (!pcm->idleclk) + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } + } + + writel(clkctl, regs + S3C_PCM_CLKCTL); + writel(ctl, regs + S3C_PCM_CTL); +} + +static int s3c_pcm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_pcm_info *pcm = to_info(rtd->dai->cpu_dai); + unsigned long flags; + + dev_dbg(pcm->dev, "Entered %s\n", __func__); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&pcm->lock, flags); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + s3c_pcm_snd_rxctrl(pcm, 1); + else + s3c_pcm_snd_txctrl(pcm, 1); + + spin_unlock_irqrestore(&pcm->lock, flags); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&pcm->lock, flags); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + s3c_pcm_snd_rxctrl(pcm, 0); + else + s3c_pcm_snd_txctrl(pcm, 0); + + spin_unlock_irqrestore(&pcm->lock, flags); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *socdai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_link *dai = rtd->dai; + struct s3c_pcm_info *pcm = to_info(dai->cpu_dai); + struct s3c_dma_params *dma_data; + void __iomem *regs = pcm->regs; + struct clk *clk; + int sclk_div, sync_div; + unsigned long flags; + u32 clkctl; + + dev_dbg(pcm->dev, "Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = pcm->dma_playback; + else + dma_data = pcm->dma_capture; + + snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data); + + /* Strictly check for sample size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + default: + return -EINVAL; + } + + spin_lock_irqsave(&pcm->lock, flags); + + /* Get hold of the PCMSOURCE_CLK */ + clkctl = readl(regs + S3C_PCM_CLKCTL); + if (clkctl & S3C_PCM_CLKCTL_SERCLKSEL_PCLK) + clk = pcm->pclk; + else + clk = pcm->cclk; + + /* Set the SCLK divider */ + sclk_div = clk_get_rate(clk) / pcm->sclk_per_fs / + params_rate(params) / 2 - 1; + + clkctl &= ~(S3C_PCM_CLKCTL_SCLKDIV_MASK + << S3C_PCM_CLKCTL_SCLKDIV_SHIFT); + clkctl |= ((sclk_div & S3C_PCM_CLKCTL_SCLKDIV_MASK) + << S3C_PCM_CLKCTL_SCLKDIV_SHIFT); + + /* Set the SYNC divider */ + sync_div = pcm->sclk_per_fs - 1; + + clkctl &= ~(S3C_PCM_CLKCTL_SYNCDIV_MASK + << S3C_PCM_CLKCTL_SYNCDIV_SHIFT); + clkctl |= ((sync_div & S3C_PCM_CLKCTL_SYNCDIV_MASK) + << S3C_PCM_CLKCTL_SYNCDIV_SHIFT); + + writel(clkctl, regs + S3C_PCM_CLKCTL); + + spin_unlock_irqrestore(&pcm->lock, flags); + + dev_dbg(pcm->dev, "PCMSOURCE_CLK-%lu SCLK=%ufs SCLK_DIV=%d SYNC_DIV=%d\n", + clk_get_rate(clk), pcm->sclk_per_fs, + sclk_div, sync_div); + + return 0; +} + +static int s3c_pcm_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct s3c_pcm_info *pcm = to_info(cpu_dai); + void __iomem *regs = pcm->regs; + unsigned long flags; + int ret = 0; + u32 ctl; + + dev_dbg(pcm->dev, "Entered %s\n", __func__); + + spin_lock_irqsave(&pcm->lock, flags); + + ctl = readl(regs + S3C_PCM_CTL); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* Nothing to do, NB_NF by default */ + break; + default: + dev_err(pcm->dev, "Unsupported clock inversion!\n"); + ret = -EINVAL; + goto exit; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* Nothing to do, Master by default */ + break; + default: + dev_err(pcm->dev, "Unsupported master/slave format!\n"); + ret = -EINVAL; + goto exit; + } + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: + pcm->idleclk = 1; + break; + case SND_SOC_DAIFMT_GATED: + pcm->idleclk = 0; + break; + default: + dev_err(pcm->dev, "Invalid Clock gating request!\n"); + ret = -EINVAL; + goto exit; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + ctl |= S3C_PCM_CTL_TXMSB_AFTER_FSYNC; + ctl |= S3C_PCM_CTL_RXMSB_AFTER_FSYNC; + break; + case SND_SOC_DAIFMT_DSP_B: + ctl &= ~S3C_PCM_CTL_TXMSB_AFTER_FSYNC; + ctl &= ~S3C_PCM_CTL_RXMSB_AFTER_FSYNC; + break; + default: + dev_err(pcm->dev, "Unsupported data format!\n"); + ret = -EINVAL; + goto exit; + } + + writel(ctl, regs + S3C_PCM_CTL); + +exit: + spin_unlock_irqrestore(&pcm->lock, flags); + + return ret; +} + +static int s3c_pcm_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct s3c_pcm_info *pcm = to_info(cpu_dai); + + switch (div_id) { + case S3C_PCM_SCLK_PER_FS: + pcm->sclk_per_fs = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int s3c_pcm_set_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct s3c_pcm_info *pcm = to_info(cpu_dai); + void __iomem *regs = pcm->regs; + u32 clkctl = readl(regs + S3C_PCM_CLKCTL); + + switch (clk_id) { + case S3C_PCM_CLKSRC_PCLK: + clkctl |= S3C_PCM_CLKCTL_SERCLKSEL_PCLK; + break; + + case S3C_PCM_CLKSRC_MUX: + clkctl &= ~S3C_PCM_CLKCTL_SERCLKSEL_PCLK; + + if (clk_get_rate(pcm->cclk) != freq) + clk_set_rate(pcm->cclk, freq); + + break; + + default: + return -EINVAL; + } + + writel(clkctl, regs + S3C_PCM_CLKCTL); + + return 0; +} + +static struct snd_soc_dai_ops s3c_pcm_dai_ops = { + .set_sysclk = s3c_pcm_set_sysclk, + .set_clkdiv = s3c_pcm_set_clkdiv, + .trigger = s3c_pcm_trigger, + .hw_params = s3c_pcm_hw_params, + .set_fmt = s3c_pcm_set_fmt, +}; + +#define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 + +#define S3C_PCM_DECLARE(n) \ +{ \ + .name = "samsung-pcm", \ + .id = (n), \ + .symmetric_rates = 1, \ + .ops = &s3c_pcm_dai_ops, \ + .playback = { \ + .channels_min = 2, \ + .channels_max = 2, \ + .rates = S3C_PCM_RATES, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + }, \ + .capture = { \ + .channels_min = 2, \ + .channels_max = 2, \ + .rates = S3C_PCM_RATES, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + }, \ +} + +struct snd_soc_dai s3c_pcm_dai[] = { + S3C_PCM_DECLARE(0), + S3C_PCM_DECLARE(1), +}; +EXPORT_SYMBOL_GPL(s3c_pcm_dai); + +static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) +{ + struct s3c_pcm_info *pcm; + struct snd_soc_dai *dai; + struct resource *mem_res, *dmatx_res, *dmarx_res; + struct s3c_audio_pdata *pcm_pdata; + int ret; + + /* Check for valid device index */ + if ((pdev->id < 0) || pdev->id >= ARRAY_SIZE(s3c_pcm)) { + dev_err(&pdev->dev, "id %d out of range\n", pdev->id); + return -EINVAL; + } + + pcm_pdata = pdev->dev.platform_data; + + /* Check for availability of necessary resource */ + dmatx_res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmatx_res) { + dev_err(&pdev->dev, "Unable to get PCM-TX dma resource\n"); + return -ENXIO; + } + + dmarx_res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmarx_res) { + dev_err(&pdev->dev, "Unable to get PCM-RX dma resource\n"); + return -ENXIO; + } + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem_res) { + dev_err(&pdev->dev, "Unable to get register resource\n"); + return -ENXIO; + } + + if (pcm_pdata && pcm_pdata->cfg_gpio && pcm_pdata->cfg_gpio(pdev)) { + dev_err(&pdev->dev, "Unable to configure gpio\n"); + return -EINVAL; + } + + pcm = &s3c_pcm[pdev->id]; + pcm->dev = &pdev->dev; + + spin_lock_init(&pcm->lock); + + dai = &s3c_pcm_dai[pdev->id]; + dai->dev = &pdev->dev; + + /* Default is 128fs */ + pcm->sclk_per_fs = 128; + + pcm->cclk = clk_get(&pdev->dev, "audio-bus"); + if (IS_ERR(pcm->cclk)) { + dev_err(&pdev->dev, "failed to get audio-bus\n"); + ret = PTR_ERR(pcm->cclk); + goto err1; + } + clk_enable(pcm->cclk); + + /* record our pcm structure for later use in the callbacks */ + dai->private_data = pcm; + + if (!request_mem_region(mem_res->start, + resource_size(mem_res), "samsung-pcm")) { + dev_err(&pdev->dev, "Unable to request register region\n"); + ret = -EBUSY; + goto err2; + } + + pcm->regs = ioremap(mem_res->start, 0x100); + if (pcm->regs == NULL) { + dev_err(&pdev->dev, "cannot ioremap registers\n"); + ret = -ENXIO; + goto err3; + } + + pcm->pclk = clk_get(&pdev->dev, "pcm"); + if (IS_ERR(pcm->pclk)) { + dev_err(&pdev->dev, "failed to get pcm_clock\n"); + ret = -ENOENT; + goto err4; + } + clk_enable(pcm->pclk); + + ret = snd_soc_register_dai(dai); + if (ret != 0) { + dev_err(&pdev->dev, "failed to get pcm_clock\n"); + goto err5; + } + + s3c_pcm_stereo_in[pdev->id].dma_addr = mem_res->start + + S3C_PCM_RXFIFO; + s3c_pcm_stereo_out[pdev->id].dma_addr = mem_res->start + + S3C_PCM_TXFIFO; + + s3c_pcm_stereo_in[pdev->id].channel = dmarx_res->start; + s3c_pcm_stereo_out[pdev->id].channel = dmatx_res->start; + + pcm->dma_capture = &s3c_pcm_stereo_in[pdev->id]; + pcm->dma_playback = &s3c_pcm_stereo_out[pdev->id]; + + return 0; + +err5: + clk_disable(pcm->pclk); + clk_put(pcm->pclk); +err4: + iounmap(pcm->regs); +err3: + release_mem_region(mem_res->start, resource_size(mem_res)); +err2: + clk_disable(pcm->cclk); + clk_put(pcm->cclk); +err1: + return ret; +} + +static __devexit int s3c_pcm_dev_remove(struct platform_device *pdev) +{ + struct s3c_pcm_info *pcm = &s3c_pcm[pdev->id]; + struct resource *mem_res; + + iounmap(pcm->regs); + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(mem_res->start, resource_size(mem_res)); + + clk_disable(pcm->cclk); + clk_disable(pcm->pclk); + clk_put(pcm->pclk); + clk_put(pcm->cclk); + + return 0; +} + +static struct platform_driver s3c_pcm_driver = { + .probe = s3c_pcm_dev_probe, + .remove = s3c_pcm_dev_remove, + .driver = { + .name = "samsung-pcm", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c_pcm_init(void) +{ + return platform_driver_register(&s3c_pcm_driver); +} +module_init(s3c_pcm_init); + +static void __exit s3c_pcm_exit(void) +{ + platform_driver_unregister(&s3c_pcm_driver); +} +module_exit(s3c_pcm_exit); + +/* Module information */ +MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); +MODULE_DESCRIPTION("S3C PCM Controller Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c-pcm.h b/sound/soc/samsung/s3c-pcm.h new file mode 100644 index 0000000..69ff997 --- /dev/null +++ b/sound/soc/samsung/s3c-pcm.h @@ -0,0 +1,123 @@ +/* sound/soc/s3c24xx/s3c-pcm.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef __S3C_PCM_H +#define __S3C_PCM_H __FILE__ + +/*Register Offsets */ +#define S3C_PCM_CTL (0x00) +#define S3C_PCM_CLKCTL (0x04) +#define S3C_PCM_TXFIFO (0x08) +#define S3C_PCM_RXFIFO (0x0C) +#define S3C_PCM_IRQCTL (0x10) +#define S3C_PCM_IRQSTAT (0x14) +#define S3C_PCM_FIFOSTAT (0x18) +#define S3C_PCM_CLRINT (0x20) + +/* PCM_CTL Bit-Fields */ +#define S3C_PCM_CTL_TXDIPSTICK_MASK (0x3f) +#define S3C_PCM_CTL_TXDIPSTICK_SHIFT (13) +#define S3C_PCM_CTL_RXDIPSTICK_MSK (0x3f<<7) +#define S3C_PCM_CTL_TXDMA_EN (0x1<<6) +#define S3C_PCM_CTL_RXDMA_EN (0x1<<5) +#define S3C_PCM_CTL_TXMSB_AFTER_FSYNC (0x1<<4) +#define S3C_PCM_CTL_RXMSB_AFTER_FSYNC (0x1<<3) +#define S3C_PCM_CTL_TXFIFO_EN (0x1<<2) +#define S3C_PCM_CTL_RXFIFO_EN (0x1<<1) +#define S3C_PCM_CTL_ENABLE (0x1<<0) + +/* PCM_CLKCTL Bit-Fields */ +#define S3C_PCM_CLKCTL_SERCLK_EN (0x1<<19) +#define S3C_PCM_CLKCTL_SERCLKSEL_PCLK (0x1<<18) +#define S3C_PCM_CLKCTL_SCLKDIV_MASK (0x1ff) +#define S3C_PCM_CLKCTL_SYNCDIV_MASK (0x1ff) +#define S3C_PCM_CLKCTL_SCLKDIV_SHIFT (9) +#define S3C_PCM_CLKCTL_SYNCDIV_SHIFT (0) + +/* PCM_TXFIFO Bit-Fields */ +#define S3C_PCM_TXFIFO_DVALID (0x1<<16) +#define S3C_PCM_TXFIFO_DATA_MSK (0xffff<<0) + +/* PCM_RXFIFO Bit-Fields */ +#define S3C_PCM_RXFIFO_DVALID (0x1<<16) +#define S3C_PCM_RXFIFO_DATA_MSK (0xffff<<0) + +/* PCM_IRQCTL Bit-Fields */ +#define S3C_PCM_IRQCTL_IRQEN (0x1<<14) +#define S3C_PCM_IRQCTL_WRDEN (0x1<<12) +#define S3C_PCM_IRQCTL_TXEMPTYEN (0x1<<11) +#define S3C_PCM_IRQCTL_TXALMSTEMPTYEN (0x1<<10) +#define S3C_PCM_IRQCTL_TXFULLEN (0x1<<9) +#define S3C_PCM_IRQCTL_TXALMSTFULLEN (0x1<<8) +#define S3C_PCM_IRQCTL_TXSTARVEN (0x1<<7) +#define S3C_PCM_IRQCTL_TXERROVRFLEN (0x1<<6) +#define S3C_PCM_IRQCTL_RXEMPTEN (0x1<<5) +#define S3C_PCM_IRQCTL_RXALMSTEMPTEN (0x1<<4) +#define S3C_PCM_IRQCTL_RXFULLEN (0x1<<3) +#define S3C_PCM_IRQCTL_RXALMSTFULLEN (0x1<<2) +#define S3C_PCM_IRQCTL_RXSTARVEN (0x1<<1) +#define S3C_PCM_IRQCTL_RXERROVRFLEN (0x1<<0) + +/* PCM_IRQSTAT Bit-Fields */ +#define S3C_PCM_IRQSTAT_IRQPND (0x1<<13) +#define S3C_PCM_IRQSTAT_WRD_XFER (0x1<<12) +#define S3C_PCM_IRQSTAT_TXEMPTY (0x1<<11) +#define S3C_PCM_IRQSTAT_TXALMSTEMPTY (0x1<<10) +#define S3C_PCM_IRQSTAT_TXFULL (0x1<<9) +#define S3C_PCM_IRQSTAT_TXALMSTFULL (0x1<<8) +#define S3C_PCM_IRQSTAT_TXSTARV (0x1<<7) +#define S3C_PCM_IRQSTAT_TXERROVRFL (0x1<<6) +#define S3C_PCM_IRQSTAT_RXEMPT (0x1<<5) +#define S3C_PCM_IRQSTAT_RXALMSTEMPT (0x1<<4) +#define S3C_PCM_IRQSTAT_RXFULL (0x1<<3) +#define S3C_PCM_IRQSTAT_RXALMSTFULL (0x1<<2) +#define S3C_PCM_IRQSTAT_RXSTARV (0x1<<1) +#define S3C_PCM_IRQSTAT_RXERROVRFL (0x1<<0) + +/* PCM_FIFOSTAT Bit-Fields */ +#define S3C_PCM_FIFOSTAT_TXCNT_MSK (0x3f<<14) +#define S3C_PCM_FIFOSTAT_TXFIFOEMPTY (0x1<<13) +#define S3C_PCM_FIFOSTAT_TXFIFOALMSTEMPTY (0x1<<12) +#define S3C_PCM_FIFOSTAT_TXFIFOFULL (0x1<<11) +#define S3C_PCM_FIFOSTAT_TXFIFOALMSTFULL (0x1<<10) +#define S3C_PCM_FIFOSTAT_RXCNT_MSK (0x3f<<4) +#define S3C_PCM_FIFOSTAT_RXFIFOEMPTY (0x1<<3) +#define S3C_PCM_FIFOSTAT_RXFIFOALMSTEMPTY (0x1<<2) +#define S3C_PCM_FIFOSTAT_RXFIFOFULL (0x1<<1) +#define S3C_PCM_FIFOSTAT_RXFIFOALMSTFULL (0x1<<0) + +#define S3C_PCM_CLKSRC_PCLK 0 +#define S3C_PCM_CLKSRC_MUX 1 + +#define S3C_PCM_SCLK_PER_FS 0 + +/** + * struct s3c_pcm_info - S3C PCM Controller information + * @dev: The parent device passed to use from the probe. + * @regs: The pointer to the device register block. + * @dma_playback: DMA information for playback channel. + * @dma_capture: DMA information for capture channel. + */ +struct s3c_pcm_info { + spinlock_t lock; + struct device *dev; + void __iomem *regs; + + unsigned int sclk_per_fs; + + /* Whether to keep PCMSCLK enabled even when idle(no active xfer) */ + unsigned int idleclk; + + struct clk *pclk; + struct clk *cclk; + + struct s3c_dma_params *dma_playback; + struct s3c_dma_params *dma_capture; +}; + +#endif /* __S3C_PCM_H */ diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c new file mode 100644 index 0000000..709adef --- /dev/null +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -0,0 +1,190 @@ +/* sound/soc/s3c24xx/s3c2412-i2s.c + * + * ALSA Soc Audio Layer - S3C2412 I2S driver + * + * Copyright (c) 2006 Wolfson Microelectronics PLC. + * Graeme Gregory graeme.gregory(a)wolfsonmicro.com + * linux(a)wolfsonmicro.com + * + * Copyright (c) 2007, 2004-2005 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <linux/kernel.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <mach/hardware.h> + +#include <mach/regs-gpio.h> +#include <mach/dma.h> + +#include "s3c-dma.h" +#include "regs-i2s-v2.h" +#include "s3c2412-i2s.h" + +#define S3C2412_I2S_DEBUG 0 + +static struct s3c2410_dma_client s3c2412_dma_client_out = { + .name = "I2S PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c2412_dma_client_in = { + .name = "I2S PCM Stereo in" +}; + +static struct s3c_dma_params s3c2412_i2s_pcm_stereo_out = { + .client = &s3c2412_dma_client_out, + .channel = DMACH_I2S_OUT, + .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD, + .dma_size = 4, +}; + +static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = { + .client = &s3c2412_dma_client_in, + .channel = DMACH_I2S_IN, + .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD, + .dma_size = 4, +}; + +static struct s3c_i2sv2_info s3c2412_i2s; + +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static int s3c2412_i2s_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + int ret; + + pr_debug("Entered %s\n", __func__); + + ret = s3c_i2sv2_probe(pdev, dai, &s3c2412_i2s, S3C2410_PA_IIS); + if (ret) + return ret; + + s3c2412_i2s.dma_capture = &s3c2412_i2s_pcm_stereo_in; + s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out; + + s3c2412_i2s.iis_cclk = clk_get(&pdev->dev, "i2sclk"); + if (s3c2412_i2s.iis_cclk == NULL) { + pr_err("failed to get i2sclk clock\n"); + iounmap(s3c2412_i2s.regs); + return -ENODEV; + } + + /* Set MPLL as the source for IIS CLK */ + + clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll")); + clk_enable(s3c2412_i2s.iis_cclk); + + s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk; + + /* Configure the I2S pins in correct mode */ + s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); + s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK); + s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK); + s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); + s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); + + return 0; +} + +static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + struct s3c_dma_params *dma_data; + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = i2s->dma_playback; + else + dma_data = i2s->dma_capture; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + + iismod = readl(i2s->regs + S3C2412_IISMOD); + pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + iismod |= S3C2412_IISMOD_8BIT; + break; + case SNDRV_PCM_FORMAT_S16_LE: + iismod &= ~S3C2412_IISMOD_8BIT; + break; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + pr_debug("%s: w: IISMOD: %x\n", __func__, iismod); + + return 0; +} + +#define S3C2412_I2S_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +static struct snd_soc_dai_ops s3c2412_i2s_dai_ops = { + .hw_params = s3c2412_i2s_hw_params, +}; + +struct snd_soc_dai s3c2412_i2s_dai = { + .name = "s3c2412-i2s", + .id = 0, + .probe = s3c2412_i2s_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C2412_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C2412_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &s3c2412_i2s_dai_ops, +}; +EXPORT_SYMBOL_GPL(s3c2412_i2s_dai); + +static int __init s3c2412_i2s_init(void) +{ + return s3c_i2sv2_register_dai(&s3c2412_i2s_dai); +} +module_init(s3c2412_i2s_init); + +static void __exit s3c2412_i2s_exit(void) +{ + snd_soc_unregister_dai(&s3c2412_i2s_dai); +} +module_exit(s3c2412_i2s_exit); + +/* Module information */ +MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("S3C2412 I2S SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c2412-i2s.h b/sound/soc/samsung/s3c2412-i2s.h new file mode 100644 index 0000000..0b5686b --- /dev/null +++ b/sound/soc/samsung/s3c2412-i2s.h @@ -0,0 +1,29 @@ +/* sound/soc/s3c24xx/s3c2412-i2s.c + * + * ALSA Soc Audio Layer - S3C2412 I2S driver + * + * Copyright (c) 2007 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. +*/ + +#ifndef __SND_SOC_S3C24XX_S3C2412_I2S_H +#define __SND_SOC_S3C24XX_S3C2412_I2S_H __FILE__ + +#include "s3c-i2s-v2.h" + +#define S3C2412_DIV_BCLK S3C_I2SV2_DIV_BCLK +#define S3C2412_DIV_RCLK S3C_I2SV2_DIV_RCLK +#define S3C2412_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER + +#define S3C2412_CLKSRC_PCLK S3C_I2SV2_CLKSRC_PCLK +#define S3C2412_CLKSRC_I2SCLK S3C_I2SV2_CLKSRC_AUDIOBUS + +extern struct snd_soc_dai s3c2412_i2s_dai; + +#endif /* __SND_SOC_S3C24XX_S3C2412_I2S_H */ diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c new file mode 100644 index 0000000..c3ac890 --- /dev/null +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -0,0 +1,503 @@ +/* + * s3c24xx-i2s.c -- ALSA Soc Audio Layer + * + * (c) 2006 Wolfson Microelectronics PLC. + * Graeme Gregory graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * + * Copyright 2004-2005 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/jiffies.h> +#include <linux/io.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <mach/hardware.h> +#include <mach/regs-gpio.h> +#include <mach/regs-clock.h> + +#include <asm/dma.h> +#include <mach/dma.h> + +#include <plat/regs-iis.h> + +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" + +static struct s3c2410_dma_client s3c24xx_dma_client_out = { + .name = "I2S PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c24xx_dma_client_in = { + .name = "I2S PCM Stereo in" +}; + +static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_out = { + .client = &s3c24xx_dma_client_out, + .channel = DMACH_I2S_OUT, + .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, + .dma_size = 2, +}; + +static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_in = { + .client = &s3c24xx_dma_client_in, + .channel = DMACH_I2S_IN, + .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, + .dma_size = 2, +}; + +struct s3c24xx_i2s_info { + void __iomem *regs; + struct clk *iis_clk; + u32 iiscon; + u32 iismod; + u32 iisfcon; + u32 iispsr; +}; +static struct s3c24xx_i2s_info s3c24xx_i2s; + +static void s3c24xx_snd_txctrl(int on) +{ + u32 iisfcon; + u32 iiscon; + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); + iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); + iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); + + pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); + + if (on) { + iisfcon |= S3C2410_IISFCON_TXDMA | S3C2410_IISFCON_TXENABLE; + iiscon |= S3C2410_IISCON_TXDMAEN | S3C2410_IISCON_IISEN; + iiscon &= ~S3C2410_IISCON_TXIDLE; + iismod |= S3C2410_IISMOD_TXMODE; + + writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); + writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); + } else { + /* note, we have to disable the FIFOs otherwise bad things + * seem to happen when the DMA stops. According to the + * Samsung supplied kernel, this should allow the DMA + * engine and FIFOs to reset. If this isn't allowed, the + * DMA engine will simply freeze randomly. + */ + + iisfcon &= ~S3C2410_IISFCON_TXENABLE; + iisfcon &= ~S3C2410_IISFCON_TXDMA; + iiscon |= S3C2410_IISCON_TXIDLE; + iiscon &= ~S3C2410_IISCON_TXDMAEN; + iismod &= ~S3C2410_IISMOD_TXMODE; + + writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); + writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); + writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + } + + pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); +} + +static void s3c24xx_snd_rxctrl(int on) +{ + u32 iisfcon; + u32 iiscon; + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); + iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); + iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); + + pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); + + if (on) { + iisfcon |= S3C2410_IISFCON_RXDMA | S3C2410_IISFCON_RXENABLE; + iiscon |= S3C2410_IISCON_RXDMAEN | S3C2410_IISCON_IISEN; + iiscon &= ~S3C2410_IISCON_RXIDLE; + iismod |= S3C2410_IISMOD_RXMODE; + + writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); + writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); + } else { + /* note, we have to disable the FIFOs otherwise bad things + * seem to happen when the DMA stops. According to the + * Samsung supplied kernel, this should allow the DMA + * engine and FIFOs to reset. If this isn't allowed, the + * DMA engine will simply freeze randomly. + */ + + iisfcon &= ~S3C2410_IISFCON_RXENABLE; + iisfcon &= ~S3C2410_IISFCON_RXDMA; + iiscon |= S3C2410_IISCON_RXIDLE; + iiscon &= ~S3C2410_IISCON_RXDMAEN; + iismod &= ~S3C2410_IISMOD_RXMODE; + + writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); + writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); + writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + } + + pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon); +} + +/* + * Wait for the LR signal to allow synchronisation to the L/R clock + * from the codec. May only be needed for slave mode. + */ +static int s3c24xx_snd_lrsync(void) +{ + u32 iiscon; + int timeout = 50; /* 5ms */ + + pr_debug("Entered %s\n", __func__); + + while (1) { + iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); + if (iiscon & S3C2410_IISCON_LRINDEX) + break; + + if (!timeout--) + return -ETIMEDOUT; + udelay(100); + } + + return 0; +} + +/* + * Check whether CPU is the master or slave + */ +static inline int s3c24xx_snd_is_clkmaster(void) +{ + pr_debug("Entered %s\n", __func__); + + return (readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & S3C2410_IISMOD_SLAVE) ? 0:1; +} + +/* + * Set S3C24xx I2S DAI format + */ +static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); + pr_debug("hw_params r: IISMOD: %x \n", iismod); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iismod |= S3C2410_IISMOD_SLAVE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iismod &= ~S3C2410_IISMOD_SLAVE; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + iismod |= S3C2410_IISMOD_MSB; + break; + case SND_SOC_DAIFMT_I2S: + iismod &= ~S3C2410_IISMOD_MSB; + break; + default: + return -EINVAL; + } + + writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + pr_debug("hw_params w: IISMOD: %x \n", iismod); + return 0; +} + +static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_dma_params *dma_data; + u32 iismod; + + pr_debug("Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = &s3c24xx_i2s_pcm_stereo_out; + else + dma_data = &s3c24xx_i2s_pcm_stereo_in; + + snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_data); + + /* Working copies of register */ + iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); + pr_debug("hw_params r: IISMOD: %x\n", iismod); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + iismod &= ~S3C2410_IISMOD_16BIT; + dma_data->dma_size = 1; + break; + case SNDRV_PCM_FORMAT_S16_LE: + iismod |= S3C2410_IISMOD_16BIT; + dma_data->dma_size = 2; + break; + default: + return -EINVAL; + } + + writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + pr_debug("hw_params w: IISMOD: %x\n", iismod); + return 0; +} + +static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + + pr_debug("Entered %s\n", __func__); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!s3c24xx_snd_is_clkmaster()) { + ret = s3c24xx_snd_lrsync(); + if (ret) + goto exit_err; + } + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + s3c24xx_snd_rxctrl(1); + else + s3c24xx_snd_txctrl(1); + + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + s3c24xx_snd_rxctrl(0); + else + s3c24xx_snd_txctrl(0); + break; + default: + ret = -EINVAL; + break; + } + +exit_err: + return ret; +} + +/* + * Set S3C24xx Clock source + */ +static int s3c24xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); + + pr_debug("Entered %s\n", __func__); + + iismod &= ~S3C2440_IISMOD_MPLL; + + switch (clk_id) { + case S3C24XX_CLKSRC_PCLK: + break; + case S3C24XX_CLKSRC_MPLL: + iismod |= S3C2440_IISMOD_MPLL; + break; + default: + return -EINVAL; + } + + writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + return 0; +} + +/* + * Set S3C24xx Clock dividers + */ +static int s3c24xx_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + u32 reg; + + pr_debug("Entered %s\n", __func__); + + switch (div_id) { + case S3C24XX_DIV_BCLK: + reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~S3C2410_IISMOD_FS_MASK; + writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); + break; + case S3C24XX_DIV_MCLK: + reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~(S3C2410_IISMOD_384FS); + writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); + break; + case S3C24XX_DIV_PRESCALER: + writel(div, s3c24xx_i2s.regs + S3C2410_IISPSR); + reg = readl(s3c24xx_i2s.regs + S3C2410_IISCON); + writel(reg | S3C2410_IISCON_PSCEN, s3c24xx_i2s.regs + S3C2410_IISCON); + break; + default: + return -EINVAL; + } + + return 0; +} + +/* + * To avoid duplicating clock code, allow machine driver to + * get the clockrate from here. + */ +u32 s3c24xx_i2s_get_clockrate(void) +{ + return clk_get_rate(s3c24xx_i2s.iis_clk); +} +EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); + +static int s3c24xx_i2s_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + pr_debug("Entered %s\n", __func__); + + s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); + if (s3c24xx_i2s.regs == NULL) + return -ENXIO; + + s3c24xx_i2s.iis_clk = clk_get(&pdev->dev, "iis"); + if (s3c24xx_i2s.iis_clk == NULL) { + pr_err("failed to get iis_clock\n"); + iounmap(s3c24xx_i2s.regs); + return -ENODEV; + } + clk_enable(s3c24xx_i2s.iis_clk); + + /* Configure the I2S pins in correct mode */ + s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); + s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK); + s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK); + s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); + s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); + + writel(S3C2410_IISCON_IISEN, s3c24xx_i2s.regs + S3C2410_IISCON); + + s3c24xx_snd_txctrl(0); + s3c24xx_snd_rxctrl(0); + + return 0; +} + +#ifdef CONFIG_PM +static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai) +{ + pr_debug("Entered %s\n", __func__); + + s3c24xx_i2s.iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); + s3c24xx_i2s.iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); + s3c24xx_i2s.iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); + s3c24xx_i2s.iispsr = readl(s3c24xx_i2s.regs + S3C2410_IISPSR); + + clk_disable(s3c24xx_i2s.iis_clk); + + return 0; +} + +static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai) +{ + pr_debug("Entered %s\n", __func__); + clk_enable(s3c24xx_i2s.iis_clk); + + writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); + writel(s3c24xx_i2s.iismod, s3c24xx_i2s.regs + S3C2410_IISMOD); + writel(s3c24xx_i2s.iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); + writel(s3c24xx_i2s.iispsr, s3c24xx_i2s.regs + S3C2410_IISPSR); + + return 0; +} +#else +#define s3c24xx_i2s_suspend NULL +#define s3c24xx_i2s_resume NULL +#endif + + +#define S3C24XX_I2S_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +static struct snd_soc_dai_ops s3c24xx_i2s_dai_ops = { + .trigger = s3c24xx_i2s_trigger, + .hw_params = s3c24xx_i2s_hw_params, + .set_fmt = s3c24xx_i2s_set_fmt, + .set_clkdiv = s3c24xx_i2s_set_clkdiv, + .set_sysclk = s3c24xx_i2s_set_sysclk, +}; + +struct snd_soc_dai s3c24xx_i2s_dai = { + .name = "s3c24xx-i2s", + .id = 0, + .probe = s3c24xx_i2s_probe, + .suspend = s3c24xx_i2s_suspend, + .resume = s3c24xx_i2s_resume, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C24XX_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C24XX_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &s3c24xx_i2s_dai_ops, +}; +EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai); + +static int __init s3c24xx_i2s_init(void) +{ + return snd_soc_register_dai(&s3c24xx_i2s_dai); +} +module_init(s3c24xx_i2s_init); + +static void __exit s3c24xx_i2s_exit(void) +{ + snd_soc_unregister_dai(&s3c24xx_i2s_dai); +} +module_exit(s3c24xx_i2s_exit); + +/* Module information */ +MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("s3c24xx I2S SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c24xx-i2s.h b/sound/soc/samsung/s3c24xx-i2s.h new file mode 100644 index 0000000..726d91c --- /dev/null +++ b/sound/soc/samsung/s3c24xx-i2s.h @@ -0,0 +1,37 @@ +/* + * s3c24xx-i2s.c -- ALSA Soc Audio Layer + * + * Copyright 2005 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * Revision history + * 10th Nov 2006 Initial version. + */ + +#ifndef S3C24XXI2S_H_ +#define S3C24XXI2S_H_ + +/* clock sources */ +#define S3C24XX_CLKSRC_PCLK 0 +#define S3C24XX_CLKSRC_MPLL 1 + +/* Clock dividers */ +#define S3C24XX_DIV_MCLK 0 +#define S3C24XX_DIV_BCLK 1 +#define S3C24XX_DIV_PRESCALER 2 + +/* prescaler */ +#define S3C24XX_PRESCALE(a,b) \ + (((a - 1) << S3C2410_IISPSR_INTSHIFT) | ((b - 1) << S3C2410_IISPSR_EXTSHFIT)) + +u32 s3c24xx_i2s_get_clockrate(void); + +extern struct snd_soc_dai s3c24xx_i2s_dai; + +#endif /*S3C24XXI2S_H_*/ diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c new file mode 100644 index 0000000..4984754 --- /dev/null +++ b/sound/soc/samsung/s3c24xx_simtec.c @@ -0,0 +1,394 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec.c + * + * Copyright 2009 Simtec Electronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <linux/i2c.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +static struct s3c24xx_audio_simtec_pdata *pdata; +static struct clk *xtal_clk; + +static int spk_gain; +static int spk_unmute; + +/** + * speaker_gain_get - read the speaker gain setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be updated. + * + * Read the value for the AMP gain control. + */ +static int speaker_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = spk_gain; + return 0; +} + +/** + * speaker_gain_set - set the value of the speaker amp gain + * @value: The value to write. + */ +static void speaker_gain_set(int value) +{ + gpio_set_value_cansleep(pdata->amp_gain[0], value & 1); + gpio_set_value_cansleep(pdata->amp_gain[1], value >> 1); +} + +/** + * speaker_gain_put - set the speaker gain setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be set. + * + * Set the value of the speaker gain from the specified + * @ucontrol setting. + * + * Note, if the speaker amp is muted, then we do not set a gain value + * as at-least one of the ICs that is fitted will try and power up even + * if the main control is set to off. + */ +static int speaker_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int value = ucontrol->value.integer.value[0]; + + spk_gain = value; + + if (!spk_unmute) + speaker_gain_set(value); + + return 0; +} + +static const struct snd_kcontrol_new amp_gain_controls[] = { + SOC_SINGLE_EXT("Speaker Gain", 0, 0, 3, 0, + speaker_gain_get, speaker_gain_put), +}; + +/** + * spk_unmute_state - set the unmute state of the speaker + * @to: zero to unmute, non-zero to ununmute. + */ +static void spk_unmute_state(int to) +{ + pr_debug("%s: to=%d\n", __func__, to); + + spk_unmute = to; + gpio_set_value(pdata->amp_gpio, to); + + /* if we're umuting, also re-set the gain */ + if (to && pdata->amp_gain[0] > 0) + speaker_gain_set(spk_gain); +} + +/** + * speaker_unmute_get - read the speaker unmute setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be updated. + * + * Read the value for the AMP gain control. + */ +static int speaker_unmute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = spk_unmute; + return 0; +} + +/** + * speaker_unmute_put - set the speaker unmute setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be set. + * + * Set the value of the speaker gain from the specified + * @ucontrol setting. + */ +static int speaker_unmute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + spk_unmute_state(ucontrol->value.integer.value[0]); + return 0; +} + +/* This is added as a manual control as the speaker amps create clicks + * when their power state is changed, which are far more noticeable than + * anything produced by the CODEC itself. + */ +static const struct snd_kcontrol_new amp_unmute_controls[] = { + SOC_SINGLE_EXT("Speaker Switch", 0, 0, 1, 0, + speaker_unmute_get, speaker_unmute_put), +}; + +void simtec_audio_init(struct snd_soc_codec *codec) +{ + if (pdata->amp_gpio > 0) { + pr_debug("%s: adding amp routes\n", __func__); + + snd_soc_add_controls(codec, amp_unmute_controls, + ARRAY_SIZE(amp_unmute_controls)); + } + + if (pdata->amp_gain[0] > 0) { + pr_debug("%s: adding amp controls\n", __func__); + snd_soc_add_controls(codec, amp_gain_controls, + ARRAY_SIZE(amp_gain_controls)); + } +} +EXPORT_SYMBOL_GPL(simtec_audio_init); + +#define CODEC_CLOCK 12000000 + +/** + * simtec_hw_params - update hardware parameters + * @substream: The audio substream instance. + * @params: The parameters requested. + * + * Update the codec data routing and configuration settings + * from the supplied data. + */ +static int simtec_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set the CODEC as the bus clock master, I2S */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set cpu dai format\n", __func__); + return ret; + } + + /* Set the CODEC as the bus clock master */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set codec dai format\n", __func__); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + CODEC_CLOCK, SND_SOC_CLOCK_IN); + if (ret) { + pr_err( "%s: failed setting codec sysclk\n", __func__); + return ret; + } + + if (pdata->use_mpllin) { + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_MPLL, + 0, SND_SOC_CLOCK_OUT); + + if (ret) { + pr_err("%s: failed to set MPLLin as clksrc\n", + __func__); + return ret; + } + } + + if (pdata->output_cdclk) { + int cdclk_scale; + + cdclk_scale = clk_get_rate(xtal_clk) / CODEC_CLOCK; + cdclk_scale--; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + cdclk_scale); + } + + return 0; +} + +static int simtec_call_startup(struct s3c24xx_audio_simtec_pdata *pd) +{ + /* call any board supplied startup code, this currently only + * covers the bast/vr1000 which have a CPLD in the way of the + * LRCLK */ + if (pd->startup) + pd->startup(); + + return 0; +} + +static struct snd_soc_ops simtec_snd_ops = { + .hw_params = simtec_hw_params, +}; + +/** + * attach_gpio_amp - get and configure the necessary gpios + * @dev: The device we're probing. + * @pd: The platform data supplied by the board. + * + * If there is a GPIO based amplifier attached to the board, claim + * the necessary GPIO lines for it, and set default values. + */ +static int attach_gpio_amp(struct device *dev, + struct s3c24xx_audio_simtec_pdata *pd) +{ + int ret; + + /* attach gpio amp gain (if any) */ + if (pdata->amp_gain[0] > 0) { + ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain0"); + if (ret) { + dev_err(dev, "cannot get amp gpio gain0\n"); + return ret; + } + + ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain1"); + if (ret) { + dev_err(dev, "cannot get amp gpio gain1\n"); + gpio_free(pdata->amp_gain[0]); + return ret; + } + + gpio_direction_output(pd->amp_gain[0], 0); + gpio_direction_output(pd->amp_gain[1], 0); + } + + /* note, currently we assume GPA0 isn't valid amp */ + if (pdata->amp_gpio > 0) { + ret = gpio_request(pd->amp_gpio, "gpio-amp"); + if (ret) { + dev_err(dev, "cannot get amp gpio %d (%d)\n", + pd->amp_gpio, ret); + goto err_amp; + } + + /* set the amp off at startup */ + spk_unmute_state(0); + } + + return 0; + +err_amp: + if (pd->amp_gain[0] > 0) { + gpio_free(pd->amp_gain[0]); + gpio_free(pd->amp_gain[1]); + } + + return ret; +} + +static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd) +{ + if (pd->amp_gain[0] > 0) { + gpio_free(pd->amp_gain[0]); + gpio_free(pd->amp_gain[1]); + } + + if (pd->amp_gpio > 0) + gpio_free(pd->amp_gpio); +} + +#ifdef CONFIG_PM +int simtec_audio_resume(struct device *dev) +{ + simtec_call_startup(pdata); + return 0; +} + +const struct dev_pm_ops simtec_audio_pmops = { + .resume = simtec_audio_resume, +}; +EXPORT_SYMBOL_GPL(simtec_audio_pmops); +#endif + +int __devinit simtec_audio_core_probe(struct platform_device *pdev, + struct snd_soc_device *socdev) +{ + struct platform_device *snd_dev; + int ret; + + socdev->card->dai_link->ops = &simtec_snd_ops; + + pdata = pdev->dev.platform_data; + if (!pdata) { + dev_err(&pdev->dev, "no platform data supplied\n"); + return -EINVAL; + } + + simtec_call_startup(pdata); + + xtal_clk = clk_get(&pdev->dev, "xtal"); + if (IS_ERR(xtal_clk)) { + dev_err(&pdev->dev, "could not get clkout0\n"); + return -EINVAL; + } + + dev_info(&pdev->dev, "xtal rate is %ld\n", clk_get_rate(xtal_clk)); + + ret = attach_gpio_amp(&pdev->dev, pdata); + if (ret) + goto err_clk; + + snd_dev = platform_device_alloc("soc-audio", -1); + if (!snd_dev) { + dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n"); + ret = -ENOMEM; + goto err_gpio; + } + + platform_set_drvdata(snd_dev, socdev); + socdev->dev = &snd_dev->dev; + + ret = platform_device_add(snd_dev); + if (ret) { + dev_err(&pdev->dev, "failed to add soc-audio dev\n"); + goto err_pdev; + } + + platform_set_drvdata(pdev, snd_dev); + return 0; + +err_pdev: + platform_device_put(snd_dev); + +err_gpio: + detach_gpio_amp(pdata); + +err_clk: + clk_put(xtal_clk); + return ret; +} +EXPORT_SYMBOL_GPL(simtec_audio_core_probe); + +int __devexit simtec_audio_remove(struct platform_device *pdev) +{ + struct platform_device *snd_dev = platform_get_drvdata(pdev); + + platform_device_unregister(snd_dev); + + detach_gpio_amp(pdata); + clk_put(xtal_clk); + return 0; +} +EXPORT_SYMBOL_GPL(simtec_audio_remove); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio common support"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c24xx_simtec.h b/sound/soc/samsung/s3c24xx_simtec.h new file mode 100644 index 0000000..e18faee --- /dev/null +++ b/sound/soc/samsung/s3c24xx_simtec.h @@ -0,0 +1,22 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec.h + * + * Copyright 2009 Simtec Electronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +extern void simtec_audio_init(struct snd_soc_codec *codec); + +extern int simtec_audio_core_probe(struct platform_device *pdev, + struct snd_soc_device *socdev); + +extern int simtec_audio_remove(struct platform_device *pdev); + +#ifdef CONFIG_PM +extern const struct dev_pm_ops simtec_audio_pmops; +#define simtec_audio_pm &simtec_audio_pmops +#else +#define simtec_audio_pm NULL +#endif diff --git a/sound/soc/samsung/s3c24xx_simtec_hermes.c b/sound/soc/samsung/s3c24xx_simtec_hermes.c new file mode 100644 index 0000000..bdf8951 --- /dev/null +++ b/sound/soc/samsung/s3c24xx_simtec_hermes.c @@ -0,0 +1,153 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec_hermes.c + * + * Copyright 2009 Simtec Electronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +#include "../codecs/tlv320aic3x.h" + +static const struct snd_soc_dapm_widget dapm_widgets[] = { + SND_SOC_DAPM_LINE("GSM Out", NULL), + SND_SOC_DAPM_LINE("GSM In", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_LINE("ZV", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_HP("Headphone Jack", NULL), +}; + +static const struct snd_soc_dapm_route base_map[] = { + /* Headphone connected to HP{L,R}OUT and HP{L,R}COM */ + + { "Headphone Jack", NULL, "HPLOUT" }, + { "Headphone Jack", NULL, "HPLCOM" }, + { "Headphone Jack", NULL, "HPROUT" }, + { "Headphone Jack", NULL, "HPRCOM" }, + + /* ZV connected to Line1 */ + + { "LINE1L", NULL, "ZV" }, + { "LINE1R", NULL, "ZV" }, + + /* Line In connected to Line2 */ + + { "LINE2L", NULL, "Line In" }, + { "LINE2R", NULL, "Line In" }, + + /* Microphone connected to MIC3R and MIC_BIAS */ + + { "MIC3L", NULL, "Mic Jack" }, + + /* GSM connected to MONO_LOUT and MIC3L (in) */ + + { "GSM Out", NULL, "MONO_LOUT" }, + { "MIC3L", NULL, "GSM In" }, + + /* Speaker is connected to LINEOUT{LN,LP,RN,RP}, however we are + * not using the DAPM to power it up and down as there it makes + * a click when powering up. */ +}; + +/** + * simtec_hermes_init - initialise and add controls + * @codec; The codec instance to attach to. + * + * Attach our controls and configure the necessary codec + * mappings for our sound card instance. +*/ +static int simtec_hermes_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, dapm_widgets, + ARRAY_SIZE(dapm_widgets)); + + snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + simtec_audio_init(codec); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct aic3x_setup_data codec_setup = { +}; + +static struct snd_soc_dai_link simtec_dai_aic33 = { + .name = "tlv320aic33", + .stream_name = "TLV320AIC33", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &aic3x_dai, + .init = simtec_hermes_init, +}; + +/* simtec audio machine driver */ +static struct snd_soc_card snd_soc_machine_simtec_aic33 = { + .name = "Simtec-Hermes", + .platform = &s3c24xx_soc_platform, + .dai_link = &simtec_dai_aic33, + .num_links = 1, +}; + +/* simtec audio subsystem */ +static struct snd_soc_device simtec_snd_devdata_aic33 = { + .card = &snd_soc_machine_simtec_aic33, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &codec_setup, +}; + +static int __devinit simtec_audio_hermes_probe(struct platform_device *pd) +{ + dev_info(&pd->dev, "probing....\n"); + return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic33); +} + +static struct platform_driver simtec_audio_hermes_platdrv = { + .driver = { + .owner = THIS_MODULE, + .name = "s3c24xx-simtec-hermes-snd", + .pm = simtec_audio_pm, + }, + .probe = simtec_audio_hermes_probe, + .remove = __devexit_p(simtec_audio_remove), +}; + +MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd"); + +static int __init simtec_hermes_modinit(void) +{ + return platform_driver_register(&simtec_audio_hermes_platdrv); +} + +static void __exit simtec_hermes_modexit(void) +{ + platform_driver_unregister(&simtec_audio_hermes_platdrv); +} + +module_init(simtec_hermes_modinit); +module_exit(simtec_hermes_modexit); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c new file mode 100644 index 0000000..185c0ac --- /dev/null +++ b/sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c @@ -0,0 +1,137 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c + * + * Copyright 2009 Simtec Electronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +#include "../codecs/tlv320aic23.h" + +/* supported machines: + * + * Machine Connections AMP + * ------- ----------- --- + * BAST MIC, HPOUT, LOUT, LIN TPA2001D1 (HPOUTL,R) (gain hardwired) + * VR1000 HPOUT, LIN None + * VR2000 LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) + * DePicture LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) + * Anubis LIN, LOUT, MIC, HP TPA2001D1 (HPOUTL,R) + */ + +static const struct snd_soc_dapm_widget dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route base_map[] = { + { "Headphone Jack", NULL, "LHPOUT"}, + { "Headphone Jack", NULL, "RHPOUT"}, + + { "Line Out", NULL, "LOUT" }, + { "Line Out", NULL, "ROUT" }, + + { "LLINEIN", NULL, "Line In"}, + { "RLINEIN", NULL, "Line In"}, + + { "MICIN", NULL, "Mic Jack"}, +}; + +/** + * simtec_tlv320aic23_init - initialise and add controls + * @codec; The codec instance to attach to. + * + * Attach our controls and configure the necessary codec + * mappings for our sound card instance. +*/ +static int simtec_tlv320aic23_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, dapm_widgets, + ARRAY_SIZE(dapm_widgets)); + + snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + simtec_audio_init(codec); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link simtec_dai_aic23 = { + .name = "tlv320aic23", + .stream_name = "TLV320AIC23", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &tlv320aic23_dai, + .init = simtec_tlv320aic23_init, +}; + +/* simtec audio machine driver */ +static struct snd_soc_card snd_soc_machine_simtec_aic23 = { + .name = "Simtec", + .platform = &s3c24xx_soc_platform, + .dai_link = &simtec_dai_aic23, + .num_links = 1, +}; + +/* simtec audio subsystem */ +static struct snd_soc_device simtec_snd_devdata_aic23 = { + .card = &snd_soc_machine_simtec_aic23, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static int __devinit simtec_audio_tlv320aic23_probe(struct platform_device *pd) +{ + return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic23); +} + +static struct platform_driver simtec_audio_tlv320aic23_platdrv = { + .driver = { + .owner = THIS_MODULE, + .name = "s3c24xx-simtec-tlv320aic23", + .pm = simtec_audio_pm, + }, + .probe = simtec_audio_tlv320aic23_probe, + .remove = __devexit_p(simtec_audio_remove), +}; + +MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23"); + +static int __init simtec_tlv320aic23_modinit(void) +{ + return platform_driver_register(&simtec_audio_tlv320aic23_platdrv); +} + +static void __exit simtec_tlv320aic23_modexit(void) +{ + platform_driver_unregister(&simtec_audio_tlv320aic23_platdrv); +} + +module_init(simtec_tlv320aic23_modinit); +module_exit(simtec_tlv320aic23_modexit); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c new file mode 100644 index 0000000..052d596 --- /dev/null +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -0,0 +1,373 @@ +/* + * Modifications by Christian Pellegrin <chripell(a)evolware.org> + * + * s3c24xx_uda134x.c -- S3C24XX_UDA134X ALSA SoC Audio board driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/mutex.h> +#include <linux/gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/s3c24xx_uda134x.h> +#include <sound/uda134x.h> + +#include <plat/regs-iis.h> + +#include "s3c-dma.h" +#include "s3c24xx-i2s.h" +#include "../codecs/uda134x.h" + + +/* #define ENFORCE_RATES 1 */ +/* + Unfortunately the S3C24XX in master mode has a limited capacity of + generating the clock for the codec. If you define this only rates + that are really available will be enforced. But be careful, most + user level application just want the usual sampling frequencies (8, + 11.025, 22.050, 44.1 kHz) and anyway resampling is a costly + operation for embedded systems. So if you aren't very lucky or your + hardware engineer wasn't very forward-looking it's better to leave + this undefined. If you do so an approximate value for the requested + sampling rate in the range -/+ 5% will be chosen. If this in not + possible an error will be returned. +*/ + +static struct clk *xtal; +static struct clk *pclk; +/* this is need because we don't have a place where to keep the + * pointers to the clocks in each substream. We get the clocks only + * when we are actually using them so we don't block stuff like + * frequency change or oscillator power-off */ +static int clk_users; +static DEFINE_MUTEX(clk_lock); + +static unsigned int rates[33 * 2]; +#ifdef ENFORCE_RATES +static struct snd_pcm_hw_constraint_list hw_constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; +#endif + +static struct platform_device *s3c24xx_uda134x_snd_device; + +static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) +{ + int ret = 0; +#ifdef ENFORCE_RATES + struct snd_pcm_runtime *runtime = substream->runtime; +#endif + + mutex_lock(&clk_lock); + pr_debug("%s %d\n", __func__, clk_users); + if (clk_users == 0) { + xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal"); + if (!xtal) { + printk(KERN_ERR "%s cannot get xtal\n", __func__); + ret = -EBUSY; + } else { + pclk = clk_get(&s3c24xx_uda134x_snd_device->dev, + "pclk"); + if (!pclk) { + printk(KERN_ERR "%s cannot get pclk\n", + __func__); + clk_put(xtal); + ret = -EBUSY; + } + } + if (!ret) { + int i, j; + + for (i = 0; i < 2; i++) { + int fs = i ? 256 : 384; + + rates[i*33] = clk_get_rate(xtal) / fs; + for (j = 1; j < 33; j++) + rates[i*33 + j] = clk_get_rate(pclk) / + (j * fs); + } + } + } + clk_users += 1; + mutex_unlock(&clk_lock); + if (!ret) { +#ifdef ENFORCE_RATES + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &hw_constraints_rates); + if (ret < 0) + printk(KERN_ERR "%s cannot set constraints\n", + __func__); +#endif + } + return ret; +} + +static void s3c24xx_uda134x_shutdown(struct snd_pcm_substream *substream) +{ + mutex_lock(&clk_lock); + pr_debug("%s %d\n", __func__, clk_users); + clk_users -= 1; + if (clk_users == 0) { + clk_put(xtal); + xtal = NULL; + clk_put(pclk); + pclk = NULL; + } + mutex_unlock(&clk_lock); +} + +static int s3c24xx_uda134x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int clk = 0; + int ret = 0; + int clk_source, fs_mode; + unsigned long rate = params_rate(params); + long err, cerr; + unsigned int div; + int i, bi; + + err = 999999; + bi = 0; + for (i = 0; i < 2*33; i++) { + cerr = rates[i] - rate; + if (cerr < 0) + cerr = -cerr; + if (cerr < err) { + err = cerr; + bi = i; + } + } + if (bi / 33 == 1) + fs_mode = S3C2410_IISMOD_256FS; + else + fs_mode = S3C2410_IISMOD_384FS; + if (bi % 33 == 0) { + clk_source = S3C24XX_CLKSRC_MPLL; + div = 1; + } else { + clk_source = S3C24XX_CLKSRC_PCLK; + div = bi % 33; + } + pr_debug("%s desired rate %lu, %d\n", __func__, rate, bi); + + clk = (fs_mode == S3C2410_IISMOD_384FS ? 384 : 256) * rate; + pr_debug("%s will use: %s %s %d sysclk %d err %ld\n", __func__, + fs_mode == S3C2410_IISMOD_384FS ? "384FS" : "256FS", + clk_source == S3C24XX_CLKSRC_MPLL ? "MPLLin" : "PCLK", + div, clk, err); + + if ((err * 100 / rate) > 5) { + printk(KERN_ERR "S3C24XX_UDA134X: effective frequency " + "too different from desired (%ld%%)\n", + err * 100 / rate); + return -EINVAL; + } + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, clk_source , clk, + SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, fs_mode); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, + S3C2410_IISMOD_32FS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + S3C24XX_PRESCALE(div, div)); + if (ret < 0) + return ret; + + /* set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops s3c24xx_uda134x_ops = { + .startup = s3c24xx_uda134x_startup, + .shutdown = s3c24xx_uda134x_shutdown, + .hw_params = s3c24xx_uda134x_hw_params, +}; + +static struct snd_soc_dai_link s3c24xx_uda134x_dai_link = { + .name = "UDA134X", + .stream_name = "UDA134X", + .codec_dai = &uda134x_dai, + .cpu_dai = &s3c24xx_i2s_dai, + .ops = &s3c24xx_uda134x_ops, +}; + +static struct snd_soc_card snd_soc_s3c24xx_uda134x = { + .name = "S3C24XX_UDA134X", + .platform = &s3c24xx_soc_platform, + .dai_link = &s3c24xx_uda134x_dai_link, + .num_links = 1, +}; + +static struct s3c24xx_uda134x_platform_data *s3c24xx_uda134x_l3_pins; + +static void setdat(int v) +{ + gpio_set_value(s3c24xx_uda134x_l3_pins->l3_data, v > 0); +} + +static void setclk(int v) +{ + gpio_set_value(s3c24xx_uda134x_l3_pins->l3_clk, v > 0); +} + +static void setmode(int v) +{ + gpio_set_value(s3c24xx_uda134x_l3_pins->l3_mode, v > 0); +} + +static struct uda134x_platform_data s3c24xx_uda134x = { + .l3 = { + .setdat = setdat, + .setclk = setclk, + .setmode = setmode, + .data_hold = 1, + .data_setup = 1, + .clock_high = 1, + .mode_hold = 1, + .mode = 1, + .mode_setup = 1, + }, +}; + +static struct snd_soc_device s3c24xx_uda134x_snd_devdata = { + .card = &snd_soc_s3c24xx_uda134x, + .codec_dev = &soc_codec_dev_uda134x, + .codec_data = &s3c24xx_uda134x, +}; + +static int s3c24xx_uda134x_setup_pin(int pin, char *fun) +{ + if (gpio_request(pin, "s3c24xx_uda134x") < 0) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " + "l3 %s pin already in use", fun); + return -EBUSY; + } + gpio_direction_output(pin, 0); + return 0; +} + +static int s3c24xx_uda134x_probe(struct platform_device *pdev) +{ + int ret; + + printk(KERN_INFO "S3C24XX_UDA134X SoC Audio driver\n"); + + s3c24xx_uda134x_l3_pins = pdev->dev.platform_data; + if (s3c24xx_uda134x_l3_pins == NULL) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " + "unable to find platform data\n"); + return -ENODEV; + } + s3c24xx_uda134x.power = s3c24xx_uda134x_l3_pins->power; + s3c24xx_uda134x.model = s3c24xx_uda134x_l3_pins->model; + + if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_data, + "data") < 0) + return -EBUSY; + if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_clk, + "clk") < 0) { + gpio_free(s3c24xx_uda134x_l3_pins->l3_data); + return -EBUSY; + } + if (s3c24xx_uda134x_setup_pin(s3c24xx_uda134x_l3_pins->l3_mode, + "mode") < 0) { + gpio_free(s3c24xx_uda134x_l3_pins->l3_data); + gpio_free(s3c24xx_uda134x_l3_pins->l3_clk); + return -EBUSY; + } + + s3c24xx_uda134x_snd_device = platform_device_alloc("soc-audio", -1); + if (!s3c24xx_uda134x_snd_device) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: " + "Unable to register\n"); + return -ENOMEM; + } + + platform_set_drvdata(s3c24xx_uda134x_snd_device, + &s3c24xx_uda134x_snd_devdata); + s3c24xx_uda134x_snd_devdata.dev = &s3c24xx_uda134x_snd_device->dev; + ret = platform_device_add(s3c24xx_uda134x_snd_device); + if (ret) { + printk(KERN_ERR "S3C24XX_UDA134X SoC Audio: Unable to add\n"); + platform_device_put(s3c24xx_uda134x_snd_device); + } + + return ret; +} + +static int s3c24xx_uda134x_remove(struct platform_device *pdev) +{ + platform_device_unregister(s3c24xx_uda134x_snd_device); + gpio_free(s3c24xx_uda134x_l3_pins->l3_data); + gpio_free(s3c24xx_uda134x_l3_pins->l3_clk); + gpio_free(s3c24xx_uda134x_l3_pins->l3_mode); + return 0; +} + +static struct platform_driver s3c24xx_uda134x_driver = { + .probe = s3c24xx_uda134x_probe, + .remove = s3c24xx_uda134x_remove, + .driver = { + .name = "s3c24xx_uda134x", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c24xx_uda134x_init(void) +{ + return platform_driver_register(&s3c24xx_uda134x_driver); +} + +static void __exit s3c24xx_uda134x_exit(void) +{ + platform_driver_unregister(&s3c24xx_uda134x_driver); +} + + +module_init(s3c24xx_uda134x_init); +module_exit(s3c24xx_uda134x_exit); + +MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin <chripell(a)evolware.org>"); +MODULE_DESCRIPTION("S3C24XX_UDA134X ALSA SoC audio driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c64xx-i2s-v4.c b/sound/soc/samsung/s3c64xx-i2s-v4.c new file mode 100644 index 0000000..06db130 --- /dev/null +++ b/sound/soc/samsung/s3c64xx-i2s-v4.c @@ -0,0 +1,209 @@ +/* sound/soc/s3c24xx/s3c64xx-i2s-v4.c + * + * ALSA SoC Audio Layer - S3C64XX I2Sv4 driver + * Copyright (c) 2010 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/clk.h> +#include <linux/gpio.h> +#include <linux/io.h> + +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#include <mach/gpio-bank-c.h> +#include <mach/gpio-bank-h.h> +#include <plat/gpio-cfg.h> + +#include <mach/map.h> +#include <mach/dma.h> + +#include "s3c-dma.h" +#include "regs-i2s-v2.h" +#include "s3c64xx-i2s.h" + +static struct s3c2410_dma_client s3c64xx_dma_client_out = { + .name = "I2Sv4 PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c64xx_dma_client_in = { + .name = "I2Sv4 PCM Stereo in" +}; + +static struct s3c_dma_params s3c64xx_i2sv4_pcm_stereo_out; +static struct s3c_dma_params s3c64xx_i2sv4_pcm_stereo_in; +static struct s3c_i2sv2_info s3c64xx_i2sv4; + +struct snd_soc_dai s3c64xx_i2s_v4_dai; +EXPORT_SYMBOL_GPL(s3c64xx_i2s_v4_dai); + +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static int s3c64xx_i2sv4_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + /* configure GPIO for i2s port */ + s3c_gpio_cfgpin(S3C64XX_GPC(4), S3C64XX_GPC4_I2S_V40_DO0); + s3c_gpio_cfgpin(S3C64XX_GPC(5), S3C64XX_GPC5_I2S_V40_DO1); + s3c_gpio_cfgpin(S3C64XX_GPC(7), S3C64XX_GPC7_I2S_V40_DO2); + s3c_gpio_cfgpin(S3C64XX_GPH(6), S3C64XX_GPH6_I2S_V40_BCLK); + s3c_gpio_cfgpin(S3C64XX_GPH(7), S3C64XX_GPH7_I2S_V40_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPH(8), S3C64XX_GPH8_I2S_V40_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPH(9), S3C64XX_GPH9_I2S_V40_DI); + + return 0; +} + +static int s3c_i2sv4_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct s3c_i2sv2_info *i2s = to_info(cpu_dai); + struct s3c_dma_params *dma_data; + u32 iismod; + + dev_dbg(cpu_dai->dev, "Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = i2s->dma_playback; + else + dma_data = i2s->dma_capture; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + + iismod = readl(i2s->regs + S3C2412_IISMOD); + dev_dbg(cpu_dai->dev, "%s: r: IISMOD: %x\n", __func__, iismod); + + iismod &= ~S3C64XX_IISMOD_BLC_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + iismod |= S3C64XX_IISMOD_BLC_8BIT; + break; + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S24_LE: + iismod |= S3C64XX_IISMOD_BLC_24BIT; + break; + } + + writel(iismod, i2s->regs + S3C2412_IISMOD); + dev_dbg(cpu_dai->dev, "%s: w: IISMOD: %x\n", __func__, iismod); + + return 0; +} + +static struct snd_soc_dai_ops s3c64xx_i2sv4_dai_ops = { + .hw_params = s3c_i2sv4_hw_params, +}; + +static __devinit int s3c64xx_i2sv4_dev_probe(struct platform_device *pdev) +{ + struct s3c_i2sv2_info *i2s; + struct snd_soc_dai *dai; + int ret; + + i2s = &s3c64xx_i2sv4; + dai = &s3c64xx_i2s_v4_dai; + + if (dai->dev) { + dev_dbg(dai->dev, "%s: \ + I2Sv4 instance already registered!\n", __func__); + return -EBUSY; + } + + dai->dev = &pdev->dev; + dai->name = "s3c64xx-i2s-v4"; + dai->id = 0; + dai->symmetric_rates = 1; + dai->playback.channels_min = 2; + dai->playback.channels_max = 2; + dai->playback.rates = S3C64XX_I2S_RATES; + dai->playback.formats = S3C64XX_I2S_FMTS; + dai->capture.channels_min = 2; + dai->capture.channels_max = 2; + dai->capture.rates = S3C64XX_I2S_RATES; + dai->capture.formats = S3C64XX_I2S_FMTS; + dai->probe = s3c64xx_i2sv4_probe; + dai->ops = &s3c64xx_i2sv4_dai_ops; + + i2s->feature |= S3C_FEATURE_CDCLKCON; + + i2s->dma_capture = &s3c64xx_i2sv4_pcm_stereo_in; + i2s->dma_playback = &s3c64xx_i2sv4_pcm_stereo_out; + + i2s->dma_capture->channel = DMACH_HSI_I2SV40_RX; + i2s->dma_capture->dma_addr = S3C64XX_PA_IISV4 + S3C2412_IISRXD; + i2s->dma_playback->channel = DMACH_HSI_I2SV40_TX; + i2s->dma_playback->dma_addr = S3C64XX_PA_IISV4 + S3C2412_IISTXD; + + i2s->dma_capture->client = &s3c64xx_dma_client_in; + i2s->dma_capture->dma_size = 4; + i2s->dma_playback->client = &s3c64xx_dma_client_out; + i2s->dma_playback->dma_size = 4; + + i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus"); + if (IS_ERR(i2s->iis_cclk)) { + dev_err(&pdev->dev, "failed to get audio-bus\n"); + ret = PTR_ERR(i2s->iis_cclk); + goto err; + } + + clk_enable(i2s->iis_cclk); + + ret = s3c_i2sv2_probe(pdev, dai, i2s, 0); + if (ret) + goto err_clk; + + ret = s3c_i2sv2_register_dai(dai); + if (ret != 0) + goto err_i2sv2; + + return 0; + +err_i2sv2: + /* Not implemented for I2Sv2 core yet */ +err_clk: + clk_put(i2s->iis_cclk); +err: + return ret; +} + +static __devexit int s3c64xx_i2sv4_dev_remove(struct platform_device *pdev) +{ + dev_err(&pdev->dev, "Device removal not yet supported\n"); + return 0; +} + +static struct platform_driver s3c64xx_i2sv4_driver = { + .probe = s3c64xx_i2sv4_dev_probe, + .remove = s3c64xx_i2sv4_dev_remove, + .driver = { + .name = "s3c64xx-iis-v4", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c64xx_i2sv4_init(void) +{ + return platform_driver_register(&s3c64xx_i2sv4_driver); +} +module_init(s3c64xx_i2sv4_init); + +static void __exit s3c64xx_i2sv4_exit(void) +{ + platform_driver_unregister(&s3c64xx_i2sv4_driver); +} +module_exit(s3c64xx_i2sv4_exit); + +/* Module information */ +MODULE_AUTHOR("Jaswinder Singh, <jassi.brar(a)samsung.com>"); +MODULE_DESCRIPTION("S3C64XX I2Sv4 SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c64xx-i2s.c b/sound/soc/samsung/s3c64xx-i2s.c new file mode 100644 index 0000000..1d85cb8 --- /dev/null +++ b/sound/soc/samsung/s3c64xx-i2s.c @@ -0,0 +1,190 @@ +/* sound/soc/s3c24xx/s3c64xx-i2s.c + * + * ALSA SoC Audio Layer - S3C64XX I2S driver + * + * Copyright 2008 Openmoko, Inc. + * Copyright 2008 Simtec Electronics + * Ben Dooks <ben(a)simtec.co.uk> + * http://armlinux.simtec.co.uk/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/clk.h> +#include <linux/gpio.h> +#include <linux/io.h> + +#include <sound/soc.h> + +#include <mach/gpio-bank-d.h> +#include <mach/gpio-bank-e.h> +#include <plat/gpio-cfg.h> + +#include <mach/map.h> +#include <mach/dma.h> + +#include "s3c-dma.h" +#include "regs-i2s-v2.h" +#include "s3c64xx-i2s.h" + +/* The value should be set to maximum of the total number + * of I2Sv3 controllers that any supported SoC has. + */ +#define MAX_I2SV3 2 + +static struct s3c2410_dma_client s3c64xx_dma_client_out = { + .name = "I2S PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c64xx_dma_client_in = { + .name = "I2S PCM Stereo in" +}; + +static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_out[MAX_I2SV3]; +static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_in[MAX_I2SV3]; +static struct s3c_i2sv2_info s3c64xx_i2s[MAX_I2SV3]; + +struct snd_soc_dai s3c64xx_i2s_dai[MAX_I2SV3]; +EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai); + +static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static int s3c64xx_i2s_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + /* configure GPIO for i2s port */ + switch (dai->id) { + case 0: + s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_I2S0_CLK); + s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_I2S0_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPD(2), S3C64XX_GPD2_I2S0_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPD(3), S3C64XX_GPD3_I2S0_DI); + s3c_gpio_cfgpin(S3C64XX_GPD(4), S3C64XX_GPD4_I2S0_D0); + break; + case 1: + s3c_gpio_cfgpin(S3C64XX_GPE(0), S3C64XX_GPE0_I2S1_CLK); + s3c_gpio_cfgpin(S3C64XX_GPE(1), S3C64XX_GPE1_I2S1_CDCLK); + s3c_gpio_cfgpin(S3C64XX_GPE(2), S3C64XX_GPE2_I2S1_LRCLK); + s3c_gpio_cfgpin(S3C64XX_GPE(3), S3C64XX_GPE3_I2S1_DI); + s3c_gpio_cfgpin(S3C64XX_GPE(4), S3C64XX_GPE4_I2S1_D0); + } + + return 0; +} + + +static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops; + +static __devinit int s3c64xx_iis_dev_probe(struct platform_device *pdev) +{ + struct s3c_i2sv2_info *i2s; + struct snd_soc_dai *dai; + int ret; + + if (pdev->id >= MAX_I2SV3) { + dev_err(&pdev->dev, "id %d out of range\n", pdev->id); + return -EINVAL; + } + + i2s = &s3c64xx_i2s[pdev->id]; + dai = &s3c64xx_i2s_dai[pdev->id]; + dai->dev = &pdev->dev; + dai->name = "s3c64xx-i2s"; + dai->id = pdev->id; + dai->symmetric_rates = 1; + dai->playback.channels_min = 2; + dai->playback.channels_max = 2; + dai->playback.rates = S3C64XX_I2S_RATES; + dai->playback.formats = S3C64XX_I2S_FMTS; + dai->capture.channels_min = 2; + dai->capture.channels_max = 2; + dai->capture.rates = S3C64XX_I2S_RATES; + dai->capture.formats = S3C64XX_I2S_FMTS; + dai->probe = s3c64xx_i2s_probe; + dai->ops = &s3c64xx_i2s_dai_ops; + + i2s->feature |= S3C_FEATURE_CDCLKCON; + + i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id]; + i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id]; + + if (pdev->id == 0) { + i2s->dma_capture->channel = DMACH_I2S0_IN; + i2s->dma_capture->dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISRXD; + i2s->dma_playback->channel = DMACH_I2S0_OUT; + i2s->dma_playback->dma_addr = S3C64XX_PA_IIS0 + S3C2412_IISTXD; + } else { + i2s->dma_capture->channel = DMACH_I2S1_IN; + i2s->dma_capture->dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISRXD; + i2s->dma_playback->channel = DMACH_I2S1_OUT; + i2s->dma_playback->dma_addr = S3C64XX_PA_IIS1 + S3C2412_IISTXD; + } + + i2s->dma_capture->client = &s3c64xx_dma_client_in; + i2s->dma_capture->dma_size = 4; + i2s->dma_playback->client = &s3c64xx_dma_client_out; + i2s->dma_playback->dma_size = 4; + + i2s->iis_cclk = clk_get(&pdev->dev, "audio-bus"); + if (IS_ERR(i2s->iis_cclk)) { + dev_err(&pdev->dev, "failed to get audio-bus\n"); + ret = PTR_ERR(i2s->iis_cclk); + goto err; + } + + clk_enable(i2s->iis_cclk); + + ret = s3c_i2sv2_probe(pdev, dai, i2s, 0); + if (ret) + goto err_clk; + + ret = s3c_i2sv2_register_dai(dai); + if (ret != 0) + goto err_i2sv2; + + return 0; + +err_i2sv2: + /* Not implemented for I2Sv2 core yet */ +err_clk: + clk_put(i2s->iis_cclk); +err: + return ret; +} + +static __devexit int s3c64xx_iis_dev_remove(struct platform_device *pdev) +{ + dev_err(&pdev->dev, "Device removal not yet supported\n"); + return 0; +} + +static struct platform_driver s3c64xx_iis_driver = { + .probe = s3c64xx_iis_dev_probe, + .remove = s3c64xx_iis_dev_remove, + .driver = { + .name = "s3c64xx-iis", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c64xx_i2s_init(void) +{ + return platform_driver_register(&s3c64xx_iis_driver); +} +module_init(s3c64xx_i2s_init); + +static void __exit s3c64xx_i2s_exit(void) +{ + platform_driver_unregister(&s3c64xx_iis_driver); +} +module_exit(s3c64xx_i2s_exit); + +/* Module information */ +MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("S3C64XX I2S SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/s3c64xx-i2s.h b/sound/soc/samsung/s3c64xx-i2s.h new file mode 100644 index 0000000..7a40f43 --- /dev/null +++ b/sound/soc/samsung/s3c64xx-i2s.h @@ -0,0 +1,42 @@ +/* sound/soc/s3c24xx/s3c64xx-i2s.h + * + * ALSA SoC Audio Layer - S3C64XX I2S driver + * + * Copyright 2008 Openmoko, Inc. + * Copyright 2008 Simtec Electronics + * Ben Dooks <ben(a)simtec.co.uk> + * http://armlinux.simtec.co.uk/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SND_SOC_S3C24XX_S3C64XX_I2S_H +#define __SND_SOC_S3C24XX_S3C64XX_I2S_H __FILE__ + +struct clk; + +#include "s3c-i2s-v2.h" + +#define S3C64XX_DIV_BCLK S3C_I2SV2_DIV_BCLK +#define S3C64XX_DIV_RCLK S3C_I2SV2_DIV_RCLK +#define S3C64XX_DIV_PRESCALER S3C_I2SV2_DIV_PRESCALER + +#define S3C64XX_CLKSRC_PCLK S3C_I2SV2_CLKSRC_PCLK +#define S3C64XX_CLKSRC_MUX S3C_I2SV2_CLKSRC_AUDIOBUS +#define S3C64XX_CLKSRC_CDCLK S3C_I2SV2_CLKSRC_CDCLK + +#define S3C64XX_I2S_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +#define S3C64XX_I2S_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +extern struct snd_soc_dai s3c64xx_i2s_dai[]; +extern struct snd_soc_dai s3c64xx_i2s_v4_dai; + +#endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */ diff --git a/sound/soc/samsung/smdk2443_wm9710.c b/sound/soc/samsung/smdk2443_wm9710.c new file mode 100644 index 0000000..3622588 --- /dev/null +++ b/sound/soc/samsung/smdk2443_wm9710.c @@ -0,0 +1,81 @@ +/* + * smdk2443_wm9710.c -- SoC audio for smdk2443 + * + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/ac97.h" +#include "s3c-dma.h" +#include "s3c-ac97.h" + +static struct snd_soc_card smdk2443; + +static struct snd_soc_dai_link smdk2443_dai[] = { +{ + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], + .codec_dai = &ac97_dai, +}, +}; + +static struct snd_soc_card smdk2443 = { + .name = "SMDK2443", + .platform = &s3c24xx_soc_platform, + .dai_link = smdk2443_dai, + .num_links = ARRAY_SIZE(smdk2443_dai), +}; + +static struct snd_soc_device smdk2443_snd_ac97_devdata = { + .card = &smdk2443, + .codec_dev = &soc_codec_dev_ac97, +}; + +static struct platform_device *smdk2443_snd_ac97_device; + +static int __init smdk2443_init(void) +{ + int ret; + + smdk2443_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!smdk2443_snd_ac97_device) + return -ENOMEM; + + platform_set_drvdata(smdk2443_snd_ac97_device, + &smdk2443_snd_ac97_devdata); + smdk2443_snd_ac97_devdata.dev = &smdk2443_snd_ac97_device->dev; + ret = platform_device_add(smdk2443_snd_ac97_device); + + if (ret) + platform_device_put(smdk2443_snd_ac97_device); + + return ret; +} + +static void __exit smdk2443_exit(void) +{ + platform_device_unregister(smdk2443_snd_ac97_device); +} + +module_init(smdk2443_init); +module_exit(smdk2443_exit); + +/* Module information */ +MODULE_AUTHOR("Graeme Gregory, graeme.gregory(a)wolfsonmicro.com, www.wolfsonmicro.com"); +MODULE_DESCRIPTION("ALSA SoC WM9710 SMDK2443"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/smdk64xx_wm8580.c b/sound/soc/samsung/smdk64xx_wm8580.c new file mode 100644 index 0000000..07e8e51 --- /dev/null +++ b/sound/soc/samsung/smdk64xx_wm8580.c @@ -0,0 +1,266 @@ +/* + * smdk64xx_wm8580.c + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/platform_device.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/wm8580.h" +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +/* SMDK64XX has a 12MHZ crystal attached to WM8580 */ +#define SMDK64XX_WM8580_FREQ 12000000 + +static int smdk64xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int pll_out; + int bfs, rfs, ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + bfs = 16; + break; + case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S16_LE: + bfs = 32; + break; + default: + return -EINVAL; + } + + /* The Fvco for WM8580 PLLs must fall within [90,100]MHz. + * This criterion can't be met if we request PLL output + * as {8000x256, 64000x256, 11025x256}Hz. + * As a wayout, we rather change rfs to a minimum value that + * results in (params_rate(params) * rfs), and itself, acceptable + * to both - the CODEC and the CPU. + */ + switch (params_rate(params)) { + case 16000: + case 22050: + case 32000: + case 44100: + case 48000: + case 88200: + case 96000: + rfs = 256; + break; + case 64000: + rfs = 384; + break; + case 8000: + case 11025: + rfs = 512; + break; + default: + return -EINVAL; + } + pll_out = params_rate(params) * rfs; + + /* Set the Codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* Set the AP DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_CDCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* We use PCLK for basic ops in SoC-Slave mode */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Set WM8580 to drive MCLK from its PLLA */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, + WM8580_CLKSRC_PLLA); + if (ret < 0) + return ret; + + /* Explicitly set WM8580-DAC to source from MCLK */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_DAC_CLKSEL, + WM8580_CLKSRC_MCLK); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, + SMDK64XX_WM8580_FREQ, pll_out); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_BCLK, bfs); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, rfs); + if (ret < 0) + return ret; + + return 0; +} + +/* + * SMDK64XX WM8580 DAI operations. + */ +static struct snd_soc_ops smdk64xx_ops = { + .hw_params = smdk64xx_hw_params, +}; + +/* SMDK64xx Playback widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = { + SND_SOC_DAPM_HP("Front-L/R", NULL), + SND_SOC_DAPM_HP("Center/Sub", NULL), + SND_SOC_DAPM_HP("Rear-L/R", NULL), +}; + +/* SMDK64xx Capture widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = { + SND_SOC_DAPM_MIC("MicIn", NULL), + SND_SOC_DAPM_LINE("LineIn", NULL), +}; + +/* SMDK-PAIFTX connections */ +static const struct snd_soc_dapm_route audio_map_tx[] = { + /* MicIn feeds AINL */ + {"AINL", NULL, "MicIn"}, + + /* LineIn feeds AINL/R */ + {"AINL", NULL, "LineIn"}, + {"AINR", NULL, "LineIn"}, +}; + +/* SMDK-PAIFRX connections */ +static const struct snd_soc_dapm_route audio_map_rx[] = { + /* Front Left/Right are fed VOUT1L/R */ + {"Front-L/R", NULL, "VOUT1L"}, + {"Front-L/R", NULL, "VOUT1R"}, + + /* Center/Sub are fed VOUT2L/R */ + {"Center/Sub", NULL, "VOUT2L"}, + {"Center/Sub", NULL, "VOUT2R"}, + + /* Rear Left/Right are fed VOUT3L/R */ + {"Rear-L/R", NULL, "VOUT3L"}, + {"Rear-L/R", NULL, "VOUT3R"}, +}; + +static int smdk64xx_wm8580_init_paiftx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Capture widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt, + ARRAY_SIZE(wm8580_dapm_widgets_cpt)); + + /* Set up PAIFTX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx)); + + /* Enabling the microphone requires the fitting of a 0R + * resistor to connect the line from the microphone jack. + */ + snd_soc_dapm_disable_pin(codec, "MicIn"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static int smdk64xx_wm8580_init_paifrx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Playback widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk, + ARRAY_SIZE(wm8580_dapm_widgets_pbk)); + + /* Set up PAIFRX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx)); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link smdk64xx_dai[] = { +{ /* Primary Playback i/f */ + .name = "WM8580 PAIF RX", + .stream_name = "Playback", + .cpu_dai = &s3c64xx_i2s_v4_dai, + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX], + .init = smdk64xx_wm8580_init_paifrx, + .ops = &smdk64xx_ops, +}, +{ /* Primary Capture i/f */ + .name = "WM8580 PAIF TX", + .stream_name = "Capture", + .cpu_dai = &s3c64xx_i2s_v4_dai, + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX], + .init = smdk64xx_wm8580_init_paiftx, + .ops = &smdk64xx_ops, +}, +}; + +static struct snd_soc_card smdk64xx = { + .name = "smdk64xx", + .platform = &s3c24xx_soc_platform, + .dai_link = smdk64xx_dai, + .num_links = ARRAY_SIZE(smdk64xx_dai), +}; + +static struct snd_soc_device smdk64xx_snd_devdata = { + .card = &smdk64xx, + .codec_dev = &soc_codec_dev_wm8580, +}; + +static struct platform_device *smdk64xx_snd_device; + +static int __init smdk64xx_audio_init(void) +{ + int ret; + + smdk64xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!smdk64xx_snd_device) + return -ENOMEM; + + platform_set_drvdata(smdk64xx_snd_device, &smdk64xx_snd_devdata); + smdk64xx_snd_devdata.dev = &smdk64xx_snd_device->dev; + ret = platform_device_add(smdk64xx_snd_device); + + if (ret) + platform_device_put(smdk64xx_snd_device); + + return ret; +} +module_init(smdk64xx_audio_init); + +MODULE_AUTHOR("Jaswinder Singh, jassi.brar(a)samsung.com"); +MODULE_DESCRIPTION("ALSA SoC SMDK64XX WM8580"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/smdk_wm9713.c b/sound/soc/samsung/smdk_wm9713.c new file mode 100644 index 0000000..24fd39f --- /dev/null +++ b/sound/soc/samsung/smdk_wm9713.c @@ -0,0 +1,94 @@ +/* + * smdk_wm9713.c -- SoC audio for SMDK + * + * Copyright 2010 Samsung Electronics Co. Ltd. + * Author: Jaswinder Singh Brar <jassi.brar(a)samsung.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as + * published by the Free Software Foundation; either version 2 of the + * License, or (at your option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/device.h> +#include <sound/soc.h> + +#include "../codecs/wm9713.h" +#include "s3c-dma.h" +#include "s3c-ac97.h" + +static struct snd_soc_card smdk; + +/* + * Default CFG switch settings to use this driver: + * + * SMDK6410: Set CFG1 1-3 On, CFG2 1-4 Off + */ + +/* + Playback (HeadPhone):- + $ amixer sset 'Headphone' unmute + $ amixer sset 'Right Headphone Out Mux' 'Headphone' + $ amixer sset 'Left Headphone Out Mux' 'Headphone' + $ amixer sset 'Right HP Mixer PCM' unmute + $ amixer sset 'Left HP Mixer PCM' unmute + + Capture (LineIn):- + $ amixer sset 'Right Capture Source' 'Line' + $ amixer sset 'Left Capture Source' 'Line' +*/ + +static struct snd_soc_dai_link smdk_dai = { + .name = "AC97", + .stream_name = "AC97 PCM", + .cpu_dai = &s3c_ac97_dai[S3C_AC97_DAI_PCM], + .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI], +}; + +static struct snd_soc_card smdk = { + .name = "SMDK", + .platform = &s3c24xx_soc_platform, + .dai_link = &smdk_dai, + .num_links = 1, +}; + +static struct snd_soc_device smdk_snd_ac97_devdata = { + .card = &smdk, + .codec_dev = &soc_codec_dev_wm9713, +}; + +static struct platform_device *smdk_snd_ac97_device; + +static int __init smdk_init(void) +{ + int ret; + + smdk_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!smdk_snd_ac97_device) + return -ENOMEM; + + platform_set_drvdata(smdk_snd_ac97_device, + &smdk_snd_ac97_devdata); + smdk_snd_ac97_devdata.dev = &smdk_snd_ac97_device->dev; + + ret = platform_device_add(smdk_snd_ac97_device); + if (ret) + platform_device_put(smdk_snd_ac97_device); + + return ret; +} + +static void __exit smdk_exit(void) +{ + platform_device_unregister(smdk_snd_ac97_device); +} + +module_init(smdk_init); +module_exit(smdk_exit); + +/* Module information */ +MODULE_AUTHOR("Jaswinder Singh Brar, jassi.brar(a)samsung.com"); +MODULE_DESCRIPTION("ALSA SoC SMDK+WM9713"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

2 1

[alsa-devel] Writing a driver for the Blackfin BF537 with two CS42448 on SPORT1
by Adam Rosenberg 25 Jun '10

25 Jun '10

I am writing my first ALSA driver for uClinux on Blackfin BF537. I have a custom board with two Cirrus Logic CS42448 chips connected to the primary and secondary data lines of SPORT1. I was basing my driver off of the driver for the Analog Devices AD1836 but it does not look like that driver was written using the latest practices. I already have the code modified well enough to communicate with both chips and produce some noise, but it is obvious that I am not filling the DMA buffer properly. Please let me know where I can find a driver that would be a better example for me to reference. When I have completed the driver I will also need help on where to submit it so that it will be supported by ALSA for others. Thank you for your help, Adam Adam Rosenberg Software Engineer Alcorn McBride Inc. 3300 South Hiawassee Building 105 Orlando, FL 32835 (407) 296 - 5800 ext. 5490

2 1

[alsa-devel] [PATCH] eukrea-tlv320: add support for our i.MX25 board
by Eric Bénard 25 Jun '10

25 Jun '10

* tdm slot has to be configured to get sound working on i.MX25 Signed-off-by: Eric Bénard <eric(a)eukrea.com> --- sound/soc/imx/Kconfig | 2 +- sound/soc/imx/eukrea-tlv320.c | 3 ++- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index fd292f1..fd616ae 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -31,7 +31,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 bool "Eukrea TLV320" - depends on MACH_EUKREA_MBIMX27_BASEBOARD + depends on MACH_EUKREA_MBIMX27_BASEBOARD || MACH_EUKREA_MBIMXSD_BASEBOARD select SND_MXC_SOC_SSI select SND_SOC_TLV320AIC23 help diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c index 968380a..45f5e4b 100644 --- a/sound/soc/imx/eukrea-tlv320.c +++ b/sound/soc/imx/eukrea-tlv320.c @@ -60,6 +60,7 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, pr_err("%s: failed setting codec sysclk\n", __func__); return ret; } + snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, SND_SOC_CLOCK_IN); @@ -100,7 +101,7 @@ static int __init eukrea_tlv320_init(void) { int ret; - if (!machine_is_eukrea_cpuimx27()) + if (!machine_is_eukrea_cpuimx27() && !machine_is_eukrea_cpuimx25sd()) /* return happy. We might run on a totally different machine */ return 0; -- 1.6.3.3

3 13

[alsa-devel] [PATCH 1/2] ASoC: uda134x: replace a macro with a value in platform struct.
by Vladimir Zapolskiy 25 Jun '10

25 Jun '10

This change wipes out a hardcoded macro, which enables codec bias level control. Now is_powered_on_standby value shall be used instead. Signed-off-by: Vladimir Zapolskiy <vzapolskiy(a)gmail.com> --- include/sound/uda134x.h | 12 ++++++++++++ sound/soc/codecs/uda134x.c | 21 +++++---------------- 2 files changed, 17 insertions(+), 16 deletions(-) diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h index 509efb0..e475659 100644 --- a/include/sound/uda134x.h +++ b/include/sound/uda134x.h @@ -18,6 +18,18 @@ struct uda134x_platform_data { struct l3_pins l3; void (*power) (int); int model; + /* + ALSA SOC usually puts the device in standby mode when it's not used + for sometime. If you unset is_powered_on_standby the driver will + turn off the ADC/DAC when this callback is invoked and turn it back + on when needed. Unfortunately this will result in a very light bump + (it can be audible only with good earphones). If this bothers you + set is_powered_on_standby, you will have slightly higher power + consumption. Please note that sending the L3 command for ADC is + enough to make the bump, so it doesn't make difference if you + completely take off power from the codec. + */ + int is_powered_on_standby; #define UDA134X_UDA1340 1 #define UDA134X_UDA1341 2 #define UDA134X_UDA1344 3 diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index e8066b6..5c7b054 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -27,19 +27,6 @@ #include "uda134x.h" -#define POWER_OFF_ON_STANDBY 1 -/* - ALSA SOC usually puts the device in standby mode when it's not used - for sometime. If you define POWER_OFF_ON_STANDBY the driver will - turn off the ADC/DAC when this callback is invoked and turn it back - on when needed. Unfortunately this will result in a very light bump - (it can be audible only with good earphones). If this bothers you - just comment this line, you will have slightly higher power - consumption . Please note that sending the L3 command for ADC is - enough to make the bump, so it doesn't make difference if you - completely take off power from the codec. - */ - #define UDA134X_RATES SNDRV_PCM_RATE_8000_48000 #define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE) @@ -559,9 +546,11 @@ static int uda134x_soc_probe(struct platform_device *pdev) codec->num_dai = 1; codec->read = uda134x_read_reg_cache; codec->write = uda134x_write; -#ifdef POWER_OFF_ON_STANDBY - codec->set_bias_level = uda134x_set_bias_level; -#endif + + if (!pd->is_powered_on_standby) { + codec->set_bias_level = uda134x_set_bias_level; + } + INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); -- 1.7.0.4

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Re: [alsa-devel] redirecting audio to another desktop machine
by Hans-Kristian Arntzen 25 Jun '10

25 Jun '10

On 06/21/2010 12:00 PM, alsa-devel-request(a)alsa-project.org wrote: > Date: Mon, 21 Jun 2010 14:57:00 +0530 > From: Viral Mehta<Viral.Mehta(a)lntinfotech.com> > Subject: [alsa-devel] redirecting audio to another desktop machine > To: "alsa-devel(a)alsa-project.org"<alsa-devel(a)alsa-project.org> > Message-ID: > <70376CA23424B34D86F1C7DE6B9973430254343B3E(a)VSHINMSMBX01.vshodc.lntinfotech.com> > > Content-Type: text/plain; charset="us-ascii" > > Hi list, > > I have an embedded device running linux kernel and a minimal filesystem. > > I want to access this device remotely which means I should be able to connect to the device just like we do Remote Desktop over ethernet. > I am able to do this. I am able to send key-event and framebuffer data whenever my desktop machine connects to the device. > > But, I am not able to send audio data on remote side. > I am really not sure which device file (/dev/snd/ ???) I should read and capture the data and should send across.... > Or can I read the playback device ? I think NO. > Do I need to create some kind of virtual sound device ? > > Any pointers/suggestions are appreciated. > > Thanks, > Viral > > > If you want to reroute an ALSA application over the network, you can use output plugins. Afaik, both PulseAudio and RSound provides plugins for this. You generally want to set up a virtual device in .asoundrc or similar. -- Hans-Kristian Arntzen

3 5

[alsa-devel] AndreaElectronics PureAudio USB-SA
by rmfc 25 Jun '10

25 Jun '10

Hello all, I'm trying to use any USB sound card with some DSP processing that allows a microphone array to capture audio and use in speech recognition (mostly applied in noisy environments). The best device that I found so far is AndreaElectronics PureAudio USB-SA but unfortunately it isn't properly supported in Linux/Alsa. The device is recognized as an USB Audio device (snd_usb_audio) and I can record audio in Audacity. The problem is that none of DSP features is explicitly activated in USB configuration so it works almost like any other microphone. I then tried Asus Xonar U1 but the problem is the same, no configuration is done in such devices, so they record audio without enough quality to be used by a speech recognizer (not enough energy, audio DC components when software gain is applied in Sound Preferences...). Is there any way to configure an USB Audio device interfaces? Here's the output of "lsusb" of AndreaElectronics USB-SA sound card. ------------------------------------------------------------ Bus 003 Device 002: ID 08a8:000c Andrea Electronics Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x08a8 Andrea Electronics idProduct 0x000c bcdDevice 1.00 iManufacturer 3 Andrea Electronics iProduct 1 Andrea PureAudio USB-SA iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 293 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 128 bInCollection 2 baInterfaceNr( 0) 1 baInterfaceNr( 1) 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0603 Line Connector bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 6 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 9 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 8 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 5 (SELECTOR_UNIT) bUnitID 8 bNrInPins 3 baSource( 0) 10 baSource( 1) 11 baSource( 2) 12 iSelector 0 AudioControl Interface Descriptor: bLength 13 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 9 bSourceID 1 bControlSize 2 bmaControls( 0) 0x01 bmaControls( 0) 0x02 Mute Loudness bmaControls( 1) 0x02 bmaControls( 1) 0x00 Volume bmaControls( 2) 0x02 bmaControls( 2) 0x00 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 10 bSourceID 2 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 11 bSourceID 3 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 12 bSourceID 4 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x00 bmaControls( 2) 0x00 iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 4 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 44100 tSamFreq[ 3] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 1 Milliseconds wLockDelay 1 Milliseconds Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 4 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 44100 tSamFreq[ 3] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.00 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 50 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x87 EP 7 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 1 Device Status: 0x0000 (Bus Powered) ------------------------------------------------------------ Renato

2 1

[alsa-devel] [PATCH] ASoC: DaVinci: Added support for stereo I2S
by Raffaele Recalcati 25 Jun '10

25 Jun '10

From: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Added audio playback support with [frame sync master - clock master] mode and with [frame sync master - clock slave]. Clock slave can be important when the external codec need system clock and bit clock synchronized. In the clock master case there is a FIXME message in the source code, because we (Davide and myself) have guessed a frequency of 122000000 that seems the base to be divided. This patch has been developed against the http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git git tree and has been tested on bmx board (similar to dm365 evm, but using uda1345 as external audio codec). Signed-off-by: Raffaele Recalcati <raffaele.recalcati(a)bticino.it> Signed-off-by: Davide Bonfanti <davide.bonfanti(a)bticino.it> --- arch/arm/mach-davinci/include/mach/asp.h | 7 ++ sound/soc/davinci/davinci-i2s.c | 141 ++++++++++++++++++++++++++---- 2 files changed, 129 insertions(+), 19 deletions(-) diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index 834725f..b1faeb9 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -63,6 +63,13 @@ struct snd_platform_data { unsigned sram_size_playback; unsigned sram_size_capture; + /* + * This define works when both clock and FS are output for the cpu + * and makes clock very fast (FS is not simmetrical, but sampling + * frequency is better approximated + */ + int i2s_fast_clock; + /* McASP specific fields */ int tdm_slots; u8 op_mode; diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index adadcd3..8811d25 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -68,16 +68,23 @@ #define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_RCR_RFIG (1 << 18) #define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_RCR_RFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_RCR_RPHASE (1 << 31) #define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5) #define DAVINCI_MCBSP_XCR_XFRLEN1(v) ((v) << 8) #define DAVINCI_MCBSP_XCR_XDATDLY(v) ((v) << 16) #define DAVINCI_MCBSP_XCR_XFIG (1 << 18) #define DAVINCI_MCBSP_XCR_XWDLEN2(v) ((v) << 21) +#define DAVINCI_MCBSP_XCR_XFRLEN2(v) ((v) << 24) +#define DAVINCI_MCBSP_XCR_XPHASE (1 << 31) + +#define CLKGDV(v) (v) /* Bits 0:7 */ #define DAVINCI_MCBSP_SRGR_FWID(v) ((v) << 8) #define DAVINCI_MCBSP_SRGR_FPER(v) ((v) << 16) #define DAVINCI_MCBSP_SRGR_FSGM (1 << 28) +#define DAVINCI_MCBSP_SRGR_CLKSM (1 << 29) #define DAVINCI_MCBSP_PCR_CLKRP (1 << 0) #define DAVINCI_MCBSP_PCR_CLKXP (1 << 1) @@ -144,8 +151,17 @@ struct davinci_mcbsp_dev { * won't end up being swapped because of the underrun. */ unsigned enable_channel_combine:1; + + int i2s_fast_clock; +}; + +struct davinci_mcbsp_data { + unsigned int fmt; + int clk_div; }; +static struct davinci_mcbsp_data mcbsp_data; + static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, int reg, u32 val) { @@ -255,24 +271,27 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int pcr; unsigned int srgr; srgr = DAVINCI_MCBSP_SRGR_FSGM | - DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) | - DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1); + DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) | + DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1); + /* Attention srgr is updated by hw_params! */ + mcbsp_data.fmt = fmt; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFM: case SND_SOC_DAIFMT_CBS_CFS: /* cpu is master */ pcr = DAVINCI_MCBSP_PCR_FSXM | - DAVINCI_MCBSP_PCR_FSRM | - DAVINCI_MCBSP_PCR_CLKXM | - DAVINCI_MCBSP_PCR_CLKRM; + DAVINCI_MCBSP_PCR_FSRM | + DAVINCI_MCBSP_PCR_CLKXM | + DAVINCI_MCBSP_PCR_CLKRM; break; case SND_SOC_DAIFMT_CBM_CFS: /* McBSP CLKR pin is the input for the Sample Rate Generator. * McBSP FSR and FSX are driven by the Sample Rate Generator. */ pcr = DAVINCI_MCBSP_PCR_SCLKME | - DAVINCI_MCBSP_PCR_FSXM | - DAVINCI_MCBSP_PCR_FSRM; + DAVINCI_MCBSP_PCR_FSXM | + DAVINCI_MCBSP_PCR_FSRM; break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is master */ @@ -372,6 +391,17 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return 0; } +static int davinci_i2s_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + int srgr; + + mcbsp_data.clk_div = div; + return 0; +} + + static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -380,11 +410,12 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct davinci_pcm_dma_params *dma_params = &dev->dma_params[substream->stream]; struct snd_interval *i = NULL; - int mcbsp_word_length; - unsigned int rcr, xcr, srgr; + int mcbsp_word_length, master; + unsigned int rcr, xcr, srgr, clk_div, freq, framesize; u32 spcr; snd_pcm_format_t fmt; unsigned element_cnt = 1; + struct clk *clk; /* general line settings */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); @@ -396,12 +427,60 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr); } - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); - srgr = DAVINCI_MCBSP_SRGR_FSGM; - srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + master = mcbsp_data.fmt & SND_SOC_DAIFMT_MASTER_MASK; + fmt = params_format(params); + mcbsp_word_length = asp_word_length[fmt]; - i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); - srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + if (master == SND_SOC_DAIFMT_CBS_CFS) { + clk = clk_get(NULL, "pll1_sysclk6"); + if (clk) + freq = clk_get_rate(clk); + freq = 122000000; /* FIXME ask to Texas */ + if (dev->i2s_fast_clock) { + clk_div = 256; + do { + framesize = (freq / (--clk_div)) / + params->rate_num * + params->rate_den; + } while (((framesize < 33) || (framesize > 4095)) && + (clk_div)); + clk_div--; + srgr = DAVINCI_MCBSP_SRGR_FSGM | + DAVINCI_MCBSP_SRGR_CLKSM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * + 8 - 1); + srgr |= DAVINCI_MCBSP_SRGR_FPER(framesize - 1); + } else { + /* simmetric waveforms */ + srgr = DAVINCI_MCBSP_SRGR_FSGM | + DAVINCI_MCBSP_SRGR_CLKSM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * + 8 - 1); + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * + 16 - 1); + clk_div = freq / (mcbsp_word_length * 16) / + params->rate_num * params->rate_den; + } + clk_div &= 0xFF; + srgr |= clk_div; + } else if (master == SND_SOC_DAIFMT_CBS_CFM) { + /* Clock given on CLKS */ + srgr = DAVINCI_MCBSP_SRGR_FSGM; + clk_div = mcbsp_data.clk_div - 1; + srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1); + srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * 16 - 1); + clk_div &= 0xFF; + srgr |= clk_div; + } else { + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); + srgr = DAVINCI_MCBSP_SRGR_FSGM; + srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1); + pr_debug("%s - %d FWID set: re-read srgr = %X\n", + __func__, __LINE__, snd_interval_value(i) - 1); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); + srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1); + } davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr); rcr = DAVINCI_MCBSP_RCR_RFIG; @@ -426,22 +505,45 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, element_cnt = 1; fmt = double_fmt[fmt]; } + if (master == SND_SOC_DAIFMT_CBS_CFS || + master == SND_SOC_DAIFMT_CBS_CFM) { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(0); + rcr |= DAVINCI_MCBSP_RCR_RPHASE; + xcr |= DAVINCI_MCBSP_XCR_XPHASE; + } else { + /* FIXME ask to Texas */ + rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(element_cnt - 1); + } } dma_params->acnt = dma_params->data_type = data_type[fmt]; dma_params->fifo_level = 0; mcbsp_word_length = asp_word_length[fmt]; - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + + if (master == SND_SOC_DAIFMT_CBS_CFS || + master == SND_SOC_DAIFMT_CBS_CFM) { + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(0); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(0); + } else { + /* FIXME ask to Texas */ + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); + } rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | - DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); + DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); xcr |= DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) | - DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length); + DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr); else davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr); + + pr_debug("%s - %d srgr=%X\n", __func__, __LINE__, srgr); + pr_debug("%s - %d xcr=%X\n", __func__, __LINE__, xcr); + pr_debug("%s - %d rcr=%X\n", __func__, __LINE__, rcr); return 0; } @@ -500,7 +602,7 @@ static struct snd_soc_dai_ops davinci_i2s_dai_ops = { .trigger = davinci_i2s_trigger, .hw_params = davinci_i2s_hw_params, .set_fmt = davinci_i2s_set_dai_fmt, - + .set_clkdiv = davinci_i2s_dai_set_clkdiv, }; struct snd_soc_dai davinci_i2s_dai = { @@ -548,6 +650,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) } if (pdata) { dev->enable_channel_combine = pdata->enable_channel_combine; + dev->i2s_fast_clock = pdata->i2s_fast_clock; dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].sram_size = pdata->sram_size_playback; dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = -- 1.7.0.4

3 3

[alsa-devel] Compilation error
by lars.bjorndal＠broadpark.no 25 Jun '10

25 Jun '10

I get the following output while trying to compile the alsa driver on my Fedora 12 system. It worked well with the previous kernel version. # ./alsa-compile.sh --install File environment.base is cached. > cd /tmp/alsa-compile-script File environment.compile is cached. alsa-driver tree alsa-driver-1.0.23.31.g2379a.273.g5b97d is present. Reusing it. Use './alsa-compile.sh --clean=alsa-driver' command to refetch and rebuild. Sources unpacked to alsa-driver-1.0.23.31.g2379a.273.g5b97d > cd alsa-driver-1.0.23.31.g2379a.273.g5b97d Running make: make dep make[1]: Entering directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d' make[2]: Entering directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include' make -C sound prepare make[3]: Entering directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include/sound' make prepare2 make[4]: Entering directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include/sound' make[4]: Nothing to be done for `prepare2'. make[4]: Leaving directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include/sound' make[3]: Leaving directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include/sound' make[2]: Leaving directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/include' ... make[2]: Entering directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/misc' make[2]: Leaving directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/misc' make[1]: Leaving directory `/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d' make -C /lib/modules/2.6.32.14-127.fc12.i686.PAE/build SUBDIRS=/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d CPP="gcc -E" CC="gcc" modules make[1]: Entering directory `/usr/src/kernels/2.6.32.14-127.fc12.i686.PAE' CC [M] /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/ua101.o In file included from /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/ua101.c:2: /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c: In function 'alloc_stream_buffers': /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c:1071: error: implicit declaration of function 'usb_alloc_coherent' /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c:1072: warning: assignment makes pointer from integer without a cast /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c: In function 'free_stream_buffers': /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c:1092: error: implicit declaration of function 'usb_free_coherent' /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c: In function 'ua101_card_free': /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/../../alsa-kernel/usb/misc/ua101.c:1172: warning: unused variable 'ua' make[4]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc/ua101.o] Error 1 make[3]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb/misc] Error 2 make[2]: *** [/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d/usb] Error 2 make[1]: *** [_module_/tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.32.14-127.fc12.i686.PAE' make: *** [compile] Error 2 Working directory /tmp/alsa-compile-script/alsa-driver-1.0.23.31.g2379a.273.g5b97d. Compilation of alsa-driver failed. Report this problem to the <alsa-devel(a)alsa-project.org> mailing list. Lars

1 0

[alsa-devel] Verifying mixer dB data/Invalid dB data from USB cards, especially Aureon 5.1 MkII
by Lennart Poettering 25 Jun '10

25 Jun '10

Heya, a while back I wrote a little tool "dbmeasure" which can be used to determine dB attenuation data for alsa mixer controls. I have now extended this little dB tools suite to include a tiny tool "dbverify" that can be used to verify the correctness of existing dB data exposed by the drivers. It should be relatively easy to use that tool, suitable even for non-guru folks who want to debug the dB information from alsa. http://git.0pointer.de/?p=dbmeasure.git git://git.0pointer.de/dbmeasure.git Just run "make" in a git checkout and then try: $ ./dbverify Aureon51MkII Master 30 200 That will verify the dB data of the "Master" control of the sound card that goes by the id "Aureon51MkII". It will compare the discrete volume steps 30 and 200: first it will play a full amplitude sine wave at mixer volume step 30, and then will change to mixer volume step 200, but attenuate the sine wave in software compensating for the volume change. So once you hear the sine wave attenuated by the hw, and once by the software. If the card's dB data is correct both sines should have the same volume. Now, playing around with this, I could verify what I already mentioned to Jarsolav elsewhere earlier: it seems that particularly USB cards seem to expose invalid dB data (at least all mine do). For example, for the above mentioned Aureon 5.1 it's really *way* off -- if you posess that card just run the line above, and listen to the difference. It's really bad. OTOH the dB data of the HDA cards I own is pretty accurate as it seems, at least I was unable to decide just by listening which sine wave was attenuated by sw, and which by hw. Getting back to the invalid dB data from the USB cards: the question is whether the USB descriptor data is badly parsed, and the dB mismatch hence systematic for USB cards, or if these cards are just crappy and include invalid data? In which case I wonder what we could do about that? The Aureon's dB range already looks suspicous, since both the max and the min dB value are < 0 (-47.69 to -1.97), so maybe we could add some heuristics to filter out data that already looks suspicous? Invalid dB data from the driver is a real problem for the "flat volume" logic in PA. We basically allow each app to control the full hw volume range individually, and then set the hw volume to the max of what all apps wanted and attenuate the other streams accordingly. On the Aureon this doesn't work at all, since the attenuation of the streams is miscalculated due to the invalid dB data. Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4

8 93

[alsa-devel] [RFC 5/5] ASoC multi-component support : jack
by Liam Girdwood 24 Jun '10

24 Jun '10

Cache configuration is now stored as part of the codec driver structure. Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/soc-cache.c | 34 +++++++++++++++++----------------- 1 files changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 472af38..83cd8ed 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -19,7 +19,7 @@ static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec, unsigned int reg) { u16 *cache = codec->reg_cache; - if (reg >= codec->reg_cache_size) + if (reg >= codec->driver->reg_cache_size) return -1; return cache[reg]; } @@ -31,12 +31,12 @@ static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, u8 data[2]; int ret; - BUG_ON(codec->volatile_register); + BUG_ON(codec->driver->volatile_register); data[0] = (reg << 4) | ((value >> 8) & 0x000f); data[1] = value & 0x00ff; - if (reg < codec->reg_cache_size) + if (reg < codec->driver->reg_cache_size) cache[reg] = value; if (codec->cache_only) { @@ -89,7 +89,7 @@ static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec, unsigned int reg) { u16 *cache = codec->reg_cache; - if (reg >= codec->reg_cache_size) + if (reg >= codec->driver->reg_cache_size) return -1; return cache[reg]; } @@ -101,12 +101,12 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, u8 data[2]; int ret; - BUG_ON(codec->volatile_register); + BUG_ON(codec->driver->volatile_register); data[0] = (reg << 1) | ((value >> 8) & 0x0001); data[1] = value & 0x00ff; - if (reg < codec->reg_cache_size) + if (reg < codec->driver->reg_cache_size) cache[reg] = value; if (codec->cache_only) { @@ -161,13 +161,13 @@ static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, u8 *cache = codec->reg_cache; u8 data[2]; - BUG_ON(codec->volatile_register); + BUG_ON(codec->driver->volatile_register); reg &= 0xff; data[0] = reg; data[1] = value & 0xff; - if (reg < codec->reg_cache_size) + if (reg < codec->driver->reg_cache_size) cache[reg] = value; if (codec->cache_only) { @@ -188,7 +188,7 @@ static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, { u8 *cache = codec->reg_cache; reg &= 0xff; - if (reg >= codec->reg_cache_size) + if (reg >= codec->driver->reg_cache_size) return -1; return cache[reg]; } @@ -224,7 +224,7 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec, { u16 *cache = codec->reg_cache; - if (reg >= codec->reg_cache_size || + if (reg >= codec->driver->reg_cache_size || snd_soc_codec_volatile_register(codec, reg)) { if (codec->cache_only) return -EINVAL; @@ -343,7 +343,7 @@ static unsigned int snd_soc_16_8_read(struct snd_soc_codec *codec, u16 *cache = codec->reg_cache; reg &= 0xff; - if (reg >= codec->reg_cache_size) + if (reg >= codec->driver->reg_cache_size) return -1; return cache[reg]; } @@ -355,14 +355,14 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, u8 data[3]; int ret; - BUG_ON(codec->volatile_register); + BUG_ON(codec->driver->volatile_register); data[0] = (reg >> 8) & 0xff; data[1] = reg & 0xff; data[2] = value; reg &= 0xff; - if (reg < codec->reg_cache_size) + if (reg < codec->driver->reg_cache_size) cache[reg] = value; if (codec->cache_only) { @@ -451,7 +451,7 @@ static unsigned int snd_soc_16_16_read(struct snd_soc_codec *codec, { u16 *cache = codec->reg_cache; - if (reg >= codec->reg_cache_size || + if (reg >= codec->driver->reg_cache_size || snd_soc_codec_volatile_register(codec, reg)) { if (codec->cache_only) return -EINVAL; @@ -474,7 +474,7 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, data[2] = (value >> 8) & 0xff; data[3] = value & 0xff; - if (reg < codec->reg_cache_size) + if (reg < codec->driver->reg_cache_size) cache[reg] = value; if (codec->cache_only) { @@ -571,8 +571,8 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, return -EINVAL; } - codec->write = io_types[i].write; - codec->read = io_types[i].read; + codec->driver->write = io_types[i].write; + codec->driver->read = io_types[i].read; switch (control) { case SND_SOC_CUSTOM: -- 1.7.0.4

1 0

[alsa-devel] [RFC 4/5] ASoC multi-component support : jack
by Liam Girdwood 24 Jun '10

24 Jun '10

Jack interface is per codec rather than per card in order to support sound cards with more than one codec. Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- sound/soc/soc-jack.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 29159e1..1dfc4fd 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -32,14 +32,14 @@ * Returns zero if successful, or a negative error code on failure. * On success jack will be initialised. */ -int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type, +int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, struct snd_soc_jack *jack) { - jack->card = card; + jack->codec = codec; INIT_LIST_HEAD(&jack->pins); BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier); - return snd_jack_new(card->codec->card, id, type, &jack->jack); + return snd_jack_new(codec->snd_card, id, type, &jack->jack); } EXPORT_SYMBOL_GPL(snd_soc_jack_new); @@ -67,7 +67,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) if (!jack) return; - codec = jack->card->codec; + codec = jack->codec; mutex_lock(&codec->mutex); @@ -268,7 +268,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, ret = request_irq(gpio_to_irq(gpios[i].gpio), gpio_handler, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, - jack->card->dev->driver->name, + jack->codec->dev->driver->name, &gpios[i]); if (ret) goto err; -- 1.7.0.4

1 0

[alsa-devel] [RFC 2/5] ASoC multi-component support : DAPM
by Liam Girdwood 24 Jun '10

24 Jun '10

This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Also added void snd_soc_dapm_shutdown() to shutdown all CODECs in a card. Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- include/sound/soc-dapm.h | 8 ++-- sound/soc/soc-dapm.c | 82 +++++++++++++++++++++++----------------------- 2 files changed, 45 insertions(+), 45 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index c5d9987..c4a4456 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -322,14 +322,14 @@ int snd_soc_dapm_new_controls(struct snd_soc_codec *codec, /* dapm path setup */ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec); -void snd_soc_dapm_free(struct snd_soc_device *socdev); +void snd_soc_dapm_free(struct snd_soc_codec *codec); int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, const struct snd_soc_dapm_route *route, int num); /* dapm events */ -int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, - int event); -void snd_soc_dapm_shutdown(struct snd_soc_device *socdev); +int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, + const char *stream, int event); +void snd_soc_dapm_shutdown(struct snd_soc_card *card); /* dapm sys fs - used by the core */ int snd_soc_dapm_sys_add(struct device *dev); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 03cb7c0..a0d143d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -112,43 +112,41 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( /** * snd_soc_dapm_set_bias_level - set the bias level for the system - * @socdev: audio device + * @card: audio device * @level: level to configure * * Configure the bias (power) levels for the SoC audio device. * * Returns 0 for success else error. */ -static int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev, - enum snd_soc_bias_level level) +static int snd_soc_dapm_set_bias_level(struct snd_soc_card *card, + struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - struct snd_soc_card *card = socdev->card; - struct snd_soc_codec *codec = socdev->card->codec; int ret = 0; switch (level) { case SND_SOC_BIAS_ON: - dev_dbg(socdev->dev, "Setting full bias\n"); + dev_dbg(codec->dev, "Setting full bias\n"); break; case SND_SOC_BIAS_PREPARE: - dev_dbg(socdev->dev, "Setting bias prepare\n"); + dev_dbg(codec->dev, "Setting bias prepare\n"); break; case SND_SOC_BIAS_STANDBY: - dev_dbg(socdev->dev, "Setting standby bias\n"); + dev_dbg(codec->dev, "Setting standby bias\n"); break; case SND_SOC_BIAS_OFF: - dev_dbg(socdev->dev, "Setting bias off\n"); + dev_dbg(codec->dev, "Setting bias off\n"); break; default: - dev_err(socdev->dev, "Setting invalid bias %d\n", level); + dev_err(codec->dev, "Setting invalid bias %d\n", level); return -EINVAL; } - if (card->set_bias_level) + if (card && card->set_bias_level) ret = card->set_bias_level(card, level); if (ret == 0) { - if (codec->set_bias_level) - ret = codec->set_bias_level(codec, level); + if (codec->driver->set_bias_level) + ret = codec->driver->set_bias_level(codec, level); else codec->bias_level = level; } @@ -370,7 +368,7 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w, path->long_name); - ret = snd_ctl_add(codec->card, path->kcontrol); + ret = snd_ctl_add(codec->snd_card, path->kcontrol); if (ret < 0) { printk(KERN_ERR "asoc: failed to add dapm kcontrol %s: %d\n", path->long_name, @@ -398,7 +396,7 @@ static int dapm_new_mux(struct snd_soc_codec *codec, } kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name); - ret = snd_ctl_add(codec->card, kcontrol); + ret = snd_ctl_add(codec->snd_card, kcontrol); if (ret < 0) goto err; @@ -893,7 +891,7 @@ static void dapm_seq_run(struct snd_soc_codec *codec, struct list_head *list, */ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) { - struct snd_soc_device *socdev = codec->socdev; + struct snd_soc_card *card = codec->card; struct snd_soc_dapm_widget *w; LIST_HEAD(up_list); LIST_HEAD(down_list); @@ -966,7 +964,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) } if (sys_power && codec->bias_level == SND_SOC_BIAS_OFF) { - ret = snd_soc_dapm_set_bias_level(socdev, + ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_STANDBY); if (ret != 0) pr_err("Failed to turn on bias: %d\n", ret); @@ -975,8 +973,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) /* If we're changing to all on or all off then prepare */ if ((sys_power && codec->bias_level == SND_SOC_BIAS_STANDBY) || (!sys_power && codec->bias_level == SND_SOC_BIAS_ON)) { - ret = snd_soc_dapm_set_bias_level(socdev, - SND_SOC_BIAS_PREPARE); + ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_PREPARE); if (ret != 0) pr_err("Failed to prepare bias: %d\n", ret); } @@ -989,8 +986,7 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) /* If we just powered the last thing off drop to standby bias */ if (codec->bias_level == SND_SOC_BIAS_PREPARE && !sys_power) { - ret = snd_soc_dapm_set_bias_level(socdev, - SND_SOC_BIAS_STANDBY); + ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_STANDBY); if (ret != 0) pr_err("Failed to apply standby bias: %d\n", ret); } @@ -998,15 +994,14 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) /* If we're in standby and can support bias off then do that */ if (codec->bias_level == SND_SOC_BIAS_STANDBY && codec->idle_bias_off) { - ret = snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_OFF); + ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_OFF); if (ret != 0) pr_err("Failed to turn off bias: %d\n", ret); } /* If we just powered up then move to active bias */ if (codec->bias_level == SND_SOC_BIAS_PREPARE && sys_power) { - ret = snd_soc_dapm_set_bias_level(socdev, - SND_SOC_BIAS_ON); + ret = snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_ON); if (ret != 0) pr_err("Failed to apply active bias: %d\n", ret); } @@ -1188,8 +1183,7 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, static ssize_t dapm_widget_show(struct device *dev, struct device_attribute *attr, char *buf) { - struct snd_soc_device *devdata = dev_get_drvdata(dev); - struct snd_soc_codec *codec = devdata->card->codec; + struct snd_soc_codec *codec = dev_get_drvdata(dev); struct snd_soc_dapm_widget *w; int count = 0; char *state = "not set"; @@ -1998,9 +1992,10 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls); * * Returns 0 for success else error. */ -int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, - char *stream, int event) +int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, + const char *stream, int event) { + struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_widget *w; if (stream == NULL) @@ -2168,25 +2163,19 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend); /** * snd_soc_dapm_free - free dapm resources - * @socdev: SoC device + * @card: SoC device * * Free all dapm widgets and resources. */ -void snd_soc_dapm_free(struct snd_soc_device *socdev) +void snd_soc_dapm_free(struct snd_soc_codec *codec) { - struct snd_soc_codec *codec = socdev->card->codec; - - snd_soc_dapm_sys_remove(socdev->dev); + snd_soc_dapm_sys_remove(codec->dev); dapm_free_widgets(codec); } EXPORT_SYMBOL_GPL(snd_soc_dapm_free); -/* - * snd_soc_dapm_shutdown - callback for system shutdown - */ -void snd_soc_dapm_shutdown(struct snd_soc_device *socdev) +static void soc_dapm_shutdown_codec(struct snd_soc_codec *codec) { - struct snd_soc_codec *codec = socdev->card->codec; struct snd_soc_dapm_widget *w; LIST_HEAD(down_list); int powerdown = 0; @@ -2203,12 +2192,23 @@ void snd_soc_dapm_shutdown(struct snd_soc_device *socdev) * standby. */ if (powerdown) { - snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_PREPARE); + snd_soc_dapm_set_bias_level(NULL, codec, SND_SOC_BIAS_PREPARE); dapm_seq_run(codec, &down_list, 0, dapm_down_seq); - snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_STANDBY); + snd_soc_dapm_set_bias_level(NULL, codec, SND_SOC_BIAS_STANDBY); } +} + +/* + * snd_soc_dapm_shutdown - callback for system shutdown + */ +void snd_soc_dapm_shutdown(struct snd_soc_card *card) +{ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, list) + soc_dapm_shutdown_codec(codec); - snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_OFF); + snd_soc_dapm_set_bias_level(card, codec, SND_SOC_BIAS_OFF); } /* Module information */ -- 1.7.0.4

1 0

[alsa-devel] [RFC 1/5] ASoC multi-component support : core API
by Liam Girdwood 24 Jun '10

24 Jun '10

This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. Signed-off-by: Liam Girdwood <lrg(a)slimlogic.co.uk> --- include/sound/soc-dai.h | 99 ++++++++++++------- include/sound/soc.h | 246 +++++++++++++++++++++++++++++------------------ sound/soc/soc-core.c | 2 + 3 files changed, 218 insertions(+), 129 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 377693a..495f0aa 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -91,15 +91,17 @@ struct snd_pcm_substream; SNDRV_PCM_FMTBIT_S32_LE |\ SNDRV_PCM_FMTBIT_S32_BE) -struct snd_soc_dai_ops; +struct snd_soc_dai_driver; struct snd_soc_dai; struct snd_ac97_bus_ops; /* Digital Audio Interface registration */ -int snd_soc_register_dai(struct snd_soc_dai *dai); -void snd_soc_unregister_dai(struct snd_soc_dai *dai); -int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count); -void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count); +int snd_soc_register_dai(struct device *dev, int id, + struct snd_soc_dai_driver *dai_drv); +void snd_soc_unregister_dai(struct device *dev, int id); +int snd_soc_register_dais(struct device *dev, + struct snd_soc_dai_driver *dai_drv, size_t count); +void snd_soc_unregister_dais(struct device *dev, size_t count); /* Digital Audio Interface clocking API.*/ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, @@ -126,16 +128,6 @@ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); /* Digital Audio Interface mute */ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute); -/* - * Digital Audio Interface. - * - * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97 - * operations and capabilities. Codec and platform drivers will register this - * structure for every DAI they have. - * - * This structure covers the clocking, formating and ALSA operations for each - * interface. - */ struct snd_soc_dai_ops { /* * DAI clocking configuration, all optional. @@ -191,24 +183,24 @@ struct snd_soc_dai_ops { }; /* - * Digital Audio Interface runtime data. + * Digital Audio Interface Driver. * - * Holds runtime data for a DAI. + * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97 + * operations and capabilities. Codec and platform drivers will register this + * structure for every DAI they have. + * + * This structure covers the clocking, formating and ALSA operations for each + * interface. */ -struct snd_soc_dai { +struct snd_soc_dai_driver { /* DAI description */ - char *name; - unsigned int id; + const char *name; + struct module *owner; int ac97_control; - struct device *dev; - void *ac97_pdata; /* platform_data for the ac97 codec */ - - /* DAI callbacks */ - int (*probe)(struct platform_device *pdev, - struct snd_soc_dai *dai); - void (*remove)(struct platform_device *pdev, - struct snd_soc_dai *dai); + /* DAI driver callbacks */ + int (*probe)(struct snd_soc_dai *dai); + int (*remove)(struct snd_soc_dai *dai); int (*suspend)(struct snd_soc_dai *dai); int (*resume)(struct snd_soc_dai *dai); @@ -219,26 +211,50 @@ struct snd_soc_dai { struct snd_soc_pcm_stream capture; struct snd_soc_pcm_stream playback; unsigned int symmetric_rates:1; +}; + +/* + * Digital Audio Interface runtime data. + * + * Holds runtime data for a DAI. + */ +struct snd_soc_dai { + const char *name; + int id; + struct device *dev; + void *ac97_pdata; /* platform_data for the ac97 codec */ + + /* driver ops */ + struct snd_soc_dai_driver *driver; /* DAI runtime info */ - struct snd_soc_codec *codec; + unsigned int capture_active:1; /* stream is in use */ + unsigned int playback_active:1; /* stream is in use */ + unsigned int symmetric_rates:1; + struct snd_pcm_runtime *runtime; unsigned int active; unsigned char pop_wait:1; + unsigned char probed:1; - /* DAI private data */ - void *private_data; + /* DAI DMA data */ + void *playback_dma_data; + void *capture_dma_data; - /* parent platform */ - struct snd_soc_platform *platform; + /* parent platform/codec */ + union { + struct snd_soc_platform *platform; + struct snd_soc_codec *codec; + }; struct list_head list; + struct list_head card_list; }; static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai, const struct snd_pcm_substream *ss) { return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - dai->playback.dma_data : dai->capture.dma_data; + dai->playback_dma_data : dai->capture_dma_data; } static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, @@ -246,9 +262,20 @@ static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, void *data) { if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) - dai->playback.dma_data = data; + dai->playback_dma_data = data; else - dai->capture.dma_data = data; + dai->capture_dma_data = data; +} + +static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai, + void *data) +{ + dev_set_drvdata(dai->dev, data); +} + +static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai) +{ + return dev_get_drvdata(dai->dev); } #endif diff --git a/include/sound/soc.h b/include/sound/soc.h index 65e9d03..b3fe7a8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -228,13 +228,17 @@ struct snd_soc_ops; struct snd_soc_dai_mode; struct snd_soc_pcm_runtime; struct snd_soc_dai; +struct snd_soc_dai_driver; struct snd_soc_platform; struct snd_soc_dai_link; +struct snd_soc_platform_driver; struct snd_soc_codec; +struct snd_soc_codec_driver; struct soc_enum; struct snd_soc_ac97_ops; struct snd_soc_jack; struct snd_soc_jack_pin; + #ifdef CONFIG_GPIOLIB struct snd_soc_jack_gpio; #endif @@ -249,19 +253,18 @@ enum snd_soc_control_type { SND_SOC_SPI, }; -int snd_soc_register_platform(struct snd_soc_platform *platform); -void snd_soc_unregister_platform(struct snd_soc_platform *platform); -int snd_soc_register_codec(struct snd_soc_codec *codec); -void snd_soc_unregister_codec(struct snd_soc_codec *codec); +int snd_soc_register_platform(struct device *dev, int id, + struct snd_soc_platform_driver *platform_drv); +void snd_soc_unregister_platform(struct device *dev, int id); +int snd_soc_register_codec(struct device *dev, int id, + struct snd_soc_codec_driver *codec_drv, + struct snd_soc_dai_driver *dai_drv, int num_dai); +void snd_soc_unregister_codec(struct device *dev, int id); int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg); int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, int addr_bits, int data_bits, enum snd_soc_control_type control); -/* pcm <-> DAI connect */ -void snd_soc_free_pcms(struct snd_soc_device *socdev); -int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid); - /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params); @@ -273,7 +276,7 @@ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, const struct snd_pcm_hardware *hw); /* Jack reporting */ -int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type, +int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, struct snd_soc_jack *jack); void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask); int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, @@ -390,7 +393,7 @@ struct snd_soc_jack_gpio { struct snd_soc_jack { struct snd_jack *jack; - struct snd_soc_card *card; + struct snd_soc_codec *codec; struct list_head pins; int status; struct blocking_notifier_head notifier; @@ -398,15 +401,13 @@ struct snd_soc_jack { /* SoC PCM stream information */ struct snd_soc_pcm_stream { - char *stream_name; + const char *stream_name; u64 formats; /* SNDRV_PCM_FMTBIT_* */ unsigned int rates; /* SNDRV_PCM_RATE_* */ unsigned int rate_min; /* min rate */ unsigned int rate_max; /* max rate */ unsigned int channels_min; /* min channels */ unsigned int channels_max; /* max channels */ - unsigned int active; /* stream is in use */ - void *dma_data; /* used by platform code */ }; /* SoC audio ops */ @@ -419,44 +420,37 @@ struct snd_soc_ops { int (*trigger)(struct snd_pcm_substream *, int); }; -/* SoC Audio Codec */ +/* SoC Audio Codec device */ struct snd_soc_codec { - char *name; - struct module *owner; - struct mutex mutex; + const char *name; + int id; + struct device *dev; - struct snd_soc_device *socdev; + struct snd_soc_codec_driver *driver; + struct mutex mutex; + struct snd_card *snd_card; + struct snd_soc_card *card; struct list_head list; - - /* callbacks */ - int (*set_bias_level)(struct snd_soc_codec *, - enum snd_soc_bias_level level); + struct list_head card_list; + int num_dai; /* runtime */ - struct snd_card *card; struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */ unsigned int active; - unsigned int pcm_devs; - void *drvdata; + unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ + unsigned int cache_only:1; /* Suppress writes to hardware */ + unsigned int cache_sync:1; /* Cache needs to be synced to hardware */ + unsigned int suspended:1; /* Codec is in suspend PM state */ + unsigned int probed:1; /* Codec has been probed */ + unsigned int ac97_registered:1; /* Codec has been AC97 registered */ + unsigned int sysfs_registered:1; /* codec has been sysfs registered */ /* codec IO */ void *control_data; /* codec control (i2c/3wire) data */ - unsigned int (*read)(struct snd_soc_codec *, unsigned int); - int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); - int (*display_register)(struct snd_soc_codec *, char *, - size_t, unsigned int); - int (*volatile_register)(unsigned int); - int (*readable_register)(unsigned int); hw_write_t hw_write; unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int); void *reg_cache; - short reg_cache_size; - short reg_cache_step; - - unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */ - unsigned int cache_only:1; /* Suppress writes to hardware */ - unsigned int cache_sync:1; /* Cache needs to be synced to hardware */ /* dapm */ u32 pop_time; @@ -466,10 +460,6 @@ struct snd_soc_codec { enum snd_soc_bias_level suspend_bias_level; struct delayed_work delayed_work; - /* codec DAI's */ - struct snd_soc_dai *dai; - unsigned int num_dai; - #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_codec_root; struct dentry *debugfs_reg; @@ -478,23 +468,44 @@ struct snd_soc_codec { #endif }; -/* codec device */ -struct snd_soc_codec_device { - int (*probe)(struct platform_device *pdev); - int (*remove)(struct platform_device *pdev); - int (*suspend)(struct platform_device *pdev, pm_message_t state); - int (*resume)(struct platform_device *pdev); +/* codec driver */ +struct snd_soc_codec_driver { + const char *name; + struct module *owner; + + /* driver ops */ + int (*probe)(struct snd_soc_codec *); + int (*remove)(struct snd_soc_codec *); + int (*suspend)(struct snd_soc_codec *, + pm_message_t state); + int (*resume)(struct snd_soc_codec *); + + /* codec IO */ + unsigned int (*read)(struct snd_soc_codec *, unsigned int); + int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); + int (*display_register)(struct snd_soc_codec *, char *, + size_t, unsigned int); + int (*volatile_register)(unsigned int); + int (*readable_register)(unsigned int); + short reg_cache_size; + short reg_cache_step; + short reg_word_size; + const void *reg_cache_default; + + /* codec bias level */ + int (*set_bias_level)(struct snd_soc_codec *, + enum snd_soc_bias_level level); }; /* SoC platform interface */ -struct snd_soc_platform { - char *name; - struct list_head list; +struct snd_soc_platform_driver { + const char *name; + struct module *owner; - int (*probe)(struct platform_device *pdev); - int (*remove)(struct platform_device *pdev); - int (*suspend)(struct snd_soc_dai_link *dai_link); - int (*resume)(struct snd_soc_dai_link *dai_link); + int (*probe)(struct snd_soc_platform *); + int (*remove)(struct snd_soc_platform *); + int (*suspend)(struct snd_soc_dai *dai); + int (*resume)(struct snd_soc_dai *dai); /* pcm creation and destruction */ int (*pcm_new)(struct snd_card *, struct snd_soc_dai *, @@ -509,23 +520,35 @@ struct snd_soc_platform { struct snd_soc_dai *); /* platform stream ops */ - struct snd_pcm_ops *pcm_ops; + struct snd_pcm_ops *ops; }; -/* SoC machine DAI configuration, glues a codec and cpu DAI together */ -struct snd_soc_dai_link { - char *name; /* Codec name */ - char *stream_name; /* Stream name */ +struct snd_soc_platform { + const char *name; + int id; - /* DAI */ - struct snd_soc_dai *codec_dai; - struct snd_soc_dai *cpu_dai; + struct device *dev; + struct snd_soc_platform_driver *driver; - /* machine stream operations */ - struct snd_soc_ops *ops; + unsigned int suspended:1; /* platform is suspended */ + unsigned int probed:1; - /* codec/machine specific init - e.g. add machine controls */ - int (*init)(struct snd_soc_codec *codec); + struct list_head list; + struct list_head card_list; +}; + +struct snd_soc_dai_link { + /* config - must be set by machine driver */ + const char *name; /* Codec name */ + const char *stream_name; /* Stream name */ + int codec_id; /* for multi-codec */ + int platform_id; /* for multi-platform */ + int cpu_dai_id; + int codec_dai_id; + struct snd_soc_dai_driver *codec_dai_drv; + struct snd_soc_dai_driver *cpu_dai_drv; + struct snd_soc_platform_driver *platform_drv; + struct snd_soc_codec_driver *codec_drv; /* Keep DAI active over suspend */ unsigned int ignore_suspend:1; @@ -533,21 +556,24 @@ struct snd_soc_dai_link { /* Symmetry requirements */ unsigned int symmetric_rates:1; - /* Symmetry data - only valid if symmetry is being enforced */ - unsigned int rate; + /* codec/machine specific init - e.g. add machine controls */ + int (*init)(struct snd_soc_pcm_runtime *rtd); - /* DAI pcm */ - struct snd_pcm *pcm; + /* machine stream operations */ + struct snd_soc_ops *ops; }; /* SoC card */ struct snd_soc_card { - char *name; + const char *name; struct device *dev; + struct snd_card *snd_card; + struct module *owner; struct list_head list; + struct mutex mutex; - int instantiated; + int instantiated:1; int (*probe)(struct platform_device *pdev); int (*remove)(struct platform_device *pdev); @@ -568,28 +594,38 @@ struct snd_soc_card { /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; int num_links; + struct snd_soc_pcm_runtime *rtd; + int num_rtd; - struct snd_soc_device *socdev; - - struct snd_soc_codec *codec; - - struct snd_soc_platform *platform; - struct delayed_work delayed_work; struct work_struct deferred_resume_work; + + /* lists of probed devices belonging to this card */ + struct list_head codec_dev_list; + struct list_head platform_dev_list; + struct list_head dai_dev_list; }; -/* SoC Device - the audio subsystem */ -struct snd_soc_device { - struct device *dev; +/* SoC machine DAI configuration, glues a codec and cpu DAI together */ +struct snd_soc_pcm_runtime { + struct device dev; struct snd_soc_card *card; - struct snd_soc_codec_device *codec_dev; - void *codec_data; -}; + struct snd_soc_dai_link *dai_link; + + unsigned int complete:1; + unsigned int dev_registered:1; + + /* Symmetry data - only valid if symmetry is being enforced */ + unsigned int rate; + long pmdown_time; -/* runtime channel data */ -struct snd_soc_pcm_runtime { - struct snd_soc_dai_link *dai; - struct snd_soc_device *socdev; + /* runtime devices */ + struct snd_pcm *pcm; + struct snd_soc_codec *codec; + struct snd_soc_platform *platform; + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; + + struct delayed_work delayed_work; }; /* mixer control */ @@ -615,24 +651,48 @@ struct soc_enum { static inline unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg) { - return codec->read(codec, reg); + return codec->driver->read(codec, reg); } static inline unsigned int snd_soc_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { - return codec->write(codec, reg, val); + return codec->driver->write(codec, reg, val); } +/* device driver data */ + static inline void snd_soc_codec_set_drvdata(struct snd_soc_codec *codec, - void *data) + void *data) { - codec->drvdata = data; + dev_set_drvdata(codec->dev, data); } static inline void *snd_soc_codec_get_drvdata(struct snd_soc_codec *codec) { - return codec->drvdata; + return dev_get_drvdata(codec->dev); +} + +static inline void snd_soc_platform_set_drvdata(struct snd_soc_platform *platform, + void *data) +{ + dev_set_drvdata(platform->dev, data); +} + +static inline void *snd_soc_platform_get_drvdata(struct snd_soc_platform *platform) +{ + return dev_get_drvdata(platform->dev); +} + +static inline void snd_soc_pcm_set_drvdata(struct snd_soc_pcm_runtime *rtd, + void *data) +{ + dev_set_drvdata(&rtd->dev, data); +} + +static inline void *snd_soc_pcm_get_drvdata(struct snd_soc_pcm_runtime *rtd) +{ + return dev_get_drvdata(&rtd->dev); } #include <sound/soc-dai.h> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8b79d90..3deb226 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3,6 +3,8 @@ * * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. + * Copyright (C) 2010 Slimlogic Ltd. + * Copyright (C) 2010 Texas Instruments Inc. * * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> * with code, comments and ideas from :- -- 1.7.0.4

1 0

[alsa-devel] [PATCH 1/1] Daily snapshot not building on 2.6.32
by Brad Figg 24 Jun '10

24 Jun '10

I've had to apply the following patch to successfully build the daily snapshot on 2.6.32. Without this patch I get: CC [M] /karmic/home/work/alsa/alsa-driver/usb/urb.o /karmic/home/work/alsa/alsa-driver/usb/urb.c: In function ‘release_urb_ctx’: /karmic/home/work/alsa/alsa-driver/usb/urb.c:112: error: implicit declaration of function ‘usb_free_ coherent’ /karmic/home/work/alsa/alsa-driver/usb/urb.c: In function ‘snd_usb_init_substream_urbs’: /karmic/home/work/alsa/alsa-driver/usb/urb.c:327: error: implicit declaration of function ‘usb_alloc_coherent’ /karmic/home/work/alsa/alsa-driver/usb/urb.c:328: warning: assignment makes pointer from integer without a cast /karmic/home/work/alsa/alsa-driver/usb/urb.c:341: warning: assignment makes pointer from integer without a cast make[3]: *** [/karmic/home/work/alsa/alsa-driver/usb/urb.o] Error 1 Note: I would have preferred to provide this as a git patch but I don't know where the repository for the daily snapshot files is located. --- diff -ru a/usb/misc/Makefile b/usb/misc/Makefile --- a/usb/misc/Makefile 2010-03-10 01:12:19.000000000 -0800 +++ b/usb/misc/Makefile 2010-06-23 11:25:33.176133657 -0700 @@ -7,4 +7,8 @@ include $(SND_TOPDIR)/alsa-kernel/usb/misc/Makefile +EXTRA_CFLAGS += -I$(SND_TOPDIR)/usb + include $(SND_TOPDIR)/Rules.make + +ua101.c: usbaudio.inc $(SND_TOPDIR)/alsa-kernel/usb/misc/ua101.c diff -ru a/usb/misc/ua101.c b/usb/misc/ua101.c --- a/usb/misc/ua101.c 2010-03-10 01:12:19.000000000 -0800 +++ b/usb/misc/ua101.c 2010-06-23 11:25:53.054884851 -0700 @@ -1,3 +1,5 @@ +#define __NO_VERSION__ +#include "usbaudio.inc" #include "adriver.h" #include "../../alsa-kernel/usb/misc/ua101.c" EXPORT_NO_SYMBOLS; -- Brad Figg brad.figg(a)canonical.com http://www.canonical.com

2 12

[alsa-devel] USB midi support for KAOSS pad
by Pedro Ferreira 24 Jun '10

24 Jun '10

Hello all, I tried connecting my KAOSS PAD 3 to a linux laptop, and apparently ALSA MIDI detects the device. However, dmesg shows several messages, all similar to: {{{ [ 7088.722283] ALSA midi.c:1597: MIDIStreaming version 01.00 [ 7088.722290] ALSA midi.c:1632: EP 02: 1 jack(s) [ 7088.722295] ALSA midi.c:1647: EP 82: 1 jack(s) [ 7088.722315] ALSA midi.c:1567: created 1 output and 1 input ports [ 7088.806326] usb 2-1.1: reset full speed USB device using ehci_hcd and address 15 }}} If i open qjackctl, i can see the device being added and removed successively. Does anyone know what can be going wrong? I would be more than happy to contribute with usbmon logs, etc... or even try to research a bit if someone points me in the right direction. Thanks in advance, Pedro

2 6

[alsa-devel] DMA Physical Address through "snd_pcm_lib_malloc_pages"
by Adish Kuvelker 24 Jun '10

24 Jun '10

Hello, I am trying to allocate memory for DMA buffer through "snd_pcm_lib_malloc_pages" in hw_params callout function, with prior use of "snd_pcm_lib_preallocate_pages_for_all". The chapter on PCM Interface in "writing an ALSA driver" states that if the standard function "snd_pcm_lib_malloc_pages" is use for allocating the DMA buffers then the DMA fields of dma_addr, dma_area, dma_bytes are set by the ALSA middle layer. However my code walk through didn't point me to the area where this physical memory setting is done. My DMA channel is not serving a PCI device. I would be very grateful if any of you guys can point me to the code where this settings get done by calling "snd_pcm_lib_malloc_pages". Going through the code for a few devices registered under "pci" I found that the call to "snd_pcm_lib_malloc_pages" is followed by call to "snd_pcm_sgbuf_get_addr" to get the physical address. Also a call to "__pa" is done in the "parisc/harmony.c". Do I have to set the dma parameters in the runtime structure even after allocating memory using "snd_pcm_lib_malloc_pages"? Regards; Aadish

2 7

[alsa-devel] force snd-usb-audio to use "Vendor Specific Class"
by Alexey Fisher 24 Jun '10

24 Jun '10

Hallo all, is there any way to force snd-usb-audio to use some "Vendor Specific Class" device? i currently got two webcams from my friends to take a look on. They are cheep gspca based webcams with Mic. I wont to test if snd-usb-audio can use it if i force it. Devices it self do not provide any hints for the driver. lsusb -vd 093a:2468 | grep -i class bDeviceClass 255 Vendor Specific Class bDeviceSubClass 255 Vendor Specific Subclass bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 255 Vendor Specific Subclass bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 255 Vendor Specific Subclass bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 255 Vendor Specific Subclass bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 255 Vendor Specific Subclass bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 255 Vendor Specific Subclass bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 255 Vendor Specific Subclass bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 255 Vendor Specific Subclass bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 255 Vendor Specific Subclass bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 255 Vendor Specific Subclass regards, Alexey

3 4

[alsa-devel] [GIT PULL] ASoC updates for 2.6.36
by Liam Girdwood 24 Jun '10

24 Jun '10

The following changes since commit 66517915e0954ee027b889f452511945f7a9f3ec: Peter Huewe (1): ASoC: Fix I2C dependency for SND_FSI_AK4642 and SND_FSI_DA7210 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.36 Eric Bénard (1): codecs/tlv320aic23: fix bias management for suspend/resume Grazvydas Ignotas (1): ASoC: pandora: fix CLKX polarity Jarkko Nikula (2): ASoC: RX-51: Add Jack Function kcontrol ASoC: RX-51: Add basic jack detection sound/soc/codecs/tlv320aic23.c | 7 ++-- sound/soc/omap/omap3pandora.c | 36 ++++--------------- sound/soc/omap/rx51.c | 73 +++++++++++++++++++++++++++++++++++++++- 3 files changed, 84 insertions(+), 32 deletions(-)

2 1

Re: [alsa-devel] PROBLEM: SIS7019 stops recording after 42 min
by David Dillow 24 Jun '10

24 Jun '10

Not trimming for the benefit of alsa-devel, cc'd. Please trim replies. On Sat, 2010-06-19 at 16:13 +0200, Hans Schou wrote: > Hi > > I have a problem with recording sound when using the sound chip > SIS7019 with both kernel 2.6.26 and 2.6.34. After recording about 42 > minutes it kind a stops recording, more precisely it is taking a pause > of exactly 10 seconds between each reading. > > As recorder I have tried several programs and all of them fails after > 42 minutes. Some programs uses Alsa and some uses the old deprecated > method. In this example I have logged sox rec. > > Recording method in a script: > strace -tt -o strace.log rec -c 1 -r 44100 -2 sox.wav & > sleep 3000 > kill $? I think the answer is no, but to be sure -- are you talking directly to the hardware device, or are you going through pulseaudio? While rec is running, can you capture the configuration using head -1000 /proc/asound/card0/pcm0c/sub0/* Once you have the information, there is no need to run it all the way out. I think that's the right path, but I'm going from memory so you may have to look around under /proc/asound. I'm trying to get my MicroClient up and running again so I can try to reproduce this. This information will help me narrow my focus to the correct area of the driver. Can you try using arecord? You can use options to tell it how to configure the PCM. Especially interesting will be to configure 2 periods per buffer vs whatever rec uses. 2 periods per buffer uses the hardware interrupts to clock out periods, where as 3+ uses a more complex mechanism. You can also use -M to use mmap vs not. > Recording start 17:41:29 and undtil 18:23:38.977278 it writes about > 6-7 times per second to disk. Then it begin only writing every 10 > seconds. At this point it had written 27222 times to disk and the > amount of data written so far is 28684288 bytes (including > wav-header). > > WorkARound: When the recording stops, one can just exit 'rec' and > start it again and record another 42 minutes. Reboot is not necessary. > Apparently, the re-init of the sound chip fixes the problem. > > On other systems with nVidia CK804 or emu10k I do not have the same problem. > > I have looked at sound/pci/sis7019.c but I do not really know what I > should look for. None of the numbers in my observations is in the > power of 2 which I would expect it was. Like, I expected it stopped > recording after 64MB or so, but that is not the case. > > Linux version 2.6.34-norhtec-sis (root@norh) (gcc version 4.1.2 > 20061115 (prerelease) (Debian 4.1.1-21)) #7 Wed Jun 16 00:32:56 CEST > 2010 > (the kernel used was compiled on this system.) > > # scripts/ver_linux > Gnu C 4.1.2 > Gnu make 3.81 > binutils 2.17 > util-linux 2.12r > mount 2.12r > module-init-tools 3.3-pre2 > e2fsprogs 1.40-WIP > Linux C Library scripts/ver_linux: line 63: /proc/self/maps: No > such file or directory > Dynamic linker (ldd) 2.3.6 > Net-tools 1.60 > Console-tools 0.2.3 > Sh-utils 5.97 > udev 105 > > # lspci -vvv > 00:01.4 Multimedia audio controller: Silicon Integrated Systems [SiS] > SiS7019 Audio Accelerator > Subsystem: Silicon Integrated Systems [SiS] SiS7019 Audio > Accelerator > Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- > ParErr- Stepping- SERR- FastB2B- DisINTx- > Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium > >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- > Latency: 64 (500ns min, 6000ns max) > Interrupt: pin B routed to IRQ 10 > Region 0: I/O ports at dc00 [size=256] > Region 1: Memory at dfff8000 (32-bit, non-prefetchable) > [size=16K] > Capabilities: [60] Power Management version 2 > Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA > PME(D0+,D1+,D2+,D3hot+,D3cold+) > Status: D0 PME-Enable- DSel=0 DScale=0 PME- > Kernel driver in use: SiS7019 > Kernel modules: snd-sis7019 > > Here is a snip from the log file when it goes from reading 7 times per > second to every 10th second. Note two times "ioctl" within a second > and then it goes over to a delay with 10 seconds between each "ioctl": > > 18:23:38.611877 ioctl(4, 0x800c4151, 0xbf96b1d0) = 0 > 18:23:38.784635 gettimeofday({1276446218, 785207}, NULL) = 0 > 18:23:38.786609 write(2, "\rTime: 42:08.19 [00:00.00] of 00:"..., 78)= 78 > 18:23:38.789354 > write(3,"p\377q\377n\377k\377l\377p\377o\377o\377k\377n\377p\377q\377n\377o\377k\377l\377p"...,4096) > = 4096 > 18:23:38.791590 > write(3,"j\377l\377l\377j\377l\377k\377m\377k\377l\377n\377j\377l\377l\377n\377n\377g\377m"...,12288) > = 12288 > 18:23:38.794425 gettimeofday({1276446218, 795024}, NULL) = 0 > 18:23:38.795762 ioctl(4, 0x800c4151, 0xbf96b1d0) = 0 > 18:23:38.970451 gettimeofday({1276446218, 971018}, NULL) = 0 > 18:23:38.972293 write(2, "\rTime: 42:08.37 [00:00.00] of 00:"..., 78)= 78 > 18:23:38.975134 > write(3,"k\377m\377o\377k\377i\377j\377m\377o\377o\377m\377k\377m\377o\377k\377p\377n\377o"...,4096) > = 4096 > 18:23:38.977278 > write(3,"m\377j\377p\377n\377n\377r\377m\377o\377o\377o\377l\377h\377l\377l\377m\377o\377o"...,12288) > = 12288 > 18:23:38.980176 gettimeofday({1276446218, 980773}, NULL) = 0 > 18:23:38.981516 ioctl(4, 0x800c4151, 0xbf96b1d0) = 0 > 18:23:48.983347 ioctl(4, 0x800c4151, 0xbf96b1d0) = 0 > 18:23:48.985710 gettimeofday({1276446228, 986317}, NULL) = 0 > 18:23:48.987784 write(2, "\rTime: 42:08.56 [00:00.00] of 00:"..., 78) = 78 > 18:23:48.990733 > write(3,"j\377k\377l\377l\377l\377n\377l\377q\377q\377j\377l\377m\377m\377p\377o\377o\377q"...,4096) > = 4096 > 18:23:48.992814 > write(3,"p\377m\377k\377s\377r\377m\377o\377m\377k\377k\377j\377j\377l\377p\377o\377r\377l"...,12288) > = 12288 > 18:23:48.995694 gettimeofday({1276446228, 996297}, NULL) = 0 > 18:23:48.997046 ioctl(4, 0x800c4151, 0xbf96b1d0) = 0 > 18:23:48.999574 gettimeofday({1276446229, 186}, NULL) = 0 > 18:23:49.003421 > write(3,"l\377l\377m\377o\377q\377q\377n\377m\377q\377r\377s\377p\377n\377t\377r\377m\377n"...,4096) > = 4096 > 18:23:49.005517 > write(3,"o\377m\377l\377p\377m\377p\377o\377m\377p\377p\377q\377o\377j\377m\377r\377m\377r"...,12288) > = 12288 > 18:23:49.008337 gettimeofday({1276446229, 8945}, NULL) = 0 > 18:23:49.009699 ioctl(4, 0x800c4151, 0xbf96b1d0) = 0 > 18:23:59.007878 ioctl(4, 0x800c4151, 0xbf96b1d0) = 0 > > Version of sox: SoX v14.0.1 > > Full log file is 780KB compressed: > http://schou.dk/linux/sis7019/strace.sis7019.sox-rec.2010-06-13.log.bz2 > > Hardware system used is a Norhtec MicroClient Jr with SIS550 CPU: > http://www.norhtec.com/products/mcjrmx/index.html > > best regards/hans

3 17

Re: [alsa-devel] 2.6.35-rc3 E-MU 0404 USB problem
by Clemens Ladisch 24 Jun '10

24 Jun '10

>>> Sid Boyce wrote: >>>> Transmit fails, What exactly fails? >>>> dmesg output with 48K sample rate set is:- >>>> ALSA sound/usb/clock.c:214: 6:2:3: endpoint lacks sample rate attribute bit, cannot set. There are many devices that have only one sample rate or that have one sample rate per alternate setting. This doesn't deserve a message, much less a warning. Regards, Clemens

2 1

[alsa-devel] snd_pcm_hardware structure
by Adish Kuvelker 24 Jun '10

24 Jun '10

Hello, I am developing a ALSA driver for the AtlasIII AC'97 Controller. The AC'97 Controller FIFO is arranged as 16 32-bit entries. My PCM data would be 16-bit samples and the driver will cater to 2-front channel audio. I have configured the snd_pcm_hardware members as follows: .formats = SNDDRV_PCM_FMTBIT_S16_LE .period_bytes_max = 4, .period_bytes_min = 64, As I would need a minimum of 4-byte sample to be transferred and the "period_bytes_max" value has been set considering the maximum size of the FIFO in bytes. Is my understanding correct in doing these settings? Secondly, I am having slight problems in understanding the "snd_pcm_hardware" members periods_min and periods_max. Can anyone throw some more light on this in addition to what is explained in the "Writing an Audio Driver"? Regards; Aadish

2 7

[alsa-devel] newie, adding a file to the project
by javi perez 23 Jun '10

23 Jun '10

Hello, I am trying to add a file (aplay_miversion.c) to the project to make tests. I have modified the Makefile.am file of the aplay directory in the project. I have run ./configure and make and I don't get the file compiled. How shold I change the lines? LIBRT = @LIBRT@ INCLUDES = -I$(top_srcdir)/include LDADD = $(LIBINTL) $(LIBRT) # debug flags #LDFLAGS = -static #LDADD += -ldl bin_PROGRAMS = aplay aplay_miversion aplay_SOURCES = aplay.c aplay_miversion_SOURCES = aplay_miversion.c man_MANS = aplay.1 arecord.1 noinst_HEADERS = formats.h EXTRA_DIST = aplay.1 arecord.1 EXTRA_CLEAN = arecord arecord: aplay rm -f arecord $(LN_S) $< $@ arecord.1: aplay.1 rm -f arecord.1 $(LN_S) $< $@ install-exec-hook: arecord rm -f $(DESTDIR)$(bindir)/arecord (cd $(DESTDIR)$(bindir) && $(LN_S) aplay arecord) install-data-hook: rm -f $(DESTDIR)$(mandir)/man1/arecord.1 (cd $(DESTDIR)$(mandir)/man1 && $(LN_S) aplay.1 arecord.1)

2 1

[alsa-devel] IEC switch issues
by pl bossart 23 Jun '10

23 Jun '10

If I play my AC3 data on my HDAudio/SPDIF output using the hw:0,1 device, I can use alsamixer/amixer to mute/unmute. [ume@plb PassThough]$ amixer cset numid=12 on numid=12,iface=MIXER,name='IEC958 Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on [ume@plb PassThough]$ amixer cset numid=12 off numid=12,iface=MIXER,name='IEC958 Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off Now if I use the iec958: plugin, I can't control the mute switch any longer: [ume@plb PassThough]$ amixer cset numid=12 off amixer: Control default element write error: Operation not permitted [ume@plb PassThough]$ amixer cset numid=12 on amixer: Control default element write error: Operation not permitted What could possibly cause this issue? the IEC plugin relies on the device 1, am I missing something here? Thanks for your help - Pierre

4 6

[alsa-devel] AndreaElectronics PureAudio USB-SA
by rmfc 23 Jun '10

23 Jun '10

Hello, I'm looking for a sound card that can be used to speech recognition in noisy environments. The best device that I found so far is AndreaElectronics PureAudio USB-SA but unfortunately it isn't properly supported in Linux. The device is recognized as an USB Audio device (module used is snd_usb_audio) and I can record audio in Audacity. The problem is that none of advanced DSP features is enabled. I tried then Asus Xonar U1 but the problem is the same, no configuration is done in devices, so they record audio without quality to be used in a speech recognizer (not enough energy, audio DC components when software gain is applied in Sound Preferences...). Is there any way known way to configure a USB Audio device? If I have to implement some code, where should I start? I send you output of "lsusb" of AndreaElectronics USB-SA sound card. ------------------------------------------------------------ Bus 003 Device 002: ID 08a8:000c Andrea Electronics Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x08a8 Andrea Electronics idProduct 0x000c bcdDevice 1.00 iManufacturer 3 Andrea Electronics iProduct 1 Andrea PureAudio USB-SA iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 293 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 128 bInCollection 2 baInterfaceNr( 0) 1 baInterfaceNr( 1) 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0603 Line Connector bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 6 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 9 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 8 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 5 (SELECTOR_UNIT) bUnitID 8 bNrInPins 3 baSource( 0) 10 baSource( 1) 11 baSource( 2) 12 iSelector 0 AudioControl Interface Descriptor: bLength 13 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 9 bSourceID 1 bControlSize 2 bmaControls( 0) 0x01 bmaControls( 0) 0x02 Mute Loudness bmaControls( 1) 0x02 bmaControls( 1) 0x00 Volume bmaControls( 2) 0x02 bmaControls( 2) 0x00 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 10 bSourceID 2 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 11 bSourceID 3 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 12 bSourceID 4 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x00 bmaControls( 2) 0x00 iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 4 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 44100 tSamFreq[ 3] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 1 Milliseconds wLockDelay 1 Milliseconds Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 4 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 16000 tSamFreq[ 2] 44100 tSamFreq[ 3] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.00 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 50 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x87 EP 7 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 1 Device Status: 0x0000 (Bus Powered) ------------------------------------------------------------ Thanks for any help that you can give me. Renato

1 0

[alsa-devel] Pause functionality
by dv 23 Jun '10

23 Jun '10

Hi, it is stated in the documentation that I need to check with snd_pcm_hw_params_can_pause() to see if the device can pause or not. If it can, I can use snd_pcm_pause() to pause/resume. But what if snd_pcm_hw_params_can_pause() tells me it can't pause? What does one usually do then? Continue writing to pcm, with a buffer containing zeroed samples?

1 0

[alsa-devel] usb-audio: some cleanup patches for 2.6.36
by Daniel Mack 23 Jun '10

23 Jun '10

Here are five patches that mostly do some basic cleanups. They're unurgent and should be queued for 2.6.36. I'm still pondering about the clock topology and how to fully support all corner cases. That might take awhile to sort out. Daniel [PATCH 1/5] ALSA: usb-audio: clean up includes in clock.c [PATCH 2/5] ALSA: usb-audio: unify UAC macros and struct names [PATCH 3/5] ALSA: usb-midi: whitespace fixes [PATCH 4/5] ALSA: usb-audio: move and add some comments [PATCH 5/5] ALSA: usb-audio: simplify control interface access

2 9

[alsa-devel] [PATCH] Add volume resolution quirk for some Logitech webcams.
by Alexey Fisher 23 Jun '10

23 Jun '10

Some programs like Skype trying to set capture volume automatically. Normally it will tray, carefully step by step lover or higher, set the volume. In real word it work not really well, because devises and vendors lie about real audio settings. For example most Logitech webcams have 6400 or 3500 steps for capture volume. They do not tell that actual resolution is 384. So we have only 7 or 18 real steps. In this patch I set real resolution only for tested devices. Signed-off-by: Alexey Fisher <bug-track(a)fisher-privat.net> --- sound/usb/mixer.c | 13 +++++++++++++ 1 files changed, 13 insertions(+), 0 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 6939d0f..736d134 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1107,6 +1107,19 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, } break; + case USB_ID(0x046d, 0x0809): + case USB_ID(0x046d, 0x0991): + /* Most audio usb devices lie about volume resolution. + * Most Logitech webcams have res = 384. + * Proboly there is some logitech magic behind this number --fishor + */ + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + snd_printk(KERN_INFO + "set resolution quirk: cval->res = 384\n"); + cval->res = 384; + } + break; + } snd_printdd(KERN_INFO "[%d] FU [%s] ch = %d, val = %d/%d/%d\n", -- 1.7.0.4

2 1

[alsa-devel] [PATCH] Add volume resolution quirk for some logitech webcams
by Alexey Fisher 23 Jun '10

23 Jun '10

Some programs like Skype trying to set capture volume automatically. Normally it will tray carefully, step by step lover or higher the volume. In real word it work not really well, because devises and vendors lie about real audio settings. For example most Logitech webcams have 6400 or 3500 steps for capture volume. They do not tell that actual resolution is 384. So we have only 7 or 18 real steps. In this patch I set real resolution only for tested devices. Signed-off-by: Alexey Fisher <bug-track(a)fisher-privat.net> --- sound/usb/mixer.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 6939d0f..fd9f377 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1107,6 +1107,18 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, } break; + case USB_ID(0x046d, 0x0809): + case USB_ID(0x046d, 0x0991): + /* most audio usb divise lighe about volume resolution. + * most of them have res = 384. Proboly there is some magick behind + * this number. */ + if (!strcmp(kctl->id.name, "Mic Capture Volume")) { + snd_printk(KERN_INFO + "set volume quirk: cval->res = 384\n"); + cval->res = 384; + } + break; + } snd_printdd(KERN_INFO "[%d] FU [%s] ch = %d, val = %d/%d/%d\n", -- 1.7.0.4

2 2

[alsa-devel] [PATCH 1/2] ASoC: RX-51: Add Jack Function kcontrol
by Jarkko Nikula 23 Jun '10

23 Jun '10

Nokia RX-51/N900 has multifunction 4-pole audio-video jack that can be used as headphone, headset or audio-video connector. This patch implements the control 'Jack Function' which is used to select the desired function. At the moment only TV-out without audio is supported. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/soc/omap/rx51.c | 44 ++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 44 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 47d831e..1a2de34 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -37,14 +37,21 @@ #include "omap-pcm.h" #include "../codecs/tlv320aic3x.h" +#define RX51_TVOUT_SEL_GPIO 40 /* * REVISIT: TWL4030 GPIO base in RX-51. Now statically defined to 192. This * gpio is reserved in arch/arm/mach-omap2/board-rx51-peripherals.c */ #define RX51_SPEAKER_AMP_TWL_GPIO (192 + 7) +enum { + RX51_JACK_DISABLED, + RX51_JACK_TVOUT, /* tv-out */ +}; + static int rx51_spk_func; static int rx51_dmic_func; +static int rx51_jack_func; static void rx51_ext_control(struct snd_soc_codec *codec) { @@ -57,6 +64,9 @@ static void rx51_ext_control(struct snd_soc_codec *codec) else snd_soc_dapm_disable_pin(codec, "DMic"); + gpio_set_value(RX51_TVOUT_SEL_GPIO, + rx51_jack_func == RX51_JACK_TVOUT); + snd_soc_dapm_sync(codec); } @@ -162,6 +172,28 @@ static int rx51_set_input(struct snd_kcontrol *kcontrol, return 1; } +static int rx51_get_jack(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = rx51_jack_func; + + return 0; +} + +static int rx51_set_jack(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + if (rx51_jack_func == ucontrol->value.integer.value[0]) + return 0; + + rx51_jack_func = ucontrol->value.integer.value[0]; + rx51_ext_control(codec); + + return 1; +} + static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event), SND_SOC_DAPM_MIC("DMic", NULL), @@ -177,10 +209,12 @@ static const struct snd_soc_dapm_route audio_map[] = { static const char *spk_function[] = {"Off", "On"}; static const char *input_function[] = {"ADC", "Digital Mic"}; +static const char *jack_function[] = {"Off", "TV-OUT"}; static const struct soc_enum rx51_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function), SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function), + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function), }; static const struct snd_kcontrol_new aic34_rx51_controls[] = { @@ -188,6 +222,8 @@ static const struct snd_kcontrol_new aic34_rx51_controls[] = { rx51_get_spk, rx51_set_spk), SOC_ENUM_EXT("Input Select", rx51_enum[1], rx51_get_input, rx51_set_input), + SOC_ENUM_EXT("Jack Function", rx51_enum[2], + rx51_get_jack, rx51_set_jack), }; static int rx51_aic34_init(struct snd_soc_codec *codec) @@ -259,6 +295,11 @@ static int __init rx51_soc_init(void) if (!machine_is_nokia_rx51()) return -ENODEV; + err = gpio_request(RX51_TVOUT_SEL_GPIO, "tvout_sel"); + if (err) + goto err_gpio_tvout_sel; + gpio_direction_output(RX51_TVOUT_SEL_GPIO, 0); + rx51_snd_device = platform_device_alloc("soc-audio", -1); if (!rx51_snd_device) { err = -ENOMEM; @@ -277,6 +318,8 @@ static int __init rx51_soc_init(void) err2: platform_device_put(rx51_snd_device); err1: + gpio_free(RX51_TVOUT_SEL_GPIO); +err_gpio_tvout_sel: return err; } @@ -284,6 +327,7 @@ err1: static void __exit rx51_soc_exit(void) { platform_device_unregister(rx51_snd_device); + gpio_free(RX51_TVOUT_SEL_GPIO); } module_init(rx51_soc_init); -- 1.7.1

3 3

[alsa-devel] [PATCH] codecs/tlv320aic23: fix suspend/resume
by Eric Bénard 23 Jun '10

23 Jun '10

The PWR register actually can't be restored at resume because its cached value was set to 0xFFFF when suspending and thus the initial state won't be restored. Thus, sound doesn't work after resume (tested on an i.MX27). To workaround this, this patch saves the PWR value before turning the codec's power off and store this value in the cached registers. Signed-off-by: Eric Bénard <eric(a)eukrea.com> Cc: broonie(a)opensource.wolfsonmicro.com Cc: anuj.aggarwal(a)ti.com Cc: lrg(a)slimlogic.co.uk --- sound/soc/codecs/tlv320aic23.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index b0bae35..75e320b 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -614,9 +614,15 @@ static int tlv320aic23_suspend(struct platform_device *pdev, { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; + u16 tmp; tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); + /* backup PWR register */ + tmp = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR); + /* turn off the codec */ tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); + /* restore PWR register in the cache */ + tlv320aic23_write_reg_cache(codec, TLV320AIC23_PWR, tmp); return 0; } @@ -633,8 +639,6 @@ static int tlv320aic23_resume(struct platform_device *pdev) tlv320aic23_write(codec, reg, val); } - tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return 0; } -- 1.6.3.3

3 6

[alsa-devel] imx-ssi.c & SND_SOC_DAIFMT_I2S
by Eric Bénard 23 Jun '10

23 Jun '10

Hi Sascha, while trying to get a TI TLV320AIC23B codec configured as I2S master to work with an i.MX27 using the latest git kernel, I noticed there may be a problem in the slave I2S setting in imx-ssi.c Please find two captures of the I2S bus signals : - http://eukrea.com/tmp/initial_i2s.png : this is not an I2S signal, both left & right are sent while Frame Sync is high (so I only get mono output) - http://eukrea.com/tmp/i2s_slave.png : when I set I2S_MODE to 0x2 in SCR register (scr |= SSI_SCR_NET | 0x40; at line 101 of imx-ssi.c), thus setting the SSI in "I2S slave mode", I get a real I2S signal as shown on this capture. I understand the actual I2S configuration may work with the Freescale's PMIC but this chipset may not be fully I2S compliant and I think that the actual setting prevent real I2S codecs from being used with the i.MX. Am I wrong here ? Shouldn't we use the I2S slave mode of the SSI instead of what is done actually ? Thanks and best regards, Eric

4 7

[alsa-devel] ASoC updates for 2.6.36
by Mark Brown 23 Jun '10

23 Jun '10

The following changes since commit 66517915e0954ee027b889f452511945f7a9f3ec: ASoC: Fix I2C dependency for SND_FSI_AK4642 and SND_FSI_DA7210 (2010-06-16 16:34:17 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Eric Bénard (1): ASoC: eukrea-tlv320: add support for our i.MX25 board Lars-Peter Clausen (3): ASoC: Add JZ4740 ASoC support ASoC: Add JZ4740 codec driver ASoC: JZ4740: Add qi_lb60 board driver Mark Brown (1): ASoC: Fix sorting of DA7210 entries in Kconfig Stuart Longland (1): ASoC: Fix overflow bug in SOC_DOUBLE_R_SX_TLV sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 12 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/jz4740.c | 511 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/jz4740.h | 20 ++ sound/soc/imx/Kconfig | 2 +- sound/soc/imx/eukrea-tlv320.c | 3 +- sound/soc/jz4740/Kconfig | 23 ++ sound/soc/jz4740/Makefile | 13 + sound/soc/jz4740/jz4740-i2s.c | 540 +++++++++++++++++++++++++++++++++++++++++ sound/soc/jz4740/jz4740-i2s.h | 18 ++ sound/soc/jz4740/jz4740-pcm.c | 373 ++++++++++++++++++++++++++++ sound/soc/jz4740/jz4740-pcm.h | 22 ++ sound/soc/jz4740/qi_lb60.c | 166 +++++++++++++ sound/soc/soc-core.c | 4 +- 16 files changed, 1703 insertions(+), 8 deletions(-) create mode 100644 sound/soc/codecs/jz4740.c create mode 100644 sound/soc/codecs/jz4740.h create mode 100644 sound/soc/jz4740/Kconfig create mode 100644 sound/soc/jz4740/Makefile create mode 100644 sound/soc/jz4740/jz4740-i2s.c create mode 100644 sound/soc/jz4740/jz4740-i2s.h create mode 100644 sound/soc/jz4740/jz4740-pcm.c create mode 100644 sound/soc/jz4740/jz4740-pcm.h create mode 100644 sound/soc/jz4740/qi_lb60.c

2 1

[alsa-devel] Problems building the daily snapshot...
by Brad Figg 23 Jun '10

23 Jun '10

I had been building the daily snapshot from: http://kernel.org/pub/linux/kernel/people/tiwai/snapshot/ Sometime recently this has stopped working. I am doing: ./configure --with-debug=verbose --with-kernel=/lib/modules/2.6.32-23-generic/build make The make fails with: make[3]: Leaving directory `/karmic/home/work/alsa/alsa-driver/usb/usx2y' make[2]: Leaving directory `/karmic/home/work/alsa/alsa-driver/usb' make[2]: Entering directory `/karmic/home/work/alsa/alsa-driver/pcmcia' make[3]: Entering directory `/karmic/home/work/alsa/alsa-driver/pcmcia/pdaudiocf' make[3]: Leaving directory `/karmic/home/work/alsa/alsa-driver/pcmcia/pdaudiocf' make[3]: Entering directory `/karmic/home/work/alsa/alsa-driver/pcmcia/vx' make[3]: Leaving directory `/karmic/home/work/alsa/alsa-driver/pcmcia/vx' make[2]: Leaving directory `/karmic/home/work/alsa/alsa-driver/pcmcia' make[2]: Entering directory `/karmic/home/work/alsa/alsa-driver/misc' make[2]: Leaving directory `/karmic/home/work/alsa/alsa-driver/misc' make[1]: Leaving directory `/karmic/home/work/alsa/alsa-driver' make -C SUBDIRS=/karmic/home/work/alsa/alsa-driver CPP="gcc -E" CC="gcc" modules make: *** SUBDIRS=/karmic/home/work/alsa/alsa-driver: No such file or directory. Stop. make: *** [compile] Error 2 (The full build log can be found at: http://kernel.ubuntu.com/~bradf/build.log) I would have expected this make invocation to have been: make SUBDIRS=/karmic/home/work/alsa/alsa-driver CPP="gcc -E" CC="gcc" -C modules However, I believe this has worked this way for quite a while so that is probably not the issue. I'd be interested in looking into this further but I don't understand how the daily snapshot is created nor where the build infrastructure files (configure, et.) come from. Brad -- Brad Figg brad.figg(a)canonical.com http://www.canonical.com

1 1

[alsa-devel] [PATCH v2 00/26] Add support for the Ingenic JZ4740 System-on-a-Chip
by Lars-Peter Clausen 23 Jun '10

23 Jun '10

Foreword: Ralf suggested that it might be a good idea in order to allow for reasonable testing and to avoid build failures due to two-way dependencies in different parts of the kernel, that he applies all the patches once they have been acked by their respective maintainers, feeds them into -next and eventually sends the whole series to Linus. One exception will be the ASoC patches which will, due to major changes in the ASoC subsystem, go through the ASoC tree. So if you are a maintainer for one of the subsystem touched by this series and would rather see the patch going through your tree (given the patch is ok) please tell. This patch series adds support for the Ingenic JZ4740 System-on-a-Chip. The JZ4740 has a mostly MIPS32 compatible core (no on cpu timers) and many on chip peripherals like RTC, NAND, MMC, OHCI, UDC, ADC, I2C, SPI, AC97, I2S, I2S Codec, UART and LCD controller. The JZ4740 is mostly used in eBooks, PMPs and hand-held consoles. This series contains patches for the Qi Ben NanoNote clamshell device as the inital supported device. Changes since v1: There have been some minor changes since v1, mostly code cleanup and some functional changes. One bigger change is that there is now a MFD driver for the ADC core which does IRQ demultiplexing for the ADC unit and synchronizes access to shared registers between the different users of the ADC core. The patch adding a defconfig for the Qi LB60 has been dropped. A detailed list of changes is present in each patch. - Lars Cc: Alessandro Zummo <a.zummo(a)towertech.it> Cc: Andrew Morton <akpm(a)linux-foundation.org> Cc: Anton Vorontsov <cbouatmailru(a)gmail.com> Cc: David Brownell <dbrownell(a)users.sourceforge.net> Cc: David Woodhouse <dwmw2(a)infradead.org> Cc: Greg Kroah-Hartman <gregkh(a)suse.de> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Paul Gortmaker <p_gortmaker(a)yahoo.com> Cc: Samuel Ortiz <sameo(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-fbdev(a)vger.kernel.org Cc: linux-mmc(a)vger.kernel.org Cc: linux-mtd(a)lists.infradead.org Cc: linux-usb(a)vger.kernel.org Cc: lm-sensors(a)lm-sensors.org Cc: rtc-linux(a)googlegroups.com Lars-Peter Clausen (26): MIPS: Add base support for Ingenic JZ4740 System-on-a-Chip MIPS: jz4740: Add IRQ handler code MIPS: JZ4740: Add clock API support. MIPS: JZ4740: Add timer support MIPS: JZ4740: Add clocksource/clockevent support. MIPS: JZ4740: Add power-management and system reset support MIPS: JZ4740: Add setup code MIPS: JZ4740: Add gpio support MIPS: JZ4740: Add DMA support. MIPS: JZ4740: Add PWM support MIPS: JZ4740: Add serial support MIPS: JZ4740: Add prom support MIPS: JZ4740: Add platform devices MIPS: JZ4740: Add Kbuild files RTC: Add JZ4740 RTC driver fbdev: Add JZ4740 framebuffer driver MTD: Nand: Add JZ4740 NAND driver MMC: Add JZ4740 mmc driver USB: Add JZ4740 ohci support alsa: ASoC: Add JZ4740 codec driver alsa: ASoC: Add JZ4740 ASoC support MFD: Add JZ4740 ADC driver hwmon: Add JZ4740 ADC driver power: Add JZ4740 battery driver. MIPS: JZ4740: Add qi_lb60 board support alsa: ASoC: JZ4740: Add qi_lb60 board driver arch/mips/Kbuild.platforms | 1 + arch/mips/Kconfig | 13 + arch/mips/include/asm/bootinfo.h | 6 + arch/mips/include/asm/cpu.h | 9 +- arch/mips/include/asm/mach-jz4740/base.h | 26 + arch/mips/include/asm/mach-jz4740/clock.h | 28 + arch/mips/include/asm/mach-jz4740/dma.h | 90 +++ arch/mips/include/asm/mach-jz4740/gpio.h | 398 +++++++++++ arch/mips/include/asm/mach-jz4740/irq.h | 57 ++ arch/mips/include/asm/mach-jz4740/platform.h | 36 + arch/mips/include/asm/mach-jz4740/timer.h | 22 + arch/mips/include/asm/mach-jz4740/war.h | 25 + arch/mips/jz4740/Kconfig | 12 + arch/mips/jz4740/Makefile | 20 + arch/mips/jz4740/Platform | 5 + arch/mips/jz4740/board-qi_lb60.c | 483 +++++++++++++ arch/mips/jz4740/clock-debugfs.c | 109 +++ arch/mips/jz4740/clock.c | 920 ++++++++++++++++++++++++ arch/mips/jz4740/clock.h | 76 ++ arch/mips/jz4740/dma.c | 289 ++++++++ arch/mips/jz4740/gpio.c | 601 ++++++++++++++++ arch/mips/jz4740/irq.c | 169 +++++ arch/mips/jz4740/irq.h | 21 + arch/mips/jz4740/platform.c | 284 ++++++++ arch/mips/jz4740/pm.c | 56 ++ arch/mips/jz4740/prom.c | 68 ++ arch/mips/jz4740/pwm.c | 169 +++++ arch/mips/jz4740/reset.c | 79 ++ arch/mips/jz4740/reset.h | 7 + arch/mips/jz4740/serial.c | 33 + arch/mips/jz4740/serial.h | 21 + arch/mips/jz4740/setup.c | 29 + arch/mips/jz4740/time.c | 144 ++++ arch/mips/jz4740/timer.c | 48 ++ arch/mips/jz4740/timer.h | 136 ++++ arch/mips/kernel/cpu-probe.c | 20 + arch/mips/mm/tlbex.c | 5 + drivers/hwmon/Kconfig | 11 + drivers/hwmon/Makefile | 1 + drivers/hwmon/jz4740-hwmon.c | 206 ++++++ drivers/mfd/Kconfig | 8 + drivers/mfd/Makefile | 1 + drivers/mfd/jz4740-adc.c | 392 ++++++++++ drivers/mmc/host/Kconfig | 8 + drivers/mmc/host/Makefile | 1 + drivers/mmc/host/jz4740_mmc.c | 993 ++++++++++++++++++++++++++ drivers/mtd/nand/Kconfig | 6 + drivers/mtd/nand/Makefile | 1 + drivers/mtd/nand/jz4740_nand.c | 474 ++++++++++++ drivers/power/Kconfig | 11 + drivers/power/Makefile | 1 + drivers/power/jz4740-battery.c | 445 ++++++++++++ drivers/rtc/Kconfig | 11 + drivers/rtc/Makefile | 1 + drivers/rtc/rtc-jz4740.c | 341 +++++++++ drivers/usb/Kconfig | 1 + drivers/usb/host/ohci-hcd.c | 5 + drivers/usb/host/ohci-jz4740.c | 276 +++++++ drivers/video/Kconfig | 9 + drivers/video/Makefile | 1 + drivers/video/jz4740_fb.c | 817 +++++++++++++++++++++ include/linux/jz4740-adc.h | 32 + include/linux/jz4740_fb.h | 58 ++ include/linux/mmc/jz4740_mmc.h | 15 + include/linux/mtd/jz4740_nand.h | 34 + include/linux/power/jz4740-battery.h | 24 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/jz4740-codec.c | 514 +++++++++++++ sound/soc/codecs/jz4740-codec.h | 20 + sound/soc/jz4740/Kconfig | 23 + sound/soc/jz4740/Makefile | 13 + sound/soc/jz4740/jz4740-i2s.c | 540 ++++++++++++++ sound/soc/jz4740/jz4740-i2s.h | 18 + sound/soc/jz4740/jz4740-pcm.c | 373 ++++++++++ sound/soc/jz4740/jz4740-pcm.h | 22 + sound/soc/jz4740/qi_lb60.c | 167 +++++ 79 files changed, 10396 insertions(+), 1 deletions(-) create mode 100644 arch/mips/include/asm/mach-jz4740/base.h create mode 100644 arch/mips/include/asm/mach-jz4740/clock.h create mode 100644 arch/mips/include/asm/mach-jz4740/dma.h create mode 100644 arch/mips/include/asm/mach-jz4740/gpio.h create mode 100644 arch/mips/include/asm/mach-jz4740/irq.h create mode 100644 arch/mips/include/asm/mach-jz4740/platform.h create mode 100644 arch/mips/include/asm/mach-jz4740/timer.h create mode 100644 arch/mips/include/asm/mach-jz4740/war.h create mode 100644 arch/mips/jz4740/Kconfig create mode 100644 arch/mips/jz4740/Makefile create mode 100644 arch/mips/jz4740/Platform create mode 100644 arch/mips/jz4740/board-qi_lb60.c create mode 100644 arch/mips/jz4740/clock-debugfs.c create mode 100644 arch/mips/jz4740/clock.c create mode 100644 arch/mips/jz4740/clock.h create mode 100644 arch/mips/jz4740/dma.c create mode 100644 arch/mips/jz4740/gpio.c create mode 100644 arch/mips/jz4740/irq.c create mode 100644 arch/mips/jz4740/irq.h create mode 100644 arch/mips/jz4740/platform.c create mode 100644 arch/mips/jz4740/pm.c create mode 100644 arch/mips/jz4740/prom.c create mode 100644 arch/mips/jz4740/pwm.c create mode 100644 arch/mips/jz4740/reset.c create mode 100644 arch/mips/jz4740/reset.h create mode 100644 arch/mips/jz4740/serial.c create mode 100644 arch/mips/jz4740/serial.h create mode 100644 arch/mips/jz4740/setup.c create mode 100644 arch/mips/jz4740/time.c create mode 100644 arch/mips/jz4740/timer.c create mode 100644 arch/mips/jz4740/timer.h create mode 100644 drivers/hwmon/jz4740-hwmon.c create mode 100644 drivers/mfd/jz4740-adc.c create mode 100644 drivers/mmc/host/jz4740_mmc.c create mode 100644 drivers/mtd/nand/jz4740_nand.c create mode 100644 drivers/power/jz4740-battery.c create mode 100644 drivers/rtc/rtc-jz4740.c create mode 100644 drivers/usb/host/ohci-jz4740.c create mode 100644 drivers/video/jz4740_fb.c create mode 100644 include/linux/jz4740-adc.h create mode 100644 include/linux/jz4740_fb.h create mode 100644 include/linux/mmc/jz4740_mmc.h create mode 100644 include/linux/mtd/jz4740_nand.h create mode 100644 include/linux/power/jz4740-battery.h create mode 100644 sound/soc/codecs/jz4740-codec.c create mode 100644 sound/soc/codecs/jz4740-codec.h create mode 100644 sound/soc/jz4740/Kconfig create mode 100644 sound/soc/jz4740/Makefile create mode 100644 sound/soc/jz4740/jz4740-i2s.c create mode 100644 sound/soc/jz4740/jz4740-i2s.h create mode 100644 sound/soc/jz4740/jz4740-pcm.c create mode 100644 sound/soc/jz4740/jz4740-pcm.h create mode 100644 sound/soc/jz4740/qi_lb60.c

5 12

[alsa-devel] [PATCH] ASoC: Add new TI TLV320AIC3204 CODEC driver
by Stuart Longland 22 Jun '10

22 Jun '10

The TLV320AIC3204 is a low-power stereo audio CODEC capable of sample rates of up to 192kHz. This driver implements basic functionality, using I²C for the control channel. The audio interface supports many data bus formats, including I²S master/slave, DSP, left/right justified and TDM. What works: - Playback at various bitrates up to 96kHz - Recording at various bitrates up to 96kHz - Mixer interface - PLL generation of CODEC clocks from MCLK What could work better: - DAPM What isn't tested: - Audio interfaces other than I²S - PLL with clocks other than ~12MHz - Mono recording/playback - 192kHz recording/playback - Suspend & Resume What isn't implemented: - SPI interface support - PLL without fractional divider (would allow <10MHz clocks) - Clock sources other than MCLK - Adaptive filtering - AGC - Headset detection, JACK framework TODO list: - Clean up the register cache ... this is in the early planning phase Signed-off-by: Stuart Longland <redhatter(a)gentoo.org> --- include/sound/tlv320aic3204.h | 64 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic3204.c | 4197 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic3204.h | 1350 ++++++++++++ 5 files changed, 5618 insertions(+), 0 deletions(-) create mode 100644 include/sound/tlv320aic3204.h create mode 100644 sound/soc/codecs/tlv320aic3204.c create mode 100644 sound/soc/codecs/tlv320aic3204.h diff --git a/include/sound/tlv320aic3204.h b/include/sound/tlv320aic3204.h new file mode 100644 index 0000000..491cd22 --- /dev/null +++ b/include/sound/tlv320aic3204.h @@ -0,0 +1,64 @@ +/* + * Platform data for Texas Instruments TLV320AIC3204 codec + * + * Copyright (C) 2010 Jacques Electronics Pty Ltd + * Author: Stuart Longland <redhatter(a)gentoo.org> + * + * Based on tlv320aic3x.h: + * Author: Jarkko Nikula <jhnikula(a)gmail.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ +#ifndef __TLV320AIC3204_H__ +#define __TLV320AIC3204_H__ + +/* + * Analogue Input Quick-Charge Power Up Delay + */ +enum { + /* Power up time: Default Value (whatever that is) */ + AIC3204_PDATA_AIN_PWRUP_DEFAULT = (0x00), + /* Power up time: 3.1ms */ + AIC3204_PDATA_AIN_PWRUP_3M1S = (0x31), + /* Power up time: 6.4ms */ + AIC3204_PDATA_AIN_PWRUP_6M4S = (0x32), + /* Power up time: 1.6ms */ + AIC3204_PDATA_AIN_PWRUP_1M6S = (0x33), +}; + +/* + * Analogue Reference Power Up Delay + */ +enum { + /* Reference Power-Up delay: Slow */ + AIC3204_PDATA_REF_PWRUP_SLOW = (0 << 0), + /* Reference Power-Up delay: 40ms */ + AIC3204_PDATA_REF_PWRUP_40MS = (1 << 0), + /* Reference Power-Up delay: 80ms */ + AIC3204_PDATA_REF_PWRUP_80MS = (2 << 0), + /* Reference Power-Up delay: 120ms */ + AIC3204_PDATA_REF_PWRUP_120MS = (3 << 0), +}; + +struct aic3204_pdata { + int gpio_reset; /* < 0 if not used */ + + /* Analogue-in Power up delay */ + u8 ain_pwrup_delay; + + /* Analogue Reference Power up delay */ + u8 ref_pwrup_delay; + + /* AVDD <->DVDD Link enable */ + u8 avdd_link_en; + + /* LDO Enable */ + u8 ldo_en; + + /* Oversampling rate (both ADC and DAC) */ + u8 osr; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c37c844..80b56a4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -32,6 +32,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER + select SND_SOC_TLV320AIC3204 if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C @@ -164,6 +165,10 @@ config SND_SOC_TLV320AIC26 tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE depends on SPI +config SND_SOC_TLV320AIC3204 + tristate + depends on I2C + config SND_SOC_TLV320AIC3X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4a9c205..c649f14 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -20,6 +20,7 @@ snd-soc-ssm2602-objs := ssm2602.o snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o +snd-soc-tlv320aic3204-objs := tlv320aic3204.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o snd-soc-twl4030-objs := twl4030.o @@ -86,6 +87,7 @@ obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o +obj-$(CONFIG_SND_SOC_TLV320AIC3204) += snd-soc-tlv320aic3204.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o diff --git a/sound/soc/codecs/tlv320aic3204.c b/sound/soc/codecs/tlv320aic3204.c new file mode 100644 index 0000000..6b185c5 --- /dev/null +++ b/sound/soc/codecs/tlv320aic3204.c @@ -0,0 +1,4197 @@ +/* + * ALSA SoC TLV320AIC3204 codec driver + * + * Author: Stuart Longland, <redhatter(a)gentoo.org> + * Copyright: (C) 2010 Jacques Electronics Pty. Ltd. + * + * Based upon the TLV320AIC3X driver: + * Author: Vladimir Barinov, <vbarinov(a)embeddedalley.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/device.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/regulator/consumer.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/tlv320aic3204.h> + +#include "tlv320aic3204.h" + +/* + * Default settings; these will be used if no platform data is provided. + */ +static const struct aic3204_pdata aic3204_default_pdata = { + .gpio_reset = -1, + .ain_pwrup_delay = AIC3204_PDATA_AIN_PWRUP_DEFAULT, + .ref_pwrup_delay = AIC3204_PDATA_REF_PWRUP_SLOW, + .avdd_link_en = 1, + .ldo_en = 0, + .osr = 128, +}; + +/* + * Default Register Values; this is used to initialise the register cache. + * There are 128 pages, of 128 registers each stored here. The data was + * captured from a TLV320AIC3204 immediately after reset, and therefore should + * be representative of what's displayed in the datasheet. + */ +static const u8 aic3204_default_reg_data[128*128] = { + /* --- Page 0 --- */ + 0x00, 0x00, 0x50, 0x00, 0x00, 0x11, 0x04, 0x00, + 0x00, 0x00, 0x00, 0x01, 0x01, 0x00, 0x80, 0x02, + 0x00, 0x08, 0x01, 0x01, 0x80, 0x01, 0x00, 0x04, + 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x01, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x12, 0x03, 0x02, + 0x02, 0x00, 0x00, 0x00, 0x01, 0x01, 0x00, 0x14, + 0x0c, 0x00, 0x00, 0x00, 0x6f, 0x38, 0x00, 0x00, + 0x00, 0x00, 0x00, 0xee, 0x10, 0xd8, 0x7e, 0xe3, + 0x00, 0x00, 0x88, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x7f, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x7f, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + + /* --- Page 1 --- */ + 0x01, 0x00, 0x0e, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x10, 0x00, 0x00, 0x00, 0x00, + 0x40, 0x40, 0x40, 0x40, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x80, 0x80, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + + /* --- Page 2 --- */ + 0x02, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + + /* --- Page 3 --- */ + 0x03, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + + /* --- Page 4 --- */ + 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + + /* --- Page 5 --- */ + 0x05, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + + /* --- Page 6 --- */ + 0x06, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + + /* --- Page 7 --- */ + 0x07, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + + /* --- Page 8 --- */ + 0x08, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x01, 0x17, 0x00, 0x00, + 0x01, 0x17, 0x00, 0x00, 0x7d, 0xd3, 0x00, 0x00, + 0x7f, 0xff, 0xff, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x7f, 0xff, 0xff, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x7f, 0xff, 0xff, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x7f, 0xff, 0xff, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x7f, 0xff, 0xff, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x7f, 0xff, 0xff, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + + /* --- Page 9 --- */ + 0x09, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x7f, 0xff, 0xff, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x7f, 0xff, 0xff, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 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0x00, 0x00, 0x00, 0x00, + + /* --- Page 117 --- */ + 0x75, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + + /* --- Page 118 --- */ + 0x76, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + + /* --- Page 119 --- */ + 0x77, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + + /* --- Page 120 --- */ + 0x78, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + + /* --- Page 121 --- */ + 0x79, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + + /* --- Page 122 --- */ + 0x7a, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + + /* --- Page 123 --- */ + 0x7b, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + + /* --- Page 124 --- */ + 0x7c, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + + /* --- Page 125 --- */ + 0x7d, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + + /* --- Page 126 --- */ + 0x7e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + + /* --- Page 127 --- */ + 0x7f, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, +}; + +#define AIC3204_NUM_SUPPLIES 4 +static const char *aic3204_supply_names[AIC3204_NUM_SUPPLIES] = { + "IOVDD", /* I/O Voltage */ + "DVDD", /* Digital Core Voltage */ + "AVDD", /* Analog DAC Voltage */ + "LDOin", /* Supply for internal LDOs and output amplifiers */ +}; + +/* codec private data */ +struct aic3204_priv { + struct snd_soc_codec codec; + struct regulator_bulk_data supplies[AIC3204_NUM_SUPPLIES]; + unsigned int sysclk; + struct aic3204_pdata pdata; +}; + +/* + * Determine whether a register is readable. + */ +static int aic3204_reg_is_readable(unsigned int reg) +{ + u8 page = reg >> 7; + reg &= 0x7f; + + /* Anything beyond page 70 is undefined */ + if (page > 70) + return 0; + + /* Page 62..70 */ + if (page >= 62) { + /* Register 0 is page select */ + if (reg == 0) + return 1; + /* Registers 1..7 are reserved */ + if (reg <= 7) + return 0; + /* Registers 8..127 are valid */ + return 1; + } + + /* Pages 53..62 are undefined */ + if (page >= 53) + return 0; + + /* Pages 45..52 */ + if (page >= 45) { + /* Register 0 is page select */ + if (reg == 0) + return 1; + /* Registers 1..7 are reserved */ + if (reg <= 7) + return 0; + /* Registers 8..127 are valid */ + return 1; + } + + /* Page 44 --- XXX The datasheet may have some typos here! */ + if (page == 44) { + /* + * Register 0 is page select + * Register 1 enables adaptive filtering + */ + if (reg <= 1) + return 1; + /* Registers 2..7 are reserved */ + if (reg <= 7) + return 0; + /* Registers 8..127 are valid */ + return 1; + } + + /* Pages 35..43 are undefined */ + if (page >= 35) + return 0; + + /* Pages 26..34 */ + if (page >= 26) { + /* Register 0 is page select */ + if (reg == 0) + return 1; + /* Registers 1..7 are reserved */ + if (reg <= 7) + return 0; + /* Registers 8..127 are valid */ + return 1; + } + + /* Pages 17..25 are undefined */ + if (page >= 17) + return 0; + + /* Pages 9..16 */ + if (page >= 9) { + if (reg == 0) + return 1; + if (reg <= 7) + return 0; + return 1; + } + + /* Page 8 */ + if (page == 8) { + if (reg <= 1) + return 1; + if (reg <= 7) + return 0; + return 1; + } + + /* Page 2..7 are undefined */ + if (page >= 2) + return 0; + + /* Page 1 */ + if (page == 1) { + if (reg <= 4) + return 1; + if (reg <= 8) + return 0; + if (reg <= 20) + return 1; + if (reg == 21) + return 0; + if (reg <= 25) + return 1; + if (reg <= 50) + return 0; + if (reg <= 52) + return 1; + if (reg == 53) + return 0; + if (reg <= 55) + return 1; + if (reg == 56) + return 0; + if (reg <= 63) + return 1; + if (reg <= 70) + return 0; + if (reg == 71) + return 1; + if (reg <= 122) + return 0; + if (reg == 123) + return 1; + return 0; + } + + /* Page 0 */ + if (reg == 0) + return 1; + if (reg <= 3) + return 0; + if (reg <= 8) + return 1; + if (reg <= 10) + return 0; + if (reg <= 14) + return 1; + if (reg <= 17) + return 0; + if (reg <= 20) + return 1; + if (reg <= 24) + return 0; + if (reg <= 34) + return 1; + if (reg == 35) + return 0; + if (reg <= 38) + return 1; + if (reg <= 41) + return 0; + if (reg <= 49) + return 1; + if (reg <= 51) + return 0; + if (reg <= 56) + return 1; + if (reg <= 59) + return 0; + if (reg <= 61) + return 1; + if (reg == 62) + return 0; + if (reg <= 79) + return 1; + if (reg == 80) + return 0; + if (reg <= 109) + return 1; + return 0; +} + +/* + * read aic3204 register cache + */ +static inline unsigned int aic3204_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + if (reg >= AIC3204_CACHEREGNUM) + return -EINVAL; + + return cache[reg]; +} + +/* + * Pretty-print the value of a register + */ +static int aic3204_show_reg(struct snd_soc_codec *codec, char *buf, + int buf_sz, unsigned int reg) +{ + return snprintf(buf, buf_sz, "%02x", + aic3204_read_reg_cache(codec, reg)); +} + +/* + * write aic3204 register cache + */ +static inline void aic3204_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= AIC3204_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the aic3204 register space + */ +static int aic3204_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + /* data is + * D15..D8 aic3204 register offset + * D7...D0 register data + */ + u8 data[2]; + + /* + * Register number; upper 8 bits indicate page + */ + if (reg && (aic3204_read_reg_cache(codec, AIC3204_PAGE_SELECT) + != (reg >> 7))) { + /* Select the required page */ + data[0] = 0; + data[1] = reg >> 7; + if (codec->hw_write(codec->control_data, data, 2) != 2) + return -EIO; + aic3204_write_reg_cache(codec, AIC3204_PAGE_SELECT, reg >> 7); + dev_dbg(codec->dev, "%s: Page %d selected\n", + __func__, reg >> 7); + } + + data[0] = reg & 0x7f; + data[1] = value & 0xff; + + if (codec->hw_write(codec->control_data, data, 2) != 2) + return -EIO; + + dev_dbg(codec->dev, "%s: pg %d reg %d[%04x] <= %02x\n", + __func__, reg >> 7, reg & 0x7f, reg, value); + aic3204_write_reg_cache(codec, reg, value); + return 0; +} + +/* + * read from the aic3204 register space + */ +static int aic3204_read(struct snd_soc_codec *codec, unsigned int reg, + u8 *value) +{ + *value = reg & 0xff; + + /* + * Register number; upper 8 bits indicate page + */ + if (reg && (aic3204_read_reg_cache(codec, AIC3204_PAGE_SELECT) + != (reg >> 7))) + /* Select the required page */ + aic3204_write(codec, AIC3204_PAGE_SELECT, (reg >> 7)); + + value[0] = i2c_smbus_read_byte_data(codec->control_data, value[0]); + dev_dbg(codec->dev, "%s: pg %d reg %d[%04x] => %02x\n", + __func__, reg >> 7, reg & 0x7f, reg, value[0]); + aic3204_write_reg_cache(codec, reg, *value); + return 0; +} + +/* + * Check BCLK direction. Returns >0 if the BCLK pin is an output, 0 if it's an + * input, or negative error. + */ +static inline int aic3204_check_bclk_is_output( + struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + int reg = snd_soc_read(source->codec, AIC3204_AISR1); + if (reg < 0) { + printk(KERN_WARNING "%s: read returned %d\n", __func__, reg); + return reg; + } + reg &= AIC3204_AISR1_BCLK; + + return reg & AIC3204_AISR1_BCLK_OUT; +} + +/* + * Check WCLK direction. Returns >0 if the WCLK pin is an output, 0 if it's an + * input, or negative error. + */ + +static inline int aic3204_check_wclk_is_output( + struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + int reg = snd_soc_read(source->codec, AIC3204_AISR1); + if (reg < 0) { + printk(KERN_WARNING "%s: read returned %d\n", __func__, reg); + return reg; + } + reg &= AIC3204_AISR1_WCLK; + + return reg & AIC3204_AISR1_WCLK_OUT; +} + +/* + * Check PLL source; called by the PLL input clock DAPM widgets. + * Returns 1 if the two given widgets are presently connected, 0 otherwise. + * Negative error number returned on error. + */ +static int aic3204_check_pll_clk_src(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + int reg = snd_soc_read(source->codec, AIC3204_CLK1); + if (reg < 0) { + printk(KERN_WARNING "%s: read returned %d\n", __func__, reg); + return reg; + } + + reg &= AIC3204_CLK1_PLLSRC; + + if (reg == AIC3204_CLK1_PLLSRC_MCLK) + reg = (!strcmp(source->name, "MCLK")); + else if (reg == AIC3204_CLK1_PLLSRC_BCLK) + reg = (!strcmp(source->name, "BCLK")); + else if (reg == AIC3204_CLK1_PLLSRC_GPIO) + reg = (!strcmp(source->name, "GPIO")); + else if (reg == AIC3204_CLK1_PLLSRC_DIN) + reg = (!strcmp(source->name, "DIN")); + else + reg = 0; + + if (reg) + dev_dbg(source->codec->dev, "%s: %s is connected to %s\n", + __func__, source->name, sink->name); + return reg; +} + +/* + * Check CODEC source; called by the CODEC_CLKIN input clock DAPM widgets. + * Returns 1 if the two given widgets are presently connected, 0 otherwise. + * Negative error number returned on error. + */ +static int aic3204_check_codec_clk_src(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + int reg = snd_soc_read(source->codec, AIC3204_CLK1); + if (reg < 0) { + printk(KERN_WARNING "%s: read returned %d\n", __func__, reg); + return reg; + } + + reg &= AIC3204_CLK1_CODECCLK; + + if (reg == AIC3204_CLK1_CODECCLK_MCLK) + reg = (!strcmp(source->name, "MCLK")); + else if (reg == AIC3204_CLK1_CODECCLK_BCLK) + reg = (!strcmp(source->name, "BCLK")); + else if (reg == AIC3204_CLK1_CODECCLK_GPIO) + reg = (!strcmp(source->name, "GPIO")); + else if (reg == AIC3204_CLK1_CODECCLK_PLL) + reg = (!strcmp(source->name, "PLL_CLK")); + else + reg = 0; + + if (reg) + dev_dbg(source->codec->dev, "%s: %s is connected to %s\n", + __func__, source->name, sink->name); + return reg; +} + +/* + * Check BCLK source; called by the BCLK input clock DAPM widgets. + * Returns 1 if the two given widgets are presently connected, 0 otherwise. + * Negative error number returned on error. + */ +static int aic3204_check_bclk_src(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + int reg = snd_soc_read(source->codec, AIC3204_AISR1); + if (reg < 0) { + printk(KERN_WARNING "%s: read returned %d\n", __func__, reg); + return reg; + } + reg &= AIC3204_AISR1_BCLK; + + if (reg == AIC3204_AISR1_BCLK_IN) + /* BCLK is externally generated */ + return 0; + + reg = snd_soc_read(source->codec, AIC3204_AISR3); + if (reg < 0) { + printk(KERN_WARNING "%s: read returned %d\n", __func__, reg); + return reg; + } + reg &= AIC3204_AISR3_BDIV; + + if (reg == AIC3204_AISR3_BDIV_DAC) + reg = (!strcmp(source->name, "DAC_CLK")); + else if (reg == AIC3204_AISR3_BDIV_DAC_MOD) + reg = (!strcmp(source->name, "DAC_MOD_CLK")); + else if (reg == AIC3204_AISR3_BDIV_ADC) + reg = (!strcmp(source->name, "ADC_CLK")); + else if (reg == AIC3204_AISR3_BDIV_ADC_MOD) + reg = (!strcmp(source->name, "ADC_MOD_CLK")); + else + reg = 0; + + if (reg) + dev_dbg(source->codec->dev, "%s: %s is connected to %s\n", + __func__, source->name, sink->name); + return reg; +} + +/* + * Check WCLK source; called by the WCLK input clock DAPM widgets. + * Returns 1 if the two given widgets are presently connected, 0 otherwise. + * Negative error number returned on error. + */ +static int aic3204_check_wclk_src(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + int reg = snd_soc_read(source->codec, AIC3204_AISR1); + if (reg < 0) { + printk(KERN_WARNING "%s: read returned %d\n", __func__, reg); + return reg; + } + reg &= AIC3204_AISR1_WCLK; + + if (reg == AIC3204_AISR1_WCLK_IN) + /* WCLK is externally generated */ + return 0; + + reg = snd_soc_read(source->codec, AIC3204_AISR6); + if (reg < 0) { + printk(KERN_WARNING "%s: read returned %d\n", __func__, reg); + return reg; + } + reg &= AIC3204_AISR6_WCLKOUT; + + if (reg == AIC3204_AISR6_WCLKOUT_DAC) + reg = (!strcmp(source->name, "DAC_FS")); + else if (reg == AIC3204_AISR6_WCLKOUT_ADC) + reg = (!strcmp(source->name, "ADC_FS")); + else + reg = 0; + + if (reg) + dev_dbg(source->codec->dev, "%s: %s is connected to %s\n", + __func__, source->name, sink->name); + + return reg; +} + +/* + * Event handler for audio interface BCLK; checks the state of the ADCs and + * DACs and switches the audio interface to the presently active clock. + */ +static int aic3204_aif_bclk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { + u8 adcs_up; + u8 dacs_up; + u8 bclk_src; + /* Determine the current source */ + int reg = snd_soc_read(w->codec, AIC3204_AISR1); + if (reg < 0) { + printk(KERN_WARNING "%s: read returned %d\n", __func__, reg); + return reg; + } + reg &= AIC3204_AISR1_BCLK; + if (reg == AIC3204_AISR1_BCLK_IN) + /* BCLK is an input, not our concirn */ + return 0; + + reg = snd_soc_read(w->codec, AIC3204_AISR3); + if (reg < 0) { + printk(KERN_WARNING "%s: read returned %d\n", __func__, reg); + return reg; + } + bclk_src = reg & AIC3204_AISR3_BDIV; + + /* + * These registers are not manipulated by us, so we must read them from + * the CODEC directly. + */ + reg = aic3204_read(w->codec, AIC3204_ADCF, &adcs_up); + if (reg < 0) { + printk(KERN_WARNING "%s: read returned %d\n", __func__, reg); + return reg; + } + adcs_up &= (AIC3204_ADCF_LEFT_UP | AIC3204_ADCF_RIGHT_UP); + + reg = aic3204_read(w->codec, AIC3204_DACF1, &dacs_up); + if (reg < 0) { + printk(KERN_WARNING "%s: read returned %d\n", __func__, reg); + return reg; + } + dacs_up &= (AIC3204_DACF1_LEFT_UP | AIC3204_DACF1_RIGHT_UP); + + dev_dbg(w->codec->dev, "%s: BCLK source=0x%02x, event=%d, " + "dacf1=0x%02x, adcf=0x%02x\n", __func__, bclk_src, + event, dacs_up, adcs_up); + + /* What are we set to now? */ + if (bclk_src == AIC3204_AISR3_BDIV_DAC) { + /* Currently listening to DAC... check that it's up */ + if (adcs_up && !dacs_up) + /* No, but the ADCs are up; switch to those */ + bclk_src = AIC3204_AISR3_BDIV_ADC; + } else if (bclk_src == AIC3204_AISR3_BDIV_DAC_MOD) { + /* Currently listening to DAC... check that it's up */ + if (adcs_up && !dacs_up) + /* No, but the ADCs are up; switch to those */ + bclk_src = AIC3204_AISR3_BDIV_ADC_MOD; + } else if (bclk_src == AIC3204_AISR3_BDIV_ADC) { + /* Currently listening to ADC... check that it's up */ + if (dacs_up && !adcs_up) + /* No, but the DACs are up; switch to those */ + bclk_src = AIC3204_AISR3_BDIV_DAC; + } else if (bclk_src == AIC3204_AISR3_BDIV_ADC_MOD) { + /* Currently listening to ADC... check that it's up */ + if (adcs_up && !dacs_up) + /* No, but the DACs are up; switch to those */ + bclk_src = AIC3204_AISR3_BDIV_DAC_MOD; + } + + dev_dbg(w->codec->dev, "%s: setting source 0x%02x\n", + __func__, bclk_src); + + snd_soc_update_bits(w->codec, AIC3204_AISR3, + AIC3204_AISR3_BDIV, bclk_src); + return 0; +} + +/* + * Event handler for audio interface WCLK; checks the state of the ADCs and + * DACs and switches the audio interface to the presently active clock. + */ +static int aic3204_aif_wclk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { + u8 wclk_src; + u8 adcs_up; + u8 dacs_up; + /* Determine the current source */ + int reg = snd_soc_read(w->codec, AIC3204_AISR1); + if (reg < 0) { + printk(KERN_WARNING "%s: read returned %d\n", __func__, reg); + return reg; + } + reg &= AIC3204_AISR1_WCLK; + + if (reg == AIC3204_AISR1_WCLK_IN) + /* WCLK is an input, not our concirn */ + return 0; + + reg = snd_soc_read(w->codec, AIC3204_AISR6); + if (reg < 0) { + printk(KERN_WARNING "%s: read returned %d\n", __func__, reg); + return reg; + } + wclk_src = reg & AIC3204_AISR6_WCLKOUT; + + /* + * These registers are not manipulated by us, so we must read them from + * the CODEC directly. + */ + reg = aic3204_read(w->codec, AIC3204_ADCF, &adcs_up); + if (reg < 0) { + printk(KERN_WARNING "%s: read returned %d\n", __func__, reg); + return reg; + } + adcs_up &= (AIC3204_ADCF_LEFT_UP | AIC3204_ADCF_RIGHT_UP); + + reg = aic3204_read(w->codec, AIC3204_DACF1, &dacs_up); + if (reg < 0) { + printk(KERN_WARNING "%s: read returned %d\n", __func__, reg); + return reg; + } + dacs_up &= (AIC3204_DACF1_LEFT_UP | AIC3204_DACF1_RIGHT_UP); + + dev_dbg(w->codec->dev, "%s: WCLK source=0x%02x, event=%d, " + "dacf1=0x%02x, adcf=0x%02x\n", __func__, wclk_src, + event, dacs_up, adcs_up); + + /* What are we set to now? */ + if (wclk_src == AIC3204_AISR6_WCLKOUT_DAC) { + /* Currently listening to DAC... check that it's up */ + if (adcs_up && !dacs_up) + /* No, but the ADCs are up; switch to those */ + wclk_src = AIC3204_AISR6_WCLKOUT_ADC; + } else if (wclk_src == AIC3204_AISR6_WCLKOUT_ADC) { + /* Currently listening to ADC... check that it's up */ + if (dacs_up && !adcs_up) + /* No, but the DACs are up; switch to those */ + wclk_src = AIC3204_AISR6_WCLKOUT_DAC; + } + + dev_dbg(w->codec->dev, "%s: setting source 0x%02x\n", + __func__, wclk_src); + + snd_soc_update_bits(w->codec, AIC3204_AISR6, + AIC3204_AISR6_WCLKOUT, wclk_src); + return 0; +} + +/* + * For MICPGA inputs; inputs may either be disconnected, or connected via one + * of three resistances. + */ +static const char *aic3204_micpga_input_mux[] = { + "not-connected", "10k", "20k", "40k" +}; + +static struct soc_enum aic3204_enum_in1l_lmicpgap = + SOC_ENUM_SINGLE(AIC3204_LMICPGAPROUTE, + 6, 4, aic3204_micpga_input_mux); +static struct soc_enum aic3204_enum_in2l_lmicpgap = + SOC_ENUM_SINGLE(AIC3204_LMICPGAPROUTE, + 4, 4, aic3204_micpga_input_mux); +static struct soc_enum aic3204_enum_in3l_lmicpgap = + SOC_ENUM_SINGLE(AIC3204_LMICPGAPROUTE, + 2, 4, aic3204_micpga_input_mux); +static struct soc_enum aic3204_enum_in1r_lmicpgap = + SOC_ENUM_SINGLE(AIC3204_LMICPGAPROUTE, + 0, 4, aic3204_micpga_input_mux); + +static struct soc_enum aic3204_enum_cm1l_lmicpgan = + SOC_ENUM_SINGLE(AIC3204_LMICPGANROUTE, + 6, 4, aic3204_micpga_input_mux); +static struct soc_enum aic3204_enum_in2r_lmicpgan = + SOC_ENUM_SINGLE(AIC3204_LMICPGANROUTE, + 4, 4, aic3204_micpga_input_mux); +static struct soc_enum aic3204_enum_in3r_lmicpgan = + SOC_ENUM_SINGLE(AIC3204_LMICPGANROUTE, + 2, 4, aic3204_micpga_input_mux); +static struct soc_enum aic3204_enum_cm2l_lmicpgan = + SOC_ENUM_SINGLE(AIC3204_LMICPGANROUTE, + 0, 4, aic3204_micpga_input_mux); + +static struct soc_enum aic3204_enum_in1r_rmicpgap = + SOC_ENUM_SINGLE(AIC3204_RMICPGAPROUTE, + 6, 4, aic3204_micpga_input_mux); +static struct soc_enum aic3204_enum_in2r_rmicpgap = + SOC_ENUM_SINGLE(AIC3204_RMICPGAPROUTE, + 4, 4, aic3204_micpga_input_mux); +static struct soc_enum aic3204_enum_in3r_rmicpgap = + SOC_ENUM_SINGLE(AIC3204_RMICPGAPROUTE, + 2, 4, aic3204_micpga_input_mux); +static struct soc_enum aic3204_enum_in2l_rmicpgap = + SOC_ENUM_SINGLE(AIC3204_RMICPGAPROUTE, + 0, 4, aic3204_micpga_input_mux); + +static struct soc_enum aic3204_enum_cm1r_rmicpgan = + SOC_ENUM_SINGLE(AIC3204_RMICPGANROUTE, + 6, 4, aic3204_micpga_input_mux); +static struct soc_enum aic3204_enum_in1l_rmicpgan = + SOC_ENUM_SINGLE(AIC3204_RMICPGANROUTE, + 4, 4, aic3204_micpga_input_mux); +static struct soc_enum aic3204_enum_in3l_rmicpgan = + SOC_ENUM_SINGLE(AIC3204_RMICPGANROUTE, + 2, 4, aic3204_micpga_input_mux); +static struct soc_enum aic3204_enum_cm2r_rmicpgan = + SOC_ENUM_SINGLE(AIC3204_RMICPGANROUTE, + 0, 4, aic3204_micpga_input_mux); + +/* + * MICBIAS Voltages; The exact voltage depends on the common mode voltage + * setting... + * + * CM=0.75V CM=0.9V + * Low 1.04V 1.25V + * Medium 1.425V 1.7V + * High 2.075V 2.5V + * V+ Connects to power supply directly in both cases. + */ +static const char *aic3204_micbias_voltages[] = { + "Low", "Med", "High", "V+" +}; +static struct soc_enum aic3204_enum_micbias_voltage = + SOC_ENUM_SINGLE(AIC3204_MICBIAS, + 4, 4, aic3204_micbias_voltages); + +/* + * MICBIAS Voltage Source + */ +static const char *aic3204_micbias_srcs[] = { + "AVDD", "LDOIN" +}; +static struct soc_enum aic3204_enum_micbias_src = + SOC_ENUM_SINGLE(AIC3204_MICBIAS, + 3, 2, aic3204_micbias_srcs); + +/* + * Left DAC Data Source + */ +static const char *aic3204_ldac_srcs[] = { + "disabled", "left channel", "right channel", "mix" +}; +static struct soc_enum aic3204_enum_ldac_src = + SOC_ENUM_SINGLE(AIC3204_DACS1, + 4, 4, aic3204_ldac_srcs); + +/* + * Right DAC Data Source + */ +static const char *aic3204_rdac_srcs[] = { + "disabled", "right channel", "left channel", "mix" +}; +static struct soc_enum aic3204_enum_rdac_src = + SOC_ENUM_SINGLE(AIC3204_DACS1, + 2, 4, aic3204_rdac_srcs); + +/* + * DAC digital volumes. From -63.5 to +24 dB in 0.5 dB steps. + * Does not mute control. + */ +static DECLARE_TLV_DB_SCALE(dac_tlv, -6350, 50, 0); +/* + * ADC gain volumes. From -12dB to 20 dB in 0.5dB steps. + * Does not mute control. + */ +static DECLARE_TLV_DB_SCALE(adc_tlv, -1200, 50, 0); +#if 0 +/* + * ADC Fine gain volumes. From -0.4 to 0 dB in 0.1dB steps + * Does not mute control. + */ +static DECLARE_TLV_DB_SCALE(adc_fine_tlv, -40, 10, 0); +#endif +/* + * Output driver stage volumes; -6dB through to +29dB in 1dB steps + * Does not mute control. + */ +static DECLARE_TLV_DB_SCALE(output_stage_tlv, -600, 100, 0); +/* + * MICPGA volumes; 0dB through to +47.5dB in 0.5dB steps. + * Does not mute control + */ +static DECLARE_TLV_DB_SCALE(micpga_tlv, 0, 50, 0); + +static const struct snd_kcontrol_new aic3204_snd_controls[] = { + /* Output */ + + /* + * Note regarding SOC_DOUBLE_R_SX_TLV... + * + * This is a bit misleading; xshift refers to the bit one bit *past* + * the most significant bit. Or at least that's what it appears to be + * doing the math. Mask needs to select the last 6 bits; which is the + * signed bitfield specifiying the gain in dB. + */ + SOC_DOUBLE_R_SX_TLV("Headphone Output Volume", + AIC3204_HPLGAIN, AIC3204_HPRGAIN, + 6, -6, 29, output_stage_tlv), + SOC_DOUBLE_R("Headphone Output Playback Switch", + AIC3204_HPLGAIN, AIC3204_HPRGAIN, + AIC3204_HPLGAIN_MUTE_SHIFT, 0x01, 1), + SOC_DOUBLE_R_SX_TLV("Line Output Playback Volume", + AIC3204_LOLGAIN, AIC3204_LORGAIN, + 6, -6, 29, output_stage_tlv), + SOC_DOUBLE_R("Line Output Playback Switch", AIC3204_LOLGAIN, + AIC3204_LORGAIN, AIC3204_LOLGAIN_MUTE_SHIFT, 0x01, 1), + SOC_DOUBLE_R_SX_TLV("PCM Playback Volume", + AIC3204_LDACVOL, AIC3204_RDACVOL, + 8, -127, 48, dac_tlv), + SOC_DOUBLE("PCM Playback Switch", AIC3204_DACS2, + AIC3204_DACS2_LEFT_MUTE_SHIFT, + AIC3204_DACS2_RIGHT_MUTE_SHIFT, 0x01, 1), + SOC_DOUBLE_R_SX_TLV("Mixer Amplifier Volume", + AIC3204_MALGAIN, AIC3204_MARGAIN, + 6, -6, 29, output_stage_tlv), + + /* DAC Options */ + SOC_ENUM("Left Playback Source", aic3204_enum_ldac_src), + SOC_ENUM("Right Playback Source", aic3204_enum_rdac_src), + SOC_SINGLE("Differential Output Switch", AIC3204_DACS2, + AIC3204_DACS2_RMOD_INV_SHIFT, 1, 0), + + /* MICPGA Options */ + SOC_DOUBLE_R_TLV("MICPGA Capture Volume", + AIC3204_LMICPGAVOL, AIC3204_RMICPGAVOL, + 0, 95, 0, micpga_tlv), + SOC_DOUBLE_R("MICPGA Capture Switch", + AIC3204_LMICPGAVOL, AIC3204_RMICPGAVOL, 7, 0x1, 1), + + /* MICBIAS Options */ + SOC_SINGLE("MICBIAS Enable", AIC3204_MICBIAS, 6, 0x1, 0), + SOC_ENUM("MICBIAS Level", aic3204_enum_micbias_voltage), + SOC_ENUM("MICBIAS Source", aic3204_enum_micbias_src), + + /* ADC */ + SOC_DOUBLE_R_SX_TLV("PCM Capture Volume", + AIC3204_LADCVOL, AIC3204_RADCVOL, + 7, -24, 40, adc_tlv), + SOC_DOUBLE("PCM Capture Switch", AIC3204_ADCFINE, 7, 3, 0x01, 1), + + /* Driver/Amplifier Selection -- DAPM should handle this! */ + SOC_DOUBLE("HP Playback Switch", AIC3204_OUTDRV, + AIC3204_OUTDRV_HPL_UP_SHIFT, + AIC3204_OUTDRV_HPR_UP_SHIFT, 0x1, 0), + SOC_DOUBLE("LO Playback Switch", AIC3204_OUTDRV, + AIC3204_OUTDRV_LOL_UP_SHIFT, + AIC3204_OUTDRV_LOR_UP_SHIFT, 0x1, 0), + SOC_DOUBLE("MA Playback Switch", AIC3204_OUTDRV, + AIC3204_OUTDRV_MAL_UP_SHIFT, + AIC3204_OUTDRV_MAR_UP_SHIFT, 0x1, 0), +}; + +/* Driver/Amplifier Selection */ +static const struct snd_kcontrol_new aic3204_hpl_switch = + SOC_DAPM_SINGLE("Switch", AIC3204_OUTDRV, + AIC3204_OUTDRV_HPL_UP_SHIFT, 0x1, 0); +static const struct snd_kcontrol_new aic3204_hpr_switch = + SOC_DAPM_SINGLE("Switch", AIC3204_OUTDRV, + AIC3204_OUTDRV_HPR_UP_SHIFT, 0x1, 0); +static const struct snd_kcontrol_new aic3204_lol_switch = + SOC_DAPM_SINGLE("Switch", AIC3204_OUTDRV, + AIC3204_OUTDRV_LOL_UP_SHIFT, 0x1, 0); +static const struct snd_kcontrol_new aic3204_lor_switch = + SOC_DAPM_SINGLE("Switch", AIC3204_OUTDRV, + AIC3204_OUTDRV_LOR_UP_SHIFT, 0x1, 0); +static const struct snd_kcontrol_new aic3204_mal_switch = + SOC_DAPM_SINGLE("Switch", AIC3204_OUTDRV, + AIC3204_OUTDRV_MAL_UP_SHIFT, 0x1, 0); +static const struct snd_kcontrol_new aic3204_mar_switch = + SOC_DAPM_SINGLE("Switch", AIC3204_OUTDRV, + AIC3204_OUTDRV_MAR_UP_SHIFT, 0x1, 0); + +/* Line Output Left Driver Inputs */ +static const struct snd_kcontrol_new aic3204_lol_in_controls[] = { + SOC_DAPM_SINGLE("LOR Switch", AIC3204_LOLROUTE, + AIC3204_LOLROUTE_LOR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("MAL Switch", AIC3204_LOLROUTE, + AIC3204_LOLROUTE_MAL_SHIFT, 1, 0), + SOC_DAPM_SINGLE("LDAC+ Switch", AIC3204_LOLROUTE, + AIC3204_LOLROUTE_LDACP_SHIFT, 1, 0), + SOC_DAPM_SINGLE("RDAC- Switch", AIC3204_LOLROUTE, + AIC3204_LOLROUTE_RDACN_SHIFT, 1, 0), +}; + +/* Line Output Right Driver Inputs */ +static const struct snd_kcontrol_new aic3204_lor_in_controls[] = { + SOC_DAPM_SINGLE("MAR Switch", AIC3204_LORROUTE, + AIC3204_LORROUTE_MAR_SHIFT, 1, 0), + SOC_DAPM_SINGLE("RDAC+ Switch", AIC3204_LORROUTE, + AIC3204_LORROUTE_RDACP_SHIFT, 1, 0), +}; + +/* Headphone Output Left Driver Inputs */ +static const struct snd_kcontrol_new aic3204_hpl_in_controls[] = { + SOC_DAPM_SINGLE("MAR Switch", AIC3204_HPLROUTE, + AIC3204_HPLROUTE_MAR_SHIFT, 0x1, 0), + SOC_DAPM_SINGLE("MAL Switch", AIC3204_HPLROUTE, + AIC3204_HPLROUTE_MAL_SHIFT, 0x1, 0), + SOC_DAPM_SINGLE("IN1L Switch", AIC3204_HPLROUTE, + AIC3204_HPLROUTE_IN1L_SHIFT, 0x1, 0), + SOC_DAPM_SINGLE("LDAC+ Switch", AIC3204_HPLROUTE, + AIC3204_HPLROUTE_LDACP_SHIFT, 0x1, 0), +}; + +/* Headphone Output Right Driver Inputs */ +static const struct snd_kcontrol_new aic3204_hpr_in_controls[] = { + SOC_DAPM_SINGLE("HPL Switch", AIC3204_HPRROUTE, + AIC3204_HPRROUTE_HPL_SHIFT, 0x1, 0), + SOC_DAPM_SINGLE("MAR Switch", AIC3204_HPRROUTE, + AIC3204_HPRROUTE_MAR_SHIFT, 0x1, 0), + SOC_DAPM_SINGLE("IN1R Switch", AIC3204_HPRROUTE, + AIC3204_HPRROUTE_IN1R_SHIFT, 0x1, 0), + SOC_DAPM_SINGLE("RDAC+ Switch", AIC3204_HPRROUTE, + AIC3204_HPRROUTE_RDACP_SHIFT, 0x1, 0), + SOC_DAPM_SINGLE("LDAC- Switch", AIC3204_HPRROUTE, + AIC3204_HPRROUTE_LDACN_SHIFT, 0x1, 0), +}; + +static const struct snd_kcontrol_new aic3204_lmicpgap_in_controls[] = { + SOC_DAPM_ENUM("IN1L Switch", aic3204_enum_in1l_lmicpgap), + SOC_DAPM_ENUM("IN2L Switch", aic3204_enum_in2l_lmicpgap), + SOC_DAPM_ENUM("IN3L Switch", aic3204_enum_in3l_lmicpgap), + SOC_DAPM_ENUM("IN1R Switch", aic3204_enum_in1r_lmicpgap), +}; + +static const struct snd_kcontrol_new aic3204_lmicpgan_in_controls[] = { + SOC_DAPM_ENUM("CM1L Switch", aic3204_enum_cm1l_lmicpgan), + SOC_DAPM_ENUM("IN2R Switch", aic3204_enum_in2r_lmicpgan), + SOC_DAPM_ENUM("IN3R Switch", aic3204_enum_in3r_lmicpgan), + SOC_DAPM_ENUM("CM2L Switch", aic3204_enum_cm2l_lmicpgan), +}; + +static const struct snd_kcontrol_new aic3204_rmicpgap_in_controls[] = { + SOC_DAPM_ENUM("IN1R Switch", aic3204_enum_in1r_rmicpgap), + SOC_DAPM_ENUM("IN2R Switch", aic3204_enum_in2r_rmicpgap), + SOC_DAPM_ENUM("IN3R Switch", aic3204_enum_in3r_rmicpgap), + SOC_DAPM_ENUM("IN2L Switch", aic3204_enum_in2l_rmicpgap), +}; + +static const struct snd_kcontrol_new aic3204_rmicpgan_in_controls[] = { + SOC_DAPM_ENUM("CM1R Switch", aic3204_enum_cm1r_rmicpgan), + SOC_DAPM_ENUM("IN1L Switch", aic3204_enum_in1l_rmicpgan), + SOC_DAPM_ENUM("IN3L Switch", aic3204_enum_in3l_rmicpgan), + SOC_DAPM_ENUM("CM2R Switch", aic3204_enum_cm2r_rmicpgan), +}; + +static const struct snd_soc_dapm_widget aic3204_dapm_widgets[] = { + /* Driver/Amplifier Selection */ + SND_SOC_DAPM_SWITCH("HPL Playback Switch", AIC3204_OUTDRV, + AIC3204_OUTDRV_HPL_UP_SHIFT, 0, + &aic3204_hpl_switch), + SND_SOC_DAPM_SWITCH("HPR Playback Switch", AIC3204_OUTDRV, + AIC3204_OUTDRV_HPR_UP_SHIFT, 0, + &aic3204_hpr_switch), + SND_SOC_DAPM_SWITCH("LOL Playback Switch", AIC3204_OUTDRV, + AIC3204_OUTDRV_LOL_UP_SHIFT, 0, + &aic3204_lol_switch), + SND_SOC_DAPM_SWITCH("LOR Playback Switch", AIC3204_OUTDRV, + AIC3204_OUTDRV_LOR_UP_SHIFT, 0, + &aic3204_lor_switch), + SND_SOC_DAPM_SWITCH("MAL Playback Switch", AIC3204_OUTDRV, + AIC3204_OUTDRV_MAL_UP_SHIFT, 0, + &aic3204_mal_switch), + SND_SOC_DAPM_SWITCH("MAR Playback Switch", AIC3204_OUTDRV, + AIC3204_OUTDRV_MAR_UP_SHIFT, 0, + &aic3204_mar_switch), + + /* Line Output Driver Inputs */ + SND_SOC_DAPM_MIXER("Left Line Output Mixer", SND_SOC_NOPM, 0, 0, + aic3204_lol_in_controls, + ARRAY_SIZE(aic3204_lol_in_controls)), + SND_SOC_DAPM_MIXER("Right Line Output Mixer", SND_SOC_NOPM, 0, 0, + aic3204_lor_in_controls, + ARRAY_SIZE(aic3204_lor_in_controls)), + + /* Headphone Output Driver Inputs */ + SND_SOC_DAPM_MIXER("Left Headphone Output Mixer", SND_SOC_NOPM, 0, 0, + aic3204_hpl_in_controls, + ARRAY_SIZE(aic3204_hpl_in_controls)), + SND_SOC_DAPM_MIXER("Right Headphone Output Mixer", SND_SOC_NOPM, 0, 0, + aic3204_hpr_in_controls, + ARRAY_SIZE(aic3204_hpr_in_controls)), + + /* DACs, ADCs and audio interfaces */ + SND_SOC_DAPM_DAC("Left DAC", "Left Playback", AIC3204_DACS1, + AIC3204_DACS1_LDAC_UP_SHIFT, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right Playback", AIC3204_DACS1, + AIC3204_DACS1_RDAC_UP_SHIFT, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC3204_ADCS, + AIC3204_ADCS_RADC_UP_SHIFT, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", AIC3204_ADCS, + AIC3204_ADCS_RADC_UP_SHIFT, 0), + SND_SOC_DAPM_AIF_IN("Left Data In", "Left Playback", + 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("Right Data In", "Right Playback", + 1, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("Left Data Out", "Left Capture", + 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("Right Data Out", "Right Capture", + 1, SND_SOC_NOPM, 0, 0), + + /* System clock sources */ + SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("BCLK", AIC3204_CLK12, + AIC3204_CLK12_BCLK_STATE_SHIFT, 0, + aic3204_aif_bclk_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("WCLK", SND_SOC_NOPM, 0, 0, + aic3204_aif_wclk_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("GPIO", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DIN", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL_CLK", AIC3204_CLK2, + AIC3204_CLK2_PLL_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CODEC_CLKIN", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_CLK", AIC3204_CLK6, + AIC3204_CLK6_NDAC_STATE_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_MOD_CLK", AIC3204_CLK7, + AIC3204_CLK7_MDAC_STATE_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC_CLK", AIC3204_CLK8, + AIC3204_CLK8_NADC_STATE_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC_MOD_CLK", AIC3204_CLK9, + AIC3204_CLK9_MADC_STATE_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC_FS", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC_FS", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* MICPGA Input Sources */ + SND_SOC_DAPM_MIXER("Left MICPGA+ Output Mixer", SND_SOC_NOPM, 0, 0, + aic3204_lmicpgap_in_controls, + ARRAY_SIZE(aic3204_lmicpgap_in_controls)), + SND_SOC_DAPM_MIXER("Left MICPGA- Output Mixer", SND_SOC_NOPM, 0, 0, + aic3204_lmicpgan_in_controls, + ARRAY_SIZE(aic3204_lmicpgan_in_controls)), + SND_SOC_DAPM_MIXER("Right MICPGA+ Output Mixer", SND_SOC_NOPM, 0, 0, + aic3204_rmicpgap_in_controls, + ARRAY_SIZE(aic3204_rmicpgap_in_controls)), + SND_SOC_DAPM_MIXER("Right MICPGA- Output Mixer", SND_SOC_NOPM, 0, 0, + aic3204_rmicpgan_in_controls, + ARRAY_SIZE(aic3204_rmicpgan_in_controls)), + SND_SOC_DAPM_PGA("Left MICPGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right MICPGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MICBIAS("Mic Bias", AIC3204_MICBIAS, 6, 0), + + /* Bypass PGAs */ + SND_SOC_DAPM_PGA("IN1L Bypass PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IN1R Bypass PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Input Pins */ + SND_SOC_DAPM_INPUT("IN1L"), + SND_SOC_DAPM_INPUT("IN1R"), + SND_SOC_DAPM_INPUT("IN2L"), + SND_SOC_DAPM_INPUT("IN2R"), + SND_SOC_DAPM_INPUT("IN3L"), + SND_SOC_DAPM_INPUT("IN3R"), + + /* Output Pins */ + SND_SOC_DAPM_OUTPUT("LOL"), + SND_SOC_DAPM_OUTPUT("LOR"), + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + + /* Common-mode pins */ + SND_SOC_DAPM_VMID("CM1L"), + SND_SOC_DAPM_VMID("CM1R"), + SND_SOC_DAPM_VMID("CM2L"), + SND_SOC_DAPM_VMID("CM2R"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Clock Signals and routing */ + {"PLL_CLK", NULL, "MCLK", aic3204_check_pll_clk_src}, + {"PLL_CLK", NULL, "BCLK", aic3204_check_pll_clk_src}, + {"PLL_CLK", NULL, "GPIO", aic3204_check_pll_clk_src}, + {"PLL_CLK", NULL, "DIN", aic3204_check_pll_clk_src}, + {"CODEC_CLKIN", NULL, "MCLK", aic3204_check_codec_clk_src}, + {"CODEC_CLKIN", NULL, "BCLK", aic3204_check_codec_clk_src}, + {"CODEC_CLKIN", NULL, "GPIO", aic3204_check_codec_clk_src}, + {"CODEC_CLKIN", NULL, "PLL_CLK", aic3204_check_codec_clk_src}, + {"DAC_CLK", NULL, "CODEC_CLKIN"}, + {"DAC_MOD_CLK", NULL, "DAC_CLK"}, + {"DAC_FS", NULL, "DAC_MOD_CLK"}, + {"ADC_CLK", NULL, "CODEC_CLKIN"}, + {"ADC_MOD_CLK", NULL, "ADC_CLK"}, + {"ADC_FS", NULL, "ADC_MOD_CLK"}, + {"BCLK", NULL, "DAC_CLK", aic3204_check_bclk_src}, + {"BCLK", NULL, "DAC_MOD_CLK", aic3204_check_bclk_src}, + {"BCLK", NULL, "ADC_CLK", aic3204_check_bclk_src}, + {"BCLK", NULL, "ADC_MOD_CLK", aic3204_check_bclk_src}, + {"WCLK", NULL, "DAC_FS", aic3204_check_wclk_src}, + {"WCLK", NULL, "ADC_FS", aic3204_check_wclk_src}, + + /* Clock sinks */ + {"Left DAC", NULL, "DAC_MOD_CLK"}, + {"Right DAC", NULL, "DAC_MOD_CLK"}, + {"Left ADC", NULL, "ADC_MOD_CLK"}, + {"Right ADC", NULL, "ADC_MOD_CLK"}, + {"Left Data Out", NULL, "BCLK", aic3204_check_bclk_is_output}, + {"Right Data Out", NULL, "BCLK", aic3204_check_bclk_is_output}, + {"Left Data Out", NULL, "WCLK", aic3204_check_wclk_is_output}, + {"Right Data Out", NULL, "WCLK", aic3204_check_wclk_is_output}, + {"Left Data In", NULL, "BCLK", aic3204_check_bclk_is_output}, + {"Right Data In", NULL, "BCLK", aic3204_check_bclk_is_output}, + {"Left Data In", NULL, "WCLK", aic3204_check_wclk_is_output}, + {"Right Data In", NULL, "WCLK", aic3204_check_wclk_is_output}, + + /* Data Connections */ + {"Left Data Out", NULL, "Left ADC"}, + {"Right Data Out", NULL, "Right ADC"}, + {"Left DAC", NULL, "Left Data In"}, + {"Right DAC", NULL, "Right Data In"}, + + /* Line Output */ + {"Left Line Output Mixer", "LOR Switch", "Right Line Output Mixer"}, + {"Left Line Output Mixer", "MAL Switch", "Left MICPGA"}, + {"Left Line Output Mixer", "LDAC+ Switch", "Left DAC"}, + {"Left Line Output Mixer", "RDAC- Switch", "Right DAC"}, + {"Right Line Output Mixer", "MAR Switch", "Right MICPGA"}, + {"Right Line Output Mixer", "RDAC+ Switch", "Right DAC"}, + {"LOL", "LOL Playback Switch", "Left Line Output Mixer"}, + {"LOR", "LOR Playback Switch", "Right Line Output Mixer"}, + + /* Headphone Output */ + {"Left Headphone Output Mixer", "MAR Switch", "Right MICPGA"}, + {"Left Headphone Output Mixer", "MAL Switch", "Left MICPGA"}, + {"Left Headphone Output Mixer", "IN1L Switch", "IN1L Bypass PGA"}, + {"Left Headphone Output Mixer", "LDAC+ Switch", "Left DAC"}, + {"Right Headphone Output Mixer", "HPL Switch", + "Left Headphone Output Mixer"}, + {"Right Headphone Output Mixer", "MAR Switch", "Right MICPGA"}, + {"Right Headphone Output Mixer", "IN1R Switch", "IN1R Bypass PGA"}, + {"Right Headphone Output Mixer", "RDAC+ Switch", "Right DAC"}, + {"Right Headphone Output Mixer", "LDAC- Switch", "Left DAC"}, + {"HPL", "HPL Playback Switch", "Left Headphone Output Mixer"}, + {"HPR", "HPR Playback Switch", "Right Headphone Output Mixer"}, + + /* MICPGA Input */ + {"Left MICPGA+ Output Mixer", "IN1L Switch", "IN1L"}, + {"Left MICPGA+ Output Mixer", "IN2L Switch", "IN2L"}, + {"Left MICPGA+ Output Mixer", "IN3L Switch", "IN3L"}, + {"Left MICPGA+ Output Mixer", "IN1R Switch", "IN1R"}, + {"Left MICPGA- Output Mixer", "CM1L Switch", "CM1L"}, + {"Left MICPGA- Output Mixer", "IN2R Switch", "IN2R"}, + {"Left MICPGA- Output Mixer", "IN3R Switch", "IN3R"}, + {"Left MICPGA- Output Mixer", "CM2L Switch", "CM2L"}, + {"Right MICPGA+ Output Mixer", "IN1R Switch", "IN1R"}, + {"Right MICPGA+ Output Mixer", "IN2R Switch", "IN2R"}, + {"Right MICPGA+ Output Mixer", "IN3R Switch", "IN3R"}, + {"Right MICPGA+ Output Mixer", "IN2L Switch", "IN2L"}, + {"Right MICPGA- Output Mixer", "CM1R Switch", "CM1R"}, + {"Right MICPGA- Output Mixer", "IN1L Switch", "IN1L"}, + {"Right MICPGA- Output Mixer", "IN3L Switch", "IN3L"}, + {"Right MICPGA- Output Mixer", "CM2R Switch", "CM2R"}, + {"Left MICPGA", "MAL Playback Switch", "Left MICPGA+ Output Mixer"}, + {"Right MICPGA", "MAR Playback Switch", "Right MICPGA+ Output Mixer"}, + {"Left MICPGA", "MAL Playback Switch", "Left MICPGA- Output Mixer"}, + {"Right MICPGA", "MAR Playback Switch", "Right MICPGA- Output Mixer"}, + + /* ADC Input */ + {"Left ADC", NULL, "Left MICPGA"}, + {"Right ADC", NULL, "Right MICPGA"}, +}; + +static int aic3204_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, aic3204_dapm_widgets, + ARRAY_SIZE(aic3204_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + return 0; +} + +static int aic3204_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct aic3204_priv *aic3204 = snd_soc_codec_get_drvdata(codec); + int dac_clk = 0, bit_clk = 0; + int codec_clk = 0, bypass_pll = 0, last_clk = 0, last_error = 0; + u8 data = 0, ndac = 0, mdac = 0, bdiv = 1; + u8 pll_p = 1, pll_r = 1, pll_j = 1; + u16 pll_d = 0; + + int this_clk = 0, this_error = 0; + unsigned int this_abs_error; + u8 j, this_ndac; + u16 d; + + /* Determine the DAC clock and bit clock needed */ + dac_clk = aic3204->pdata.osr * params_rate(params); + bit_clk = params_rate(params) * params_channels(params); + + /* + * Select data word length. + * Note that the number of bits per frame used + * to determine BDIV is double the number of bits per sample, as there + * are two samples per frame. + */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + pr_debug("%s: Sample format 16 bits\n", __func__); + bit_clk *= 16; + data = AIC3204_AISR1_WL_16BITS; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + pr_debug("%s: Sample format 20 bits\n", __func__); + bit_clk *= 32; + data = AIC3204_AISR1_WL_20BITS; + break; + case SNDRV_PCM_FORMAT_S24_LE: + pr_debug("%s: Sample format 24 bits\n", __func__); + bit_clk *= 40; + data = AIC3204_AISR1_WL_24BITS; + break; + case SNDRV_PCM_FORMAT_S32_LE: + pr_debug("%s: Sample format 32 bits\n", __func__); + bit_clk *= 48; + data = AIC3204_AISR1_WL_32BITS; + break; + } + snd_soc_update_bits(codec, AIC3204_AISR1, AIC3204_AISR1_WL, data); + + pr_debug("%s: DAC Clock = %d Hz, Bit clock = %d Hz\n", __func__, + dac_clk, bit_clk); + + /* Try to find a value for div which allows us to bypass the PLL and + * generate CODEC_CLK directly. */ + for (this_ndac = 128; this_ndac >= 1; this_ndac--) { + this_clk = aic3204->sysclk / this_ndac; + + if (this_clk == dac_clk) { + pr_debug("%s: Scored exact match. NDAC=%d; " + "DAC_CLK=%d\n", + __func__, this_ndac, this_clk); + bypass_pll = 1; + ndac = this_ndac; + break; + } + } + + if (bypass_pll) { + /* Select MCLK as CODEC_CLKIN */ + aic3204_write(codec, AIC3204_CLK1, AIC3204_CLK1_CODECCLK_MCLK); + codec_clk = aic3204->sysclk; + goto setup_codec; + } + + /* + * Determine PLL Parameters... the input/output frequency ratio + * of the PLL is given in the datasheet: + * + * out R * J * (D/10000) J D + * --- = ----------------- = R * - * ------- + * in P P 10000 + * + * Limitations: + * - R range: 1..4 + * - J range: 4..63 + * - D range: 0..9999 + * - P range: 1..8 + * - If D == 0; + * 512 kHz <= in / P <= 20 MHz + * else + * 10 MHz <= in / P <= 20 MHz + * + * We will try for 8* the target frequency; then use NDAC/NADC to divide + * that down. If we set P = 1, and R = 1; we simplify the solution. + */ + pll_p = 1; + pll_r = 1; + for (this_ndac = 1; this_ndac <= 128; this_ndac++) { + int ratio, init_ratio; + + codec_clk = dac_clk * this_ndac; + + /* Ensure maximum CODEC frequency is not exceeded */ + if (codec_clk > 137000000) + continue; + if ((this_ndac % 2) && (codec_clk > 110000000)) + continue; + + /* Initial Fixed-point ratio ... this will be slightly lower */ + init_ratio = codec_clk / (aic3204->sysclk / 10000); + + pr_debug("%s: NDAC = %d, PLL J.D * 10000 = %d\n", + __func__, this_ndac, init_ratio); + if (init_ratio < 40000) + continue; + if (init_ratio > 639999) + continue; + + for (ratio = init_ratio; (ratio < init_ratio + 10000) + && (ratio < 639999); ratio++) { + j = ratio / 10000; + d = ratio % 10000; + + this_clk = aic3204->sysclk / 10000; + this_clk *= (10000 * j) + d; + this_error = this_clk - codec_clk; + this_abs_error = abs(this_error); + + if (this_abs_error > 1000) + continue; + + pr_debug("%s: Fractional PLL search; " + "P=%d R=%d J=%d D=%04d => " + "(%d * %d) / 10000 => %d " + "(%d error) [NDAC=%d]\n", + __func__, pll_p, pll_r, + j, d, aic3204->sysclk, + ratio, this_clk, this_error, + this_ndac); + if (!last_clk || (this_abs_error < last_error)) { + last_clk = this_clk; + last_error = this_abs_error; + ndac = this_ndac; + pll_j = j; + pll_d = d; + /* Early finish for exact match */ + if (!this_abs_error) + break; + } + } + } + + /* If we still haven't found anything; bail out here! */ + if (!last_clk) + goto fail; + + pr_debug("%s: Best Output: %d (%d Hz Error)\n", + __func__, last_clk, last_error); + pr_debug("%s: setting up PLL: P=%d R=%d J.D=%d.%04d\n", + __func__, pll_p, pll_r, pll_j, pll_d); + + codec_clk = last_clk; + + aic3204_write(codec, AIC3204_CLK4, (pll_d >> AIC3204_CLK4_PLL_D_VSHIFT) + & AIC3204_CLK4_PLL_D); /* PLL D[14:8] */ + aic3204_write(codec, AIC3204_CLK5, pll_d & AIC3204_CLK5_PLL_D); + /* PLL D[7:0] */ + aic3204_write(codec, AIC3204_CLK3, pll_j & AIC3204_CLK3_PLL_J); + /* PLL J */ + aic3204_write(codec, AIC3204_CLK2, + (pll_r & AIC3204_CLK2_PLL_R) /* PLL R */ + | ((pll_p << AIC3204_CLK2_PLL_P_SHIFT) + & AIC3204_CLK2_PLL_P) + /* PLL P */ + | AIC3204_CLK2_PLL_ON); /* PLL on */ + + /* Select PLL as CODEC_CLKIN */ + aic3204_write(codec, AIC3204_CLK1, AIC3204_CLK1_CODECCLK_PLL); + +setup_codec: + pr_debug("%s: CODEC CLK = %d => DAC CLK = %d\n", + __func__, codec_clk, codec_clk / ndac); + /* Calculate BDIV and NDAC */ + mdac = codec_clk / (ndac * dac_clk); + if (!mdac) + mdac = 1; + + bdiv = dac_clk / bit_clk; + if (!bdiv) + bdiv = 1; + + /* Set AOSR and DOSR */ + aic3204_write(codec, AIC3204_DOSR1, + (aic3204->pdata.osr >> AIC3204_DOSR1_MSB_VSHIFT) + & AIC3204_DOSR1_MSB); /* DOSR[9:8] */ + aic3204_write(codec, AIC3204_DOSR2, aic3204->pdata.osr + & AIC3204_DOSR2_LSB); /* DOSR[7:0] */ + aic3204_write(codec, AIC3204_AOSR, aic3204->pdata.osr); /* AOSR */ + + pr_debug("%s: NDAC = %d\n", __func__, ndac); + /* Set NDAC to divider value */ + aic3204_write(codec, AIC3204_CLK6, AIC3204_CLK6_NDAC_STATE_ON | ndac); + /* Set NADC to match NDAC */ + aic3204_write(codec, AIC3204_CLK8, AIC3204_CLK8_NADC_STATE_ON | ndac); + /* Set MDAC and turn on */ + aic3204_write(codec, AIC3204_CLK7, AIC3204_CLK7_MDAC_STATE_ON | mdac); + /* Use MDAC for MADC */ + aic3204_write(codec, AIC3204_CLK9, AIC3204_CLK7_MDAC_STATE_ON | mdac); + + /* Set BCLK and WCLK */ + pr_debug("%s: Bit clock divider = %d\n", __func__, bdiv); + /* Turn on and set BCLK N divider */ + aic3204_write(codec, AIC3204_CLK12, bdiv + | AIC3204_CLK12_BCLK_STATE_ON); + + /* + * Set BCLK and WCLK sources... + * + * Despite DAPM; this is still needed, as DAPM doesn't yet set the + * source at the right time. + * + * TODO: See if we can change the order of initialisation so this + * selection is made after bringing up the ADCs/DACs. Alternative: + * see if we can find out ahead of time which one to use. + */ + snd_soc_update_bits(codec, AIC3204_AISR3, AIC3204_AISR3_BDIV, + (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ? AIC3204_AISR3_BDIV_DAC + : AIC3204_AISR3_BDIV_ADC); + snd_soc_update_bits(codec, AIC3204_AISR6, AIC3204_AISR6_WCLKOUT, + (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ? AIC3204_AISR6_WCLKOUT_DAC + : AIC3204_AISR6_WCLKOUT_ADC); + return 0; +fail: + printk(KERN_ERR "%s(): unable to setup PLL\n", __func__); + return -EINVAL; +} + +static int aic3204_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 dacs2; + + dacs2 = aic3204_read_reg_cache(codec, AIC3204_DACS2); + + if (mute) { + dacs2 |= (1 << AIC3204_DACS2_LEFT_MUTE_SHIFT) + | (1 << AIC3204_DACS2_RIGHT_MUTE_SHIFT); + } else { + dacs2 &= ~((1 << AIC3204_DACS2_LEFT_MUTE_SHIFT) + | (1 << AIC3204_DACS2_RIGHT_MUTE_SHIFT)); + } + + aic3204_write(codec, AIC3204_DACS2, dacs2); /* Unmute DAC */ + return 0; +} + +static int aic3204_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic3204_priv *aic3204 = snd_soc_codec_get_drvdata(codec); + + aic3204->sysclk = freq; + return 0; +} + +static int aic3204_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 aisr1 = 0, aisr2 = 0, aisr3 = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + aisr1 |= AIC3204_AISR1_BCLK_OUT + | AIC3204_AISR1_WCLK_OUT; + break; + case SND_SOC_DAIFMT_CBM_CFS: + aisr1 |= AIC3204_AISR1_BCLK_OUT + | AIC3204_AISR1_WCLK_IN; + break; + case SND_SOC_DAIFMT_CBS_CFM: + aisr1 |= AIC3204_AISR1_BCLK_IN + | AIC3204_AISR1_WCLK_OUT; + break; + case SND_SOC_DAIFMT_CBS_CFS: + aisr1 |= AIC3204_AISR1_BCLK_IN + | AIC3204_AISR1_WCLK_IN; + break; + default: + printk(KERN_ERR "%s: Clock mode not supported " + "(fmt=0x%08x)\n", __func__, fmt); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + aisr1 |= AIC3204_AISR1_INT_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + aisr2 = 1; + case SND_SOC_DAIFMT_DSP_B: + aisr1 |= AIC3204_AISR1_INT_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + aisr1 |= AIC3204_AISR1_INT_RJF; + break; + case SND_SOC_DAIFMT_LEFT_J: + aisr1 |= AIC3204_AISR1_INT_LJF; + break; + default: + printk(KERN_ERR "%s: SSI mode not supported " + "(fmt=0x%08x)\n", __func__, fmt); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + aisr3 |= AIC3204_AISR3_BCLKPOL_NOR; + break; + case SND_SOC_DAIFMT_IB_NF: + aisr3 |= AIC3204_AISR3_BCLKPOL_INV; + break; + default: + printk(KERN_ERR "%s: Clock inversion mode not " + "supported (fmt=0x%08x)\n", __func__, fmt); + return -EINVAL; + } + + /* set iface */ + aic3204_write(codec, AIC3204_AISR1, aisr1); + aic3204_write(codec, AIC3204_AISR2, aisr2); + aic3204_write(codec, AIC3204_AISR3, aisr3); + + return 0; +} + +#define AIC3204_RATES SNDRV_PCM_RATE_8000_96000 +#define AIC3204_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops aic3204_dai_ops = { + .hw_params = aic3204_hw_params, + .digital_mute = aic3204_mute, + .set_sysclk = aic3204_set_dai_sysclk, + .set_fmt = aic3204_set_dai_fmt, +}; + +struct snd_soc_dai aic3204_dai = { + .name = "tlv320aic3204", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AIC3204_RATES, + .formats = AIC3204_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AIC3204_RATES, + .formats = AIC3204_FORMATS,}, + .ops = &aic3204_dai_ops, +}; +EXPORT_SYMBOL_GPL(aic3204_dai); + +static int aic3204_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i; + u8 data[2]; + u8 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < AIC3204_CACHEREGNUM; i++) { + data[0] = i; + data[1] = cache[i]; + codec->hw_write(codec->control_data, data, 2); + } + + return 0; +} + +/* + * initialise the AIC3204 driver + * register the mixer and dsp interfaces with the kernel + */ +static int aic3204_init(struct snd_soc_codec *codec) +{ + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->name = "tlv320aic3204"; + codec->owner = THIS_MODULE; + codec->read = aic3204_read_reg_cache; + codec->write = aic3204_write; + codec->dai = &aic3204_dai; + codec->num_dai = 1; + codec->reg_cache_size = AIC3204_CACHEREGNUM; + codec->reg_cache = kmemdup(aic3204_default_reg_data, + AIC3204_CACHEREGNUM, GFP_KERNEL); + codec->readable_register = aic3204_reg_is_readable; + codec->display_register = aic3204_show_reg; + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* Reset the CODEC */ + aic3204_write(codec, AIC3204_PAGE_SELECT, 0); + aic3204_write(codec, AIC3204_RESET, AIC3204_RESET_SOFT); + + return 0; +} + +static struct snd_soc_codec *aic3204_codec; + +static int aic3204_register(struct snd_soc_codec *codec) +{ + int ret; + + ret = aic3204_init(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to initialise device\n"); + return ret; + } + + aic3204_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret) { + dev_err(codec->dev, "Failed to register codec\n"); + return ret; + } + + ret = snd_soc_register_dai(&aic3204_dai); + if (ret) { + dev_err(codec->dev, "Failed to register dai\n"); + snd_soc_unregister_codec(codec); + return ret; + } + + return 0; +} + +static int aic3204_unregister(struct aic3204_priv *aic3204) +{ + snd_soc_unregister_dai(&aic3204_dai); + snd_soc_unregister_codec(&aic3204->codec); + + if (aic3204->pdata.gpio_reset >= 0) { + gpio_set_value(aic3204->pdata.gpio_reset, 0); + gpio_free(aic3204->pdata.gpio_reset); + } + regulator_bulk_disable(ARRAY_SIZE(aic3204->supplies), + aic3204->supplies); + regulator_bulk_free(ARRAY_SIZE(aic3204->supplies), aic3204->supplies); + + kfree(aic3204); + aic3204_codec = NULL; + + return 0; +} + +static int aic3204_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct snd_soc_codec *codec; + struct aic3204_priv *aic3204; + struct aic3204_pdata *pdata = i2c->dev.platform_data; + int ret, i; + + aic3204 = kzalloc(sizeof(struct aic3204_priv), GFP_KERNEL); + if (aic3204 == NULL) { + dev_err(&i2c->dev, "failed to create private data\n"); + return -ENOMEM; + } + + codec = &aic3204->codec; + codec->dev = &i2c->dev; + snd_soc_codec_set_drvdata(codec, aic3204); + codec->control_data = i2c; + codec->hw_write = (hw_write_t) i2c_master_send; + + i2c_set_clientdata(i2c, aic3204); + + dev_dbg(&i2c->dev, "%s: platform data at %p\n", + __func__, pdata); + if (pdata) + memcpy(&aic3204->pdata, pdata, sizeof(struct aic3204_pdata)); + else + memcpy(&aic3204->pdata, &aic3204_default_pdata, + sizeof(struct aic3204_pdata)); + + if (aic3204->pdata.gpio_reset >= 0) { + ret = gpio_request(aic3204->pdata.gpio_reset, + "tlv320aic3204 reset"); + if (ret != 0) + goto err_gpio; + gpio_direction_output(aic3204->pdata.gpio_reset, 0); + } + + for (i = 0; i < ARRAY_SIZE(aic3204->supplies); i++) + aic3204->supplies[i].supply = aic3204_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(aic3204->supplies), + aic3204->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err_get; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(aic3204->supplies), + aic3204->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_enable; + } + + if (aic3204->pdata.gpio_reset >= 0) { + udelay(1); + gpio_set_value(aic3204->pdata.gpio_reset, 1); + } + + return aic3204_register(codec); + +err_enable: + regulator_bulk_free(ARRAY_SIZE(aic3204->supplies), aic3204->supplies); +err_get: + if (aic3204->pdata.gpio_reset >= 0) + gpio_free(aic3204->pdata.gpio_reset); +err_gpio: + kfree(aic3204); + return ret; +} + +static int aic3204_i2c_remove(struct i2c_client *client) +{ + struct aic3204_priv *aic3204 = i2c_get_clientdata(client); + + return aic3204_unregister(aic3204); +} + +static const struct i2c_device_id aic3204_i2c_id[] = { + { "tlv320aic3204", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, aic3204_i2c_id); + +/* machine i2c codec control layer */ +static struct i2c_driver aic3204_i2c_driver = { + .driver = { + .name = "tlv320aic3204-i2c", + .owner = THIS_MODULE, + }, + .probe = aic3204_i2c_probe, + .remove = aic3204_i2c_remove, + .id_table = aic3204_i2c_id, +}; + +static inline void aic3204_i2c_init(void) +{ + int ret; + + ret = i2c_add_driver(&aic3204_i2c_driver); + if (ret) + printk(KERN_ERR "%s: error regsitering i2c driver, %d\n", + __func__, ret); +} + +static inline void aic3204_i2c_exit(void) +{ + i2c_del_driver(&aic3204_i2c_driver); +} + +static int aic3204_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct aic3204_setup_data *setup; + struct snd_soc_codec *codec; + struct aic3204_priv *aic3204; + int ret = 0; + + codec = aic3204_codec; + if (!codec) { + dev_err(&pdev->dev, "Codec not registered\n"); + return -ENODEV; + } + + socdev->card->codec = codec; + setup = socdev->codec_data; + aic3204 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s: private data at %p\n", + __func__, aic3204); + /* Initialise the CODEC with the given settings */ + /* Analogue Input Quick-Charge Power-Up delay */ + ret = aic3204_write(codec, AIC3204_AIQC, + aic3204->pdata.ain_pwrup_delay); + if (ret) + return ret; + + /* Analogue Reference Power-Up delay */ + aic3204_write(codec, AIC3204_REFPU, + aic3204->pdata.ref_pwrup_delay); + if (ret) + return ret; + + /* AVDD<->DVDD link */ + snd_soc_update_bits(codec, AIC3204_POWER, + AIC3204_POWER_AVDD_DVDD_UNLINK, + aic3204->pdata.avdd_link_en ? + 0 : AIC3204_POWER_AVDD_DVDD_UNLINK); + + /* LDO enable, analogue blocks enable */ + snd_soc_update_bits(codec, AIC3204_LDO, + AIC3204_LDO_ANALOG | AIC3204_LDO_AVDD_UP, + AIC3204_LDO_ANALOG_ENABLED | + (aic3204->pdata.ldo_en + ? AIC3204_LDO_AVDD_UP : 0)); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "aic3204: failed to create pcms\n"); + goto pcm_err; + } + + snd_soc_add_controls(codec, aic3204_snd_controls, + ARRAY_SIZE(aic3204_snd_controls)); + + aic3204_add_widgets(codec); + + return ret; + +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static int aic3204_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + kfree(codec->reg_cache); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_aic3204 = { + .probe = aic3204_probe, + .remove = aic3204_remove, + .resume = aic3204_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_aic3204); + +static int __init aic3204_modinit(void) +{ + aic3204_i2c_init(); + + return 0; +} +module_init(aic3204_modinit); + +static void __exit aic3204_exit(void) +{ + aic3204_i2c_exit(); +} +module_exit(aic3204_exit); + +MODULE_DESCRIPTION("ASoC TLV320AIC3204 codec driver"); +MODULE_AUTHOR("Jacques Electronics Pty Ltd"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic3204.h b/sound/soc/codecs/tlv320aic3204.h new file mode 100644 index 0000000..8f9ee43 --- /dev/null +++ b/sound/soc/codecs/tlv320aic3204.h @@ -0,0 +1,1350 @@ +/* + * ALSA SoC TLV320AIC3204 codec driver + * + * Author: Stuart Longland, <redhatter(a)gentoo.org> + * Copyright: (C) 2010 Jacques Electronics, Pty, Ltd. + * + * Author: Vladimir Barinov, <vbarinov(a)embeddedalley.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source(a)mvista.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _AIC3204_H +#define _AIC3204_H + +/* + * The AIC3204's registers are divided into separate pages, the first page of + * every register is a "page select", which sets the current page being + * accessed. For the purposes of simplicity; we shall consider all registers + * as being addressed by a 16-bit value, consisting of the page number as + * most-significant 9 bits, followed by the register number as given in the + * datasheet. + * + * The register access functions will look after page flipping from here. + */ + +/* + * AIC3204 register space; 128 registers each page, we pages 0..127. It's not + * clear from the datasheet exactly how many pages are in use... and it's a + * sparse register map! Ideally, I'd like to just cache the pages that are + * needed, but the ALSA framework doesn't allow this easily. + * + * It does look as if up to and including page 70 is needed. For now, we will + * cache up to page 128 however; as this ensures we get *everything*. + */ +#define AIC3204_CACHEREGNUM (128*128) + +/* Page select register */ +#define AIC3204_PAGE_SELECT 0 + +/* Register Page/Number => Address Mapping Macro */ +#define AIC3204_PGREG(page, reg) (((page) << 7) | ((reg) & 0x7f)) + +/* Software reset register */ +#define AIC3204_RESET AIC3204_PGREG(0, 1) +/* Perform software reset */ +#define AIC3204_RESET_SOFT 1 + +/* Clock Setting Register 1, Multiplexers */ +#define AIC3204_CLK1 AIC3204_PGREG(0, 4) +/* PLL Range Select: Low Range */ +#define AIC3204_CLK1_PLLRANGE_LOW (0 << 6) +/* PLL Range Select: High Range */ +#define AIC3204_CLK1_PLLRANGE_HIGH (1 << 6) +/* PLL Range Select Mask */ +#define AIC3204_CLK1_PLLRANGE (1 << 6) +/* PLL Input Clock Select: MCLK */ +#define AIC3204_CLK1_PLLSRC_MCLK (0 << 2) +/* PLL Input Clock Select: BCLK */ +#define AIC3204_CLK1_PLLSRC_BCLK (1 << 2) +/* PLL Input Clock Select: GPIO */ +#define AIC3204_CLK1_PLLSRC_GPIO (2 << 2) +/* PLL Input Clock Select: Data In */ +#define AIC3204_CLK1_PLLSRC_DIN (3 << 2) +/* PLL Input Clock Select Mask */ +#define AIC3204_CLK1_PLLSRC (3 << 2) +/* CODEC Clock Source Select: MCLK */ +#define AIC3204_CLK1_CODECCLK_MCLK (0) +/* CODEC Clock Source Select: BCLK */ +#define AIC3204_CLK1_CODECCLK_BCLK (1) +/* CODEC Clock Source Select: GPIO */ +#define AIC3204_CLK1_CODECCLK_GPIO (2) +/* CODEC Clock Source Select: PLL */ +#define AIC3204_CLK1_CODECCLK_PLL (3) +/* CODEC Clock Source Select Mask */ +#define AIC3204_CLK1_CODECCLK (3) + +/* Clock Setting Register 2, PLL P & R Values */ +#define AIC3204_CLK2 AIC3204_PGREG(0, 5) +/* PLL State: Powered Down */ +#define AIC3204_CLK2_PLL_OFF (0 << 7) +/* PLL State: Powered Up */ +#define AIC3204_CLK2_PLL_ON (1 << 7) +/* PLL State Mask */ +#define AIC3204_CLK2_PLL (1 << 7) +/* PLL State bitfield location */ +#define AIC3204_CLK2_PLL_SHIFT (7) +/* PLL P Divider value bitfield location */ +#define AIC3204_CLK2_PLL_P_SHIFT (4) +/* PLL P Divider value bitfield mask */ +#define AIC3204_CLK2_PLL_P (7 << 4) +/* PLL R Divider value bitfield location */ +#define AIC3204_CLK2_PLL_R_SHIFT (0) +/* PLL R Divider value bitfield mask */ +#define AIC3204_CLK2_PLL_R (0xf) + +/* Clock Setting Register 3, PLL J Values */ +#define AIC3204_CLK3 AIC3204_PGREG(0, 6) +/* PLL J Divider value bitfield location */ +#define AIC3204_CLK3_PLL_J_SHIFT (0) +/* PLL J Divider value bitfield mask */ +#define AIC3204_CLK3_PLL_J (0x3f) + +/* Clock Setting Register 4, PLL D MSB */ +#define AIC3204_CLK4 AIC3204_PGREG(0, 7) +/* PLL D most significant portion bitfield location */ +#define AIC3204_CLK4_PLL_D_SHIFT (0) +/* PLL D most significant portion bitfield mask */ +#define AIC3204_CLK4_PLL_D (0x3f) +/* PLL D most significant portion value shift */ +#define AIC3204_CLK4_PLL_D_VSHIFT (8) + +/* Clock Setting Register 5, PLL D LSB */ +#define AIC3204_CLK5 AIC3204_PGREG(0, 8) +/* PLL D least significant portion bitfield location */ +#define AIC3204_CLK5_PLL_D_SHIFT (0) +/* PLL D least significant portion bitfield mask */ +#define AIC3204_CLK5_PLL_D (0xff) +/* PLL D least significant portion value shift */ +#define AIC3204_CLK5_PLL_D_VSHIFT (0) + +/* Clock Setting Register 6, NDAC Values */ +#define AIC3204_CLK6 AIC3204_PGREG(0, 11) +/* NDAC State: Powered Down */ +#define AIC3204_CLK6_NDAC_STATE_OFF (0 << 7) +/* NDAC State: Powered Up */ +#define AIC3204_CLK6_NDAC_STATE_ON (1 << 7) +/* NDAC State Mask */ +#define AIC3204_CLK6_NDAC_STATE (1 << 7) +/* NDAC State bitfield location */ +#define AIC3204_CLK6_NDAC_STATE_SHIFT (7) +/* NDAC value bitfield location */ +#define AIC3204_CLK6_NDAC_SHIFT (0) +/* NDAC value bitfield mask */ +#define AIC3204_CLK6_NDAC (0x7f) + +/* Clock Setting Register 7, MDAC Values */ +#define AIC3204_CLK7 AIC3204_PGREG(0, 12) +/* MDAC State: Powered Down */ +#define AIC3204_CLK7_MDAC_STATE_OFF (0 << 7) +/* MDAC State: Powered Up */ +#define AIC3204_CLK7_MDAC_STATE_ON (1 << 7) +/* MDAC State Mask */ +#define AIC3204_CLK7_MDAC_STATE (1 << 7) +/* MDAC State bitfield location */ +#define AIC3204_CLK7_MDAC_STATE_SHIFT (7) +/* MDAC value bitfield location */ +#define AIC3204_CLK7_MDAC_SHIFT (0) +/* MDAC value bitfield mask */ +#define AIC3204_CLK7_MDAC (0x7f) + +/* DAC OSR Setting Register 1, MSB */ +#define AIC3204_DOSR1 AIC3204_PGREG(0, 13) +/* DOSR most significant portion bitfield location */ +#define AIC3204_DOSR1_MSB_SHIFT (0) +/* DOSR most significant portion bitfield mask */ +#define AIC3204_DOSR1_MSB (3) +/* DOSR most significant portion value shift */ +#define AIC3204_DOSR1_MSB_VSHIFT (8) + +/* DAC OSR Setting Register 2, LSB */ +#define AIC3204_DOSR2 AIC3204_PGREG(0, 14) +/* DOSR least significant portion bitfield location */ +#define AIC3204_DOSR2_LSB_SHIFT (0) +/* DOSR least significant portion bitfield mask */ +#define AIC3204_DOSR2_LSB (0xff) +/* DOSR least significant portion value shift */ +#define AIC3204_DOSR2_LSB_VSHIFT (0) + +/* Clock Setting Register 8, NADC Values */ +#define AIC3204_CLK8 AIC3204_PGREG(0, 18) +/* NADC State: Powered Down */ +#define AIC3204_CLK8_NADC_STATE_OFF (0 << 7) +/* NADC State: Powered Up */ +#define AIC3204_CLK8_NADC_STATE_ON (1 << 7) +/* NADC State Mask */ +#define AIC3204_CLK8_NADC_STATE (1 << 7) +/* NADC State bitfield location */ +#define AIC3204_CLK8_NADC_STATE_SHIFT (7) +/* NADC value bitfield location */ +#define AIC3204_CLK8_NADC_SHIFT (0) +/* NADC value bitfield mask */ +#define AIC3204_CLK8_NADC (0x7f) + +/* Clock Setting Register 9, MADC Values */ +#define AIC3204_CLK9 AIC3204_PGREG(0, 19) +/* MADC State: Powered Down */ +#define AIC3204_CLK9_MADC_STATE_OFF (0 << 7) +/* MADC State: Powered Up */ +#define AIC3204_CLK9_MADC_STATE_ON (1 << 7) +/* MADC State Mask */ +#define AIC3204_CLK9_MADC_STATE (1 << 7) +/* MADC State bitfield location */ +#define AIC3204_CLK9_MADC_STATE_SHIFT (7) +/* MADC value bitfield location */ +#define AIC3204_CLK9_MADC_SHIFT (0) +/* MADC value bitfield mask */ +#define AIC3204_CLK9_MADC (0x7f) + +/* ADC OSR Setting Register */ +#define AIC3204_AOSR AIC3204_PGREG(0, 20) + +/* + * Clock Register setting 10, Multiplexers + * + * XXX: Note that the datasheet goofs up here, calling this register 9, but we + * already had a register 9 just before AOSR. Therefore, we will call this + * register 10, and will increment all following by one to correct TI's + * mistake. + */ +#define AIC3204_CLK10 AIC3204_PGREG(0, 25) +/* CDIV Clock Source: MCLK */ +#define AIC3204_CLK10_CDIV_MCLK (0) +/* CDIV Clock Source: BCLK */ +#define AIC3204_CLK10_CDIV_BCLK (1) +/* CDIV Clock Source: Data IN */ +#define AIC3204_CLK10_CDIV_DIN (2) +/* CDIV Clock Source: PLL */ +#define AIC3204_CLK10_CDIV_PLL (3) +/* CDIV Clock Source: DAC */ +#define AIC3204_CLK10_CDIV_DAC (4) +/* CDIV Clock Source: DAC Modulation */ +#define AIC3204_CLK10_CDIV_DAC_MOD (5) +/* CDIV Clock Source: ADC */ +#define AIC3204_CLK10_CDIV_ADC (6) +/* CDIV Clock Source: ADC Modulation */ +#define AIC3204_CLK10_CDIV_ADC_MOD (7) +/* CDIV Clock Source Mask */ +#define AIC3204_CLK10_CDIV (7) + +/* Clock Register setting 11, CLKOUT M divider value */ +#define AIC3204_CLK11 AIC3204_PGREG(0, 26) +/* CLKOUT Divider State: Off */ +#define AIC3204_CLK11_CLKOUT_OFF (0 << 7) +/* CLKOUT Divider State: On */ +#define AIC3204_CLK11_CLKOUT_ON (1 << 7) +/* CLKOUT Divider State Mask */ +#define AIC3204_CLK11_CLKOUT (1 << 7) +/* CLKOUT Divider M value bitfield location */ +#define AIC3204_CLK11_CLKOUTM_SHIFT (0) +/* CLKOUT Divider M value bitfield mask */ +#define AIC3204_CLK11_CLKOUTM (0x7f) + +/* Audio Interface Setting Register 1 */ +#define AIC3204_AISR1 AIC3204_PGREG(0, 27) +/* Audio Interface Select: I2S */ +#define AIC3204_AISR1_INT_I2S (0 << 6) +/* Audio Interface Select: DSP */ +#define AIC3204_AISR1_INT_DSP (1 << 6) +/* Audio Interface Select: Right-Justified Format */ +#define AIC3204_AISR1_INT_RJF (2 << 6) +/* Audio Interface Select: Left-Justified Format */ +#define AIC3204_AISR1_INT_LJF (3 << 6) +/* Audio Interface Select Mask */ +#define AIC3204_AISR1_INT (3 << 6) +/* Audio Data Word Length: 16 bits */ +#define AIC3204_AISR1_WL_16BITS (0 << 4) +/* Audio Data Word Length: 20 bits */ +#define AIC3204_AISR1_WL_20BITS (1 << 4) +/* Audio Data Word Length: 24 bits */ +#define AIC3204_AISR1_WL_24BITS (2 << 4) +/* Audio Data Word Length: 32 bits */ +#define AIC3204_AISR1_WL_32BITS (3 << 4) +/* Audio Data Word Length Mask */ +#define AIC3204_AISR1_WL (3 << 4) +/* BCLK Direction Control: Input */ +#define AIC3204_AISR1_BCLK_IN (0 << 3) +/* BCLK Direction Control: Output */ +#define AIC3204_AISR1_BCLK_OUT (1 << 3) +/* BCLK Direction Control: Mask */ +#define AIC3204_AISR1_BCLK (1 << 3) +/* WCLK Direction Control: Input */ +#define AIC3204_AISR1_WCLK_IN (0 << 2) +/* WCLK Direction Control: Output */ +#define AIC3204_AISR1_WCLK_OUT (1 << 2) +/* WCLK Direction Control: Mask */ +#define AIC3204_AISR1_WCLK (1 << 2) +/* DOUT High Impedance Output Control: Never high impedance */ +#define AIC3204_AISR1_HIZ_NEVER (0 << 0) +/* DOUT High Impedance Output Control: High impedance when idle */ +#define AIC3204_AISR1_HIZ_IDLE (1 << 0) +/* DOUT High Impedance Output Control Mask */ +#define AIC3204_AISR1_HIZ (1 << 0) + +/* Audio Interface Register 2, Data offset setting */ +#define AIC3204_AISR2 AIC3204_PGREG(0, 28) + +/* Audio Interface Register 3 */ +#define AIC3204_AISR3 AIC3204_PGREG(0, 29) +/* Audio Data Loopback Control: Disabled */ +#define AIC3204_AISR3_ADLO_OFF (0 << 5) +/* Audio Data Loopback Control: Enabled */ +#define AIC3204_AISR3_ADLO_ON (1 << 5) +/* Audio Data Loopback Control Mask */ +#define AIC3204_AISR3_ADLO (1 << 5) +/* ADC->DAC Loopback Control: Disabled */ +#define AIC3204_AISR3_ADDALO_OFF (0 << 4) +/* ADC->DAC Loopback Control: Enabled */ +#define AIC3204_AISR3_ADDALO_ON (1 << 4) +/* ADC->DAC Loopback Control Mask */ +#define AIC3204_AISR3_ADDALO (1 << 4) +/* Audio Bit Clock Polarity: Normal */ +#define AIC3204_AISR3_BCLKPOL_NOR (0 << 3) +/* Audio Bit Clock Polarity: Inverted */ +#define AIC3204_AISR3_BCLKPOL_INV (1 << 3) +/* Audio Bit Clock Polarity Mask */ +#define AIC3204_AISR3_BCLKPOL (1 << 3) +/* Audio Data Interface Clock Buffers: Always powered */ +#define AIC3204_AISR3_ADICLK_ALWAYS (0 << 2) +/* Audio Data Interface Clock Buffers: Powered with CODEC only */ +#define AIC3204_AISR3_ADICLK_CODEC (1 << 2) +/* Audio Data Interface Clock Buffers' State */ +#define AIC3204_AISR3_ADICLK (1 << 2) +/* Audio Bit Clock Divider Source: DAC */ +#define AIC3204_AISR3_BDIV_DAC (0 << 0) +/* Audio Bit Clock Divider Source: DAC Modulation */ +#define AIC3204_AISR3_BDIV_DAC_MOD (1 << 0) +/* Audio Bit Clock Divider Source: ADC */ +#define AIC3204_AISR3_BDIV_ADC (2 << 0) +/* Audio Bit Clock Divider Source: ADC Modulation */ +#define AIC3204_AISR3_BDIV_ADC_MOD (3 << 0) +/* Audio Bit Clock Divider Source Mask */ +#define AIC3204_AISR3_BDIV (3 << 0) + +/* Clock Setting Register 12, BCLK N Divider */ +#define AIC3204_CLK12 AIC3204_PGREG(0, 30) +/* BCLK N State: Powered Down */ +#define AIC3204_PG0_CLK12_BCLK_STATE_OFF (0 << 7) +/* BCLK N State: Powered Up */ +#define AIC3204_CLK12_BCLK_STATE_ON (1 << 7) +/* BCLK N State Mask */ +#define AIC3204_CLK12_BCLK_STATE (1 << 7) +/* BCLK N State bitfield location */ +#define AIC3204_CLK12_BCLK_STATE_SHIFT (7) +/* BCLK N value bitfield location */ +#define AIC3204_CLK12_BCLK_SHIFT (0) +/* BCLK N value bitfield mask */ +#define AIC3204_CLK12_BCLK (0x7f) + +/* Audio Interface Setting Register 4, Secondary Audio Interface */ +#define AIC3204_AISR4 AIC3204_PGREG(0, 31) +/* Secondary Bit Clock: GPIO */ +#define AIC3204_AISR4_SECBCLK_GPIO (0 << 5) +/* Secondary Bit Clock: SCLK */ +#define AIC3204_AISR4_SECBCLK_SCLK (1 << 5) +/* Secondary Bit Clock: MISO */ +#define AIC3204_AISR4_SECBCLK_MISO (2 << 5) +/* Secondary Bit Clock: DOUT */ +#define AIC3204_AISR4_SECBCLK_DOUT (3 << 5) +/* Secondary Bit Clock Mask */ +#define AIC3204_AISR4_SECBCLK (3 << 5) +/* Secondary Word Clock: GPIO */ +#define AIC3204_AISR4_SECWCLK_GPIO (0 << 3) +/* Secondary Word Clock: SCLK */ +#define AIC3204_AISR4_SECWCLK_SCLK (1 << 3) +/* Secondary Word Clock: MISO */ +#define AIC3204_AISR4_SECWCLK_MISO (2 << 3) +/* Secondary Word Clock: DOUT */ +#define AIC3204_AISR4_SECWCLK_DOUT (3 << 3) +/* Secondary Word Clock Mask */ +#define AIC3204_AISR4_SECWCLK (3 << 3) +/* ADC Word Clock: GPIO */ +#define AIC3204_AISR4_ADCWCLK_GPIO (0 << 3) +/* ADC Word Clock: SCLK */ +#define AIC3204_AISR4_ADCWCLK_SCLK (1 << 3) +/* ADC Word Clock: MISO */ +#define AIC3204_AISR4_ADCWCLK_MISO (2 << 3) +/* ADC Word Clock Mask */ +#define AIC3204_AISR4_ADCWCLK (3 << 3) +/* Secondary Data Input: GPIO */ +#define AIC3204_AISR4_SECDIN_GPIO (0 << 0) +/* Secondary Data Input: SCLK */ +#define AIC3204_AISR4_SECDIN_SCLK (1 << 0) +/* Secondary Data Input Mask */ +#define AIC3204_AISR4_SECDIN (1 << 0) + +/* Audio Interface Setting Register 5 */ +#define AIC3204_AISR5 AIC3204_PGREG(0, 32) +/* Audio Interface Bit Clock: Primary (BCLK) */ +#define AIC3204_AISR5_BCLKIN_PRI (0 << 3) +/* Audio Interface Bit Clock: Secondary */ +#define AIC3204_AISR5_BCLKIN_SEC (1 << 3) +/* Audio Interface Bit Clock Mask */ +#define AIC3204_AISR5_WCLKIN (1 << 2) +/* Audio Interface Word Clock: Primary (BCLK) */ +#define AIC3204_AISR5_WCLKIN_PRI (0 << 2) +/* Audio Interface Word Clock: Secondary */ +#define AIC3204_AISR5_WCLKIN_SEC (1 << 2) +/* Audio Interface Word Clock Mask */ +#define AIC3204_AISR5_WCLKIN (1 << 2) +/* ADC Word Clock Control: ADC WCLK = DAC WCLK */ +#define ADC3204_PG0_AISR5_ADCWCLK_DAC (0 << 1) +/* ADC Word Clock Control: ADC WCLK = Secondary ADC WCLK */ +#define ADC3204_PG0_AISR5_ADCWCLK_SEC (1 << 1) +/* ADC Word Clock Control Mask */ +#define ADC3204_PG0_AISR5_ADCWCLK (1 << 1) +/* Audio Data In: Primary Data In */ +#define ADC3204_PG0_AISR5_DIN_PRI (0 << 0) +/* Audio Data In: Secondary Data In */ +#define ADC3204_PG0_AISR5_DIN_SEC (1 << 0) +/* Audio Data In Mask */ +#define ADC3204_PG0_AISR5_DIN (1 << 0) + +/* Audio Interface Setting Register 6 */ +#define AIC3204_AISR6 AIC3204_PGREG(0, 33) +/* BCLK Output Control: Generated Primary Bit Clock */ +#define AIC3204_AISR6_BCLKOUT_GEN (0 << 7) +/* BCLK Output Control: Secondary Bit Clock */ +#define AIC3204_AISR6_BCLKOUT_SEC (1 << 7) +/* BCLK Output Control Mask */ +#define AIC3204_AISR6_BCLKOUT (1 << 7) +/* Secondary BCLK Output Control: Primary Bit Clock Input */ +#define AIC3204_AISR6_SBCLKOUT_BCLK (0 << 6) +/* Secondary BCLK Output Control: Generated Primary Bit Clock */ +#define AIC3204_AISR6_SBCLKOUT_GEN (1 << 6) +/* Secondary BCLK Output Control Mask */ +#define AIC3204_AISR6_SBCLKOUT (1 << 6) +/* WCLK Output Control: Generated DAC_FS */ +#define AIC3204_AISR6_WCLKOUT_DAC (0 << 4) +/* WCLK Output Control: Generated ADC_FS */ +#define AIC3204_AISR6_WCLKOUT_ADC (1 << 4) +/* WCLK Output Control: Secondary WCLK Input */ +#define AIC3204_AISR6_WCLKOUT_SWCLK (2 << 4) +/* WCLK Output Control Mask */ +#define AIC3204_AISR6_WCLKOUT (3 << 4) +/* Secondary WCLK Output Control: WCLK Input */ +#define AIC3204_AISR6_SWCLKOUT_WCLK (0 << 2) +/* Secondary WCLK Output Control: Generated DAC_FS */ +#define AIC3204_AISR6_SWCLKOUT_DAC (1 << 2) +/* Secondary WCLK Output Control: Generated ADC_FS */ +#define AIC3204_AISR6_SWCLKOUT_ADC (2 << 2) +/* Secondary WCLK Output Control Mask */ +#define AIC3204_AISR6_SWCLKOUT (3 << 2) +/* Primary Data Output Control: Serial Interface */ +#define AIC3204_AISR6_DOUT_INT (0 << 1) +/* Primary Data Output Control: Secondary Data Input */ +#define AIC3204_AISR6_DOUT_SDIN (1 << 1) +/* Primary Data Output Control Mask */ +#define AIC3204_AISR6_DOUT (1 << 1) +/* Secondary Data Output Control: Primary Data In */ +#define AIC3204_AISR6_SDOUT_DIN (0 << 0) +/* Secondary Data Output Control: Serial Interface */ +#define AIC3204_AISR6_SDOUT_INT (1 << 0) +/* Secondary Data Output Control Mask */ +#define AIC3204_AISR6_SDOUT (1 << 0) + +/* Digital Interface Misc. Setting Register */ +#define AIC3204_DIMISC AIC3204_PGREG(0, 34) +/* I2C General Call Address Configuration: Ignore */ +#define AIC3204_DIMISC_I2CGC_IGNORE (0 << 5) +/* I2C General Call Address Configuration: Accept */ +#define AIC3204_DIMISC_I2CGC_ACCEPT (1 << 5) + +/* ADC Flag Register */ +#define AIC3204_ADCF AIC3204_PGREG(0, 36) +/* Left ADC PGA Status: Gain is set */ +#define AIC3204_ADCF_LEFT_PGASET (1 << 7) +/* Left ADC Power Status: Powered Up */ +#define AIC3204_ADCF_LEFT_UP (1 << 6) +/* Left ADC AGC Status: Gain is saturated */ +#define AIC3204_ADCF_LEFT_AGCSAT (1 << 5) +/* Right ADC PGA Status: Gain is set */ +#define AIC3204_ADCF_RIGHT_PGASET (1 << 3) +/* Right ADC Power Status: Powered Up */ +#define AIC3204_ADCF_RIGHT_UP (1 << 2) +/* Right ADC AGC Status: Gain is saturated */ +#define AIC3204_ADCF_RIGHT_AGCSAT (1 << 1) + +/* DAC Flag Register 1 */ +#define AIC3204_DACF1 AIC3204_PGREG(0, 37) +/* Left DAC powered up */ +#define AIC3204_DACF1_LEFT_UP (1 << 7) +/* Left Line Output Driver powered up */ +#define AIC3204_DACF1_LOL_UP (1 << 6) +/* Left Headphone Output Driver powered up */ +#define AIC3204_DACF1_HPL_UP (1 << 5) +/* Right DAC powered up */ +#define AIC3204_DACF1_RIGHT_UP (1 << 7) +/* Right Line Output Driver powered up */ +#define AIC3204_DACF1_LOR_UP (1 << 6) +/* Right Headphone Output Driver powered up */ +#define AIC3204_DACF1_HPR_UP (1 << 5) + +/* DAC Flag Register 2 */ +#define AIC3204_DACF2 AIC3204_PGREG(0, 38) +/* Left DAC PGA Status: Gain is set */ +#define AIC3204_DACF2_LEFT_PGASET (1 << 4) +/* Right DAC PGA Status: Gain is set */ +#define AIC3204_DACF2_RIGHT_PGASET (1 << 0) + +/* Sticky Flag Register 1 */ +#define AIC3204_STICK1 AIC3204_PGREG(0, 42) +/* Left DAC Overflow */ +#define AIC3204_STICK1_LDAC_OVER (1 << 7) +/* Right DAC Overflow */ +#define AIC3204_STICK1_RDAC_OVER (1 << 6) +/* Left ADC Overflow */ +#define AIC3204_STICK1_LADC_OVER (1 << 3) +/* Right ADC Overflow */ +#define AIC3204_STICK1_RADC_OVER (1 << 2) + +/* Interrupt Flag Register 1 */ +#define AIC3204_INTF1 AIC3204_PGREG(0, 43) +/* Left DAC Overflow */ +#define AIC3204_INTF1_LDAC_OVER (1 << 7) +/* Right DAC Overflow */ +#define AIC3204_INTF1_RDAC_OVER (1 << 6) +/* Left ADC Overflow */ +#define AIC3204_INTF1_LADC_OVER (1 << 3) +/* Right ADC Overflow */ +#define AIC3204_INTF1_RADC_OVER (1 << 2) + +/* Sticky Flag Register 2 */ +#define AIC3204_STICK2 AIC3204_PGREG(0, 44) +/* Left Headphone Driver Over Current */ +#define AIC3204_STICK2_HPL_OVER (1 << 7) +/* Right Headphone Driver Over Current */ +#define AIC3204_STICK2_HPR_OVER (1 << 6) +/* Headset button pressed */ +#define AIC3204_STICK2_HS_BUTTON (1 << 5) +/* Headset plug inserted/removed */ +#define AIC3204_STICK2_HS_PLUGGED (1 << 4) +/* Left Channel DRC: Over threshold */ +#define AIC3204_STICK2_LDRC_OVER (1 << 3) +/* Right Channel DRC: Over threshold */ +#define AIC3204_STICK2_RDRC_OVER (1 << 2) + +/* Sticky Flag Register 3 */ +#define AIC3204_STICK3 AIC3204_PGREG(0, 45) +/* Left AGC Noise Threshold Flag: Over threshold */ +#define AIC3204_STICK3_LAGC_OVER (1 << 6) +/* Right AGC Noise Threshold Flag: Over threshold */ +#define AIC3204_STICK3_RAGC_OVER (1 << 5) +/* Left ADC DC Measurement Available */ +#define AIC3204_STICK3_LADC_DC (1 << 2) +/* Right ADC DC Measurement Available */ +#define AIC3204_STICK3_RADC_DC (1 << 1) + +/* Interrupt Flag Register 2 */ +#define AIC3204_INTF2 AIC3204_PGREG(0, 46) +/* Left Headphone Driver Over Current */ +#define AIC3204_INTF2_HPL_OVER (1 << 7) +/* Right Headphone Driver Over Current */ +#define AIC3204_INTF2_HPR_OVER (1 << 6) +/* Headset button pressed */ +#define AIC3204_INTF2_HS_BUTTON (1 << 5) +/* Headset plug inserted/removed */ +#define AIC3204_INTF2_HS_PLUGGED (1 << 4) +/* Left Channel DRC: Over threshold */ +#define AIC3204_INTF2_LDRC_OVER (1 << 3) +/* Right Channel DRC: Over threshold */ +#define AIC3204_INTF2_RDRC_OVER (1 << 2) + +/* Interrupt Flag Register 3 */ +#define AIC3204_INTF3 AIC3204_PGREG(0, 47) +/* Left AGC Noise Threshold Flag: Over threshold */ +#define AIC3204_INTF3_LAGC_OVER (1 << 6) +/* Right AGC Noise Threshold Flag: Over threshold */ +#define AIC3204_INTF3_RAGC_OVER (1 << 5) +/* Left ADC DC Measurement Available */ +#define AIC3204_INTF3_LADC_DC (1 << 2) +/* Right ADC DC Measurement Available */ +#define AIC3204_INTF3_RADC_DC (1 << 1) + +/* INT1 Interrupt Control Register */ +#define AIC3204_INT1 AIC3204_PGREG(0, 48) +/* INT1 Generated on Headset insertion */ +#define AIC3204_INT1_HS_PLUG (1 << 7) +/* INT1 Generated on Headset Button press */ +#define AIC3204_INT1_HS_BUTTON (1 << 6) +/* INT1 Generated on DAC DRC Signal Threshold */ +#define AIC3204_INT1_DAC_DRC (1 << 5) +/* INT1 Generated on AGC Noise Interrupt */ +#define AIC3204_INT1_ADC_NOISE (1 << 4) +/* INT1 Generated on Over Current */ +#define AIC3204_INT1_HP_OVERCURRENT (1 << 3) +/* INT1 Generated on overflow event */ +#define AIC3204_INT1_OVERFLOW (1 << 2) +/* INT1 Generated on DC measurement */ +#define AIC3204_INT1_DC (1 << 1) +/* INT1 pulse control: Continuous pulse train */ +#define AIC3204_INT1_CONT_PULSE (1 << 0) + +/* INT2 Interrupt Control Register */ +#define AIC3204_INT2 AIC3204_PGREG(0, 49) +/* INT2 Generated on Headset insertion */ +#define AIC3204_INT2_HS_PLUG (1 << 7) +/* INT2 Generated on Headset Button press */ +#define AIC3204_INT2_HS_BUTTON (1 << 6) +/* INT2 Generated on DAC DRC Signal Threshold */ +#define AIC3204_INT2_DAC_DRC (1 << 5) +/* INT2 Generated on AGC Noise Interrupt */ +#define AIC3204_INT2_ADC_NOISE (1 << 4) +/* INT2 Generated on Over Current */ +#define AIC3204_INT2_HP_OVERCURRENT (1 << 3) +/* INT2 Generated on overflow event */ +#define AIC3204_INT2_OVERFLOW (1 << 2) +/* INT2 Generated on DC measurement */ +#define AIC3204_INT2_DC (1 << 1) +/* INT2 pulse control: Continuous pulse train */ +#define AIC3204_INT2_CONT_PULSE (1 << 0) + +/* GPIO/MFP5 Control Register */ +#define AIC3204_MFP5 AIC3204_PGREG(0, 52) +/* GPIO Control: Disabled */ +#define AIC3204_MFP5_FUNC_DISABLED (0 << 2) +/* GPIO Control: Secondary audio interface/digital microphone/clock input */ +#define AIC3204_MFP5_FUNC_SAI_DM_CI (1 << 2) +/* GPIO Control: General Purpose Input */ +#define AIC3204_MFP5_FUNC_INPUT (2 << 2) +/* GPIO Control: General Purpose Output */ +#define AIC3204_MFP5_FUNC_OUTPUT (3 << 2) +/* GPIO Control: CLKOUT Output */ +#define AIC3204_MFP5_FUNC_CLKOUT (4 << 2) +/* GPIO Control: INT1 Output */ +#define AIC3204_MFP5_FUNC_INT1 (4 << 2) +/* GPIO Control: INT2 Output */ +#define AIC3204_MFP5_FUNC_INT2 (4 << 2) +/* GPIO Control: ADC_WCLK */ +#define AIC3204_MFP5_FUNC_ADC_WCLK (4 << 2) +/* GPIO Control: Secondary Bit Clock */ +#define AIC3204_MFP5_FUNC_SEC_BCLK (4 << 2) +/* GPIO Control: Secondary Word Clock */ +#define AIC3204_MFP5_FUNC_SEC_WCLK (4 << 2) +/* GPIO Control: Digital Microphone Clock */ +#define AIC3204_MFP5_FUNC_DMIC_CLK (4 << 2) +/* GPIO Control Mask */ +#define AIC3204_MFP5_FUNC (15 << 2) +/* GPIO Input State */ +#define AIC3204_MFP5_IN (1 << 1) +/* GPIO Output State */ +#define AIC3204_MFP5_OUT (1 << 0) + +/* DOUT/MFP2 Function Control Register */ +#define AIC3204_MFP2 AIC3204_PGREG(0, 53) +/* DOUT Bus Keeper Enabled */ +#define AIC3204_MFP2_BK (1 << 4) +/* DOUT MUX Control: Disabled */ +#define AIC3204_MFP2_FUNC_DISABLED (0 << 1) +/* DOUT MUX Control: Primary DOUT */ +#define AIC3204_MFP2_FUNC_PRI_DOUT (1 << 1) +/* DOUT MUX Control: General Purpose Output */ +#define AIC3204_MFP2_FUNC_OUTPUT (2 << 1) +/* DOUT MUX Control: CLKOUT Clock Output */ +#define AIC3204_MFP2_FUNC_CLKOUT (3 << 1) +/* DOUT MUX Control: INT1 Output */ +#define AIC3204_MFP2_FUNC_INT1 (4 << 1) +/* DOUT MUX Control: INT2 Output */ +#define AIC3204_MFP2_FUNC_INT2 (5 << 1) +/* DOUT MUX Control: Secondary Bit Clock */ +#define AIC3204_MFP2_FUNC_SEC_BCLK (6 << 1) +/* DOUT MUX Control: Secondary Word Clock */ +#define AIC3204_MFP2_FUNC_SEC_WCLK (7 << 1) +/* DOUT MUX Control Mask */ +#define AIC3204_MFP2_FUNC (7 << 1) +/* DOUT General Purpose Output State */ +#define AIC3204_MFP2_OUT (1 << 0) + +/* DIN/MFP1 Function Control Register */ +#define AIC3204_MFP1 AIC3204_PGREG(0, 54) +/* DIN Function Control: Disabled */ +#define AIC3204_MFP1_FUNC_DISABLED (0 << 1) +/* DIN Function Control: Primary Data Input/Digital Microphone/Clock Input */ +#define AIC3204_MFP1_FUNC_DIN_DM_CI (1 << 1) +/* DIN Function Control: General Purpose Input */ +#define AIC3204_MFP1_FUNC_INPUT (2 << 1) +/* DIN Function Control Mask */ +#define AIC3204_MFP1_FUNC (3 << 1) +/* DIN Input State */ +#define AIC3204_MFP1_IN (1 << 0) + +/* MISO/MFP4 Function Control Register */ +#define AIC3204_MFP4 AIC3204_PGREG(0, 55) +/* MISO Function Control: Disabled */ +#define AIC3204_MFP4_FUNC_DISABLED (0 << 1) +/* MISO Function Control: SPI Data Output (disabled in I2C mode) */ +#define AIC3204_MFP4_FUNC_SPI_OUT (1 << 1) +/* MISO Function Control: General Purpose Output */ +#define AIC3204_MFP4_FUNC_OUTPUT (2 << 1) +/* MISO Function Control: CLKOUT Clock Output */ +#define AIC3204_MFP4_FUNC_CLKOUT (3 << 1) +/* MISO Function Control: INT1 Output */ +#define AIC3204_MFP4_FUNC_INT1 (4 << 1) +/* MISO Function Control: INT2 Output */ +#define AIC3204_MFP4_FUNC_INT2 (5 << 1) +/* MISO Function Control: ADC Word Clock Output */ +#define AIC3204_MFP4_FUNC_ADC_WCLK (6 << 1) +/* MISO Function Control: Digital Microphone Clock Output */ +#define AIC3204_MFP4_FUNC_DMIC_CLK (7 << 1) +/* MISO Function Control: Secondary Data Output */ +#define AIC3204_MFP4_FUNC_SEC_DOUT (8 << 1) +/* MISO Function Control: Secondary Bit Clock */ +#define AIC3204_MFP4_FUNC_SEC_BCLK (9 << 1) +/* MISO Function Control: Secondary Word Clock */ +#define AIC3204_MFP4_FUNC_SEC_WCLK (10 << 1) +/* MISO Function Control Mask */ +#define AIC3204_MFP4_FUNC (15 << 1) +/* MISO Output State */ +#define AIC3204_MFP4_OUT (1 << 0) + +/* SCLK/MFP3 Function Control Register */ +#define AIC3204_MFP3 AIC3204_PGREG(0, 56) +/* SCLK Function Control: Disabled */ +#define AIC3204_MFP3_FUNC_DISABLED (0 << 1) +/* SCLK Function Control: SPI Clock / Secondary Interface / Digital Mic Input */ +#define AIC3204_MFP3_FUNC_SPI_SI_DM (1 << 1) +/* SCLK Function Control: General Purpose Input */ +#define AIC3204_MFP3_FUNC_INPUT (2 << 1) +/* SCLK Function Control Mask */ +#define AIC3204_MFP3_FUNC (3 << 1) +/* SCLK Input State */ +#define AIC3204_MFP3_IN (1 << 0) + +/* DAC Signal Processing Block Control Register */ +#define AIC3204_DACSPB AIC3204_PGREG(0, 60) +/* DAC Signal Processing Block Mask */ +#define AIC3204_DACSPB_MASK 0x1f + +/* ADC Signal Processing Block Control Register */ +#define AIC3204_ADCSPB AIC3204_PGREG(0, 61) +/* ADC Signal Processing Block Mask */ +#define AIC3204_ADCSPB_MASK 0x1f + +/* DAC Channel Setup Register 1 */ +#define AIC3204_DACS1 AIC3204_PGREG(0, 63) +/* Left DAC Powered Up */ +#define AIC3204_DACS1_LDAC_UP_SHIFT (7) +/* Right DAC Powered Up */ +#define AIC3204_DACS1_RDAC_UP_SHIFT (6) +/* Left DAC Data Path Control: Disabled */ +#define AIC3204_DACS1_LDACD_DIS (0 << 4) +/* Left DAC Data Path Control: Left Data */ +#define AIC3204_DACS1_LDACD_LEFT (1 << 4) +/* Left DAC Data Path Control: Right Data */ +#define AIC3204_DACS1_LDACD_RIGHT (2 << 4) +/* Left DAC Data Path Control: Left + Right */ +#define AIC3204_DACS1_LDACD_MIX (3 << 4) +/* Left DAC Data Path Control Mask */ +#define AIC3204_DACS1_LDACD (3 << 4) +/* Right DAC Data Path Control: Disabled */ +#define AIC3204_DACS1_RDACD_DIS (0 << 2) +/* Right DAC Data Path Control: Right Data */ +#define AIC3204_DACS1_RDACD_RIGHT (1 << 2) +/* Right DAC Data Path Control: Left Data */ +#define AIC3204_DACS1_RDACD_LEFT (2 << 2) +/* Right DAC Data Path Control: Left + Right */ +#define AIC3204_DACS1_RDACD_MIX (3 << 2) +/* Right DAC Data Path Control Mask */ +#define AIC3204_DACS1_RDACD (3 << 2) +/* DAC Soft-Step: Disabled */ +#define AIC3204_DACS1_SOFT_DIS (0 << 0) +/* DAC Soft-Step: 1 step every clock */ +#define AIC3204_DACS1_SOFT_1SEC (1 << 0) +/* DAC Soft-Step: 1 step every 2 clocks */ +#define AIC3204_DACS1_SOFT_1SE2C (2 << 0) +/* DAC Soft-Step Mask */ +#define AIC3204_DACS1_SOFT (3 << 0) + +/* DAC Channel Setup Register 2 */ +#define AIC3204_DACS2 AIC3204_PGREG(0, 64) +/* Right Modulator Output Control: Output is inverted left output */ +#define AIC3204_DACS2_RMOD_INV_SHIFT (7) +/* DAC Auto Mute: Disabled */ +#define AIC3204_DACS2_AMUTE_DIS (0 << 4) +/* DAC Auto Mute: After 100 samples of DC */ +#define AIC3204_DACS2_AMUTE_100 (1 << 4) +/* DAC Auto Mute: After 200 samples of DC */ +#define AIC3204_DACS2_AMUTE_200 (2 << 4) +/* DAC Auto Mute: After 400 samples of DC */ +#define AIC3204_DACS2_AMUTE_400 (3 << 4) +/* DAC Auto Mute: After 800 samples of DC */ +#define AIC3204_DACS2_AMUTE_800 (4 << 4) +/* DAC Auto Mute: After 1600 samples of DC */ +#define AIC3204_DACS2_AMUTE_1600 (5 << 4) +/* DAC Auto Mute: After 3200 samples of DC */ +#define AIC3204_DACS2_AMUTE_3200 (6 << 4) +/* DAC Auto Mute: After 6400 samples of DC */ +#define AIC3204_DACS2_AMUTE_6400 (7 << 4) +/* DAC Auto Mute Mask */ +#define AIC3204_DACS2_AMUTE (7 << 4) +/* Left DAC Muted */ +#define AIC3204_DACS2_LEFT_MUTE_SHIFT (3) +/* Right DAC Muted */ +#define AIC3204_DACS2_RIGHT_MUTE_SHIFT (2) +/* DAC Master Volume Control: Independant */ +#define AIC3204_DACS2_VOL_IND (0 << 0) +/* DAC Master Volume Control: Left controls right */ +#define AIC3204_DACS2_VOL_LEFT (1 << 0) +/* DAC Master Volume Control: Right controls left */ +#define AIC3204_DACS2_VOL_RIGHT (2 << 0) +/* DAC Master Volume Control Mask */ +#define AIC3204_DACS2_VOL (3 << 0) + +/* Left DAC Digital Volume Control */ +#define AIC3204_LDACVOL AIC3204_PGREG(0, 65) + +/* Right DAC Digital Volume Control */ +#define AIC3204_RDACVOL AIC3204_PGREG(0, 66) + +/* Headset Detection Configuration Register */ +#define AIC3204_HSDET AIC3204_PGREG(0, 67) +/* Enable Headset Detection */ +#define AIC3204_HSDET_ENABLE (1 << 7) +/* Headset Type: No Headset */ +#define AIC3204_HSDET_NONE (0 << 5) +/* Headset Type: Stereo Headset */ +#define AIC3204_HSDET_STEREO (1 << 5) +/* Headset Type: Cellular Stereo Headset */ +#define AIC3204_HSDET_CELLSTEREO (3 << 5) +/* Headset Detection Debounce Time: 16ms */ +#define AIC3204_HSDET_PLUGDT_16 (0 << 2) +/* Headset Detection Debounce Time: 32ms */ +#define AIC3204_HSDET_PLUGDT_32 (1 << 2) +/* Headset Detection Debounce Time: 64ms */ +#define AIC3204_HSDET_PLUGDT_64 (2 << 2) +/* Headset Detection Debounce Time: 128ms */ +#define AIC3204_HSDET_PLUGDT_128 (3 << 2) +/* Headset Detection Debounce Time: 256ms */ +#define AIC3204_HSDET_PLUGDT_256 (4 << 2) +/* Headset Detection Debounce Time: 512ms */ +#define AIC3204_HSDET_PLUGDT_512 (5 << 2) +/* Headset Detection Debounce Time Mask */ +#define AIC3204_HSDET_PLUGDT (7 << 2) +/* Headset Button Debounce Time: 8ms */ +#define AIC3204_HSDET_BTNDT_8 (0 << 0) +/* Headset Button Debounce Time: 16ms */ +#define AIC3204_HSDET_BTNDT_16 (1 << 0) +/* Headset Button Debounce Time: 32ms */ +#define AIC3204_HSDET_BTNDT_32 (2 << 0) +/* Headset Button Debounce Time Mask */ +#define AIC3204_HSDET_BTNDT (3 << 0) + +/* DRC Control Register 1 */ +#define AIC3204_DRC1 AIC3204_PGREG(0, 68) +/* Left Channel DRC Enable */ +#define AIC3204_DRC1_LEFT_EN (1 << 6) +/* Right Channel DRC Enable */ +#define AIC3204_DRC1_RIGHT_EN (1 << 5) +/* DRC Threshold: -3dBFS */ +#define AIC3204_DRC1_THOLD_3DBFS (0 << 2) +/* DRC Threshold: -6dBFS */ +#define AIC3204_DRC1_THOLD_6DBFS (1 << 2) +/* DRC Threshold: -9dBFS */ +#define AIC3204_DRC1_THOLD_9DBFS (2 << 2) +/* DRC Threshold: -12dBFS */ +#define AIC3204_DRC1_THOLD_12DBFS (3 << 2) +/* DRC Threshold: -15dBFS */ +#define AIC3204_DRC1_THOLD_15DBFS (4 << 2) +/* DRC Threshold: -18dBFS */ +#define AIC3204_DRC1_THOLD_18DBFS (5 << 2) +/* DRC Threshold: -21dBFS */ +#define AIC3204_DRC1_THOLD_21DBFS (6 << 2) +/* DRC Threshold: -24dBFS */ +#define AIC3204_DRC1_THOLD_24DBFS (7 << 2) +/* DRC Threshold Mask */ +#define AIC3204_DRC1_THOLD (7 << 2) +/* DRC Hysteresis: 0dB */ +#define AIC3204_DRC1_HYST_0DB (0 << 0) +/* DRC Hysteresis: 1dB */ +#define AIC3204_DRC1_HYST_1DB (1 << 0) +/* DRC Hysteresis: 2dB */ +#define AIC3204_DRC1_HYST_2DB (2 << 0) +/* DRC Hysteresis: 3dB */ +#define AIC3204_DRC1_HYST_3DB (3 << 0) +/* DRC Hysteresis Mask */ +#define AIC3204_DRC1_HYST (3 << 0) + +/* DRC Control Register 2 */ +#define AIC3204_DRC2 AIC3204_PGREG(0, 69) +/* DRC Hold: Disabled */ +#define AIC3204_DRC2_HOLD_DISABLED (0 << 3) +/* DRC Hold: 32 Word Clocks */ +#define AIC3204_DRC2_HOLD_32WC (1 << 3) +/* DRC Hold: 64 Word Clocks */ +#define AIC3204_DRC2_HOLD_64WC (2 << 3) +/* DRC Hold: 128 Word Clocks */ +#define AIC3204_DRC2_HOLD_128WC (3 << 3) +/* DRC Hold: 256 Word Clocks */ +#define AIC3204_DRC2_HOLD_256WC (4 << 3) +/* DRC Hold: 512 Word Clocks */ +#define AIC3204_DRC2_HOLD_512WC (5 << 3) +/* DRC Hold: 1024 Word Clocks */ +#define AIC3204_DRC2_HOLD_1024WC (6 << 3) +/* DRC Hold: 2048 Word Clocks */ +#define AIC3204_DRC2_HOLD_2048WC (7 << 3) +/* DRC Hold: 4096 Word Clocks */ +#define AIC3204_DRC2_HOLD_4096WC (8 << 3) +/* DRC Hold: 8192 Word Clocks */ +#define AIC3204_DRC2_HOLD_8192WC (9 << 3) +/* DRC Hold: 16384 Word Clocks */ +#define AIC3204_DRC2_HOLD_16384WC (10 << 3) +/* DRC Hold: 32768 Word Clocks */ +#define AIC3204_DRC2_HOLD_32768WC (11 << 3) +/* DRC Hold: 65536 Word Clocks */ +#define AIC3204_DRC2_HOLD_65536WC (12 << 3) +/* DRC Hold: 98304 Word Clocks */ +#define AIC3204_DRC2_HOLD_98304WC (13 << 3) +/* DRC Hold: 131072 Word Clocks */ +#define AIC3204_DRC2_HOLD_131072WC (14 << 3) +/* DRC Hold: 163840 Word Clocks */ +#define AIC3204_DRC2_HOLD_163840WC (15 << 3) +/* DRC Hold Mask */ +#define AIC3204_DRC2_HOLD (15 << 3) + +/* DRC Control Register 3 */ +#define AIC3204_DRC3 AIC3204_PGREG(0, 70) +/* DRC Attack Rate Shift */ +#define AIC3204_DRC3_ATTACK_SHIFT 4 +/* DRC Attack Rate Mask */ +#define AIC3204_DRC3_ATTACK (15 << AIC3204_DRC3_ATTACK_SHIFT) +/* DRC Decay Rate Shift */ +#define AIC3204_DRC3_DECAY_SHIFT 0 +/* DRC Decay Rate Mask */ +#define AIC3204_DRC3_DECAY (15 << AIC3204_DRC3_DECAY_SHIFT) + +/* Beep Generator Register 1 */ +#define AIC3204_BEEP1 AIC3204_PGREG(0, 71) +/* Enable Beep Generator */ +#define AIC3204_BEEP1_EN (1 << 7) +/* Left Channel Beep Volume Shift */ +#define AIC3204_BEEP1_VOL_SHIFT 0 +/* Left Channel Beep Volume Mask */ +#define AIC3204_BEEP1_VOL (0x3f) + +/* Beep Generator Register 2 */ +#define AIC3204_BEEP2 AIC3204_PGREG(0, 72) +/* Beep Generator Master Volume: Independant Left/Right Channels */ +#define AIC3204_BEEP2_MVOL_IND (0 << 6) +/* Beep Generator Master Volume: Left Channel controls Right */ +#define AIC3204_BEEP2_MVOL_LEFT (1 << 6) +/* Beep Generator Master Volume: Right Channel controls Left */ +#define AIC3204_BEEP2_MVOL_RIGHT (2 << 6) +/* Beep Generator Master Volume Mask */ +#define AIC3204_BEEP2_MVOL (3 << 6) + +/* Beep Generator Register 3 */ +#define AIC3204_BEEP3 AIC3204_PGREG(0, 73) +/* Sample Length Value Shift */ +#define AIC3204_BEEP3_LENGTH_VSHIFT 16 +/* Sample Length Shift */ +#define AIC3204_BEEP3_LENGTH_SHIFT 0 +/* Sample Length Mask */ +#define AIC3204_BEEP3_LENGTH (0xff) + +/* Beep Generator Register 4 */ +#define AIC3204_BEEP4 AIC3204_PGREG(0, 74) +/* Sample Length Value Shift */ +#define AIC3204_BEEP4_LENGTH_VSHIFT 8 +/* Sample Length Shift */ +#define AIC3204_BEEP4_LENGTH_SHIFT 0 +/* Sample Length Mask */ +#define AIC3204_BEEP4_LENGTH (0xff) + +/* Beep Generator Register 5 */ +#define AIC3204_BEEP5 AIC3204_PGREG(0, 75) +/* Sample Length Value Shift */ +#define AIC3204_BEEP5_LENGTH_VSHIFT 0 +/* Sample Length Shift */ +#define AIC3204_BEEP5_LENGTH_SHIFT 0 +/* Sample Length Mask */ +#define AIC3204_BEEP5_LENGTH (0xff) + +/* Beep Generator Register 6 */ +#define AIC3204_BEEP6 AIC3204_PGREG(0, 76) +/* Relative Sine Frequency Value Shift */ +#define AIC3204_BEEP6_SINFREQ_VSHIFT 8 +/* Relative Sine Frequency Shift */ +#define AIC3204_BEEP6_SINFREQ_SHIFT 0 +/* Relative Sine Frequency Mask */ +#define AIC3204_BEEP6_SINFREQ (0xff) + +/* Beep Generator Register 7 */ +#define AIC3204_BEEP7 AIC3204_PGREG(0, 77) +/* Relative Sine Frequency Value Shift */ +#define AIC3204_BEEP7_SINFREQ_VSHIFT 0 +/* Relative Sine Frequency Shift */ +#define AIC3204_BEEP7_SINFREQ_SHIFT 0 +/* Relative Sine Frequency Mask */ +#define AIC3204_BEEP7_SINFREQ (0xff) + +/* Beep Generator Register 8 */ +#define AIC3204_BEEP8 AIC3204_PGREG(0, 78) +/* Relative Cosine Frequency Value Shift */ +#define AIC3204_BEEP8_COSFREQ_VSHIFT 8 +/* Relative Cosine Frequency Shift */ +#define AIC3204_BEEP8_COSFREQ_SHIFT 0 +/* Relative Cosine Frequency Mask */ +#define AIC3204_BEEP8_COSFREQ (0xff) + +/* Beep Generator Register 9 */ +#define AIC3204_BEEP9 AIC3204_PGREG(0, 79) +/* Relative Cosine Frequency Value Shift */ +#define AIC3204_BEEP9_COSFREQ_VSHIFT 0 +/* Relative Cosine Frequency Shift */ +#define AIC3204_BEEP9_COSFREQ_SHIFT 0 +/* Relative Cosine Frequency Mask */ +#define AIC3204_BEEP9_COSFREQ (0xff) + +/* ADC Channel Setup Register */ +#define AIC3204_ADCS AIC3204_PGREG(0, 81) +/* Left Channel ADC Powered Up */ +#define AIC3204_ADCS_LADC_UP_SHIFT (7) +/* Right Channel ADC Powered Up */ +#define AIC3204_ADCS_RADC_UP_SHIFT (6) +/* Digital Microphone Input: GPIO */ +#define AIC3204_ADC5_DMICIN_GPIO (0 << 4) +/* Digital Microphone Input: SCLK */ +#define AIC3204_ADC5_DMICIN_SCLK (1 << 4) +/* Digital Microphone Input: DIN */ +#define AIC3204_ADC5_DMICIN_DIN (2 << 4) +/* Digital Microphone Input Mask */ +#define AIC3204_ADC5_DMICIN (3 << 4) +/* Left ADC Source: Digital Microphone */ +#define AIC3204_ADC5_LADC_DMIC (1 << 3) +/* Right ADC Source: Digital Microphone */ +#define AIC3204_ADC5_RADC_DMIC (1 << 2) +/* ADC Volume Soft Step: 1 step / word clock */ +#define AIC3204_ADC5_VOLSS_1SPWC (0 << 0) +/* ADC Volume Soft Step: 1 step / 2 word clocks */ +#define AIC3204_ADC5_VOLSS_1SP2WC (1 << 0) +/* ADC Volume Soft Step: Disabled */ +#define AIC3204_ADC5_VOLSS_DISABLED (2 << 0) +/* ADC Volume Soft Step Mask */ +#define AIC3204_ADC5_VOLSS (3 << 0) + +/* ADC Fine Gain Adjust Register */ +#define AIC3204_ADCFINE AIC3204_PGREG(0, 82) +/* Left ADC Mute */ +#define AIC3204_ADCFINE_LEFT_MUTE (1 << 7) +/* Left ADC Gain: 0dB */ +#define AIC3204_ADCFINE_LEFT_0DB (0 << 4) +/* Left ADC Gain: -0.1dB */ +#define AIC3204_ADCFINE_LEFT_0DB1 (1 << 4) +/* Left ADC Gain: -0.2dB */ +#define AIC3204_ADCFINE_LEFT_0DB2 (2 << 4) +/* Left ADC Gain: -0.3dB */ +#define AIC3204_ADCFINE_LEFT_0DB3 (2 << 4) +/* Left ADC Gain: -0.4dB */ +#define AIC3204_ADCFINE_LEFT_0DB4 (3 << 4) +/* Left ADC Gain Mask */ +#define AIC3204_ADCFINE_LEFT (7 << 4) +/* Right ADC Mute */ +#define AIC3204_ADCFINE_RIGHT_MUTE (1 << 3) +/* Right ADC Gain: 0dB */ +#define AIC3204_ADCFINE_RIGHT_0DB (0 << 0) +/* Right ADC Gain: -0.1dB */ +#define AIC3204_ADCFINE_RIGHT_0DB1 (1 << 0) +/* Right ADC Gain: -0.2dB */ +#define AIC3204_ADCFINE_RIGHT_0DB2 (2 << 0) +/* Right ADC Gain: -0.3dB */ +#define AIC3204_ADCFINE_RIGHT_0DB3 (2 << 0) +/* Right ADC Gain: -0.4dB */ +#define AIC3204_ADCFINE_RIGHT_0DB4 (3 << 0) +/* Right ADC Gain Mask */ +#define AIC3204_ADCFINE_RIGHT (7 << 0) + +/* Left ADC Channel Volume Register */ +#define AIC3204_LADCVOL AIC3204_PGREG(0, 83) +/* Left ADC Channel Volume Mask */ +#define AIC3204_LADCVOL_MASK 0x7f + +/* Right ADC Channel Volume Register */ +#define AIC3204_RADCVOL AIC3204_PGREG(0, 84) +/* Right ADC Channel Volume Mask */ +#define AIC3204_RADCVOL_MASK 0x7f + +/* ADC Phase Adjust Register */ +#define AIC3204_ADCPHASE AIC3204_PGREG(0, 85) + +/* Left Channel AGC Control Register 1 */ +#define AIC3204_LAGC1 AIC3204_PGREG(0, 86) +/* Left Channel AGC Enabled */ +#define AIC3204_LAGC1_ENABLE (1 << 7) +/* Left Channel Target Level: -5.5dBFS */ +#define AIC3204_LAGC1_LEVEL_5DBFS5 (0 << 4) +/* Left Channel Target Level: -8.0dBFS */ +#define AIC3204_LAGC1_LEVEL_8DBFS (1 << 4) +/* Left Channel Target Level: -10dBFS */ +#define AIC3204_LAGC1_LEVEL_10DBFS (2 << 4) +/* Left Channel Target Level: -12dBFS */ +#define AIC3204_LAGC1_LEVEL_12DBFS (3 << 4) +/* Left Channel Target Level: -14dBFS */ +#define AIC3204_LAGC1_LEVEL_14DBFS (4 << 4) +/* Left Channel Target Level: -17dBFS */ +#define AIC3204_LAGC1_LEVEL_17DBFS (5 << 4) +/* Left Channel Target Level: -20dBFS */ +#define AIC3204_LAGC1_LEVEL_20DBFS (6 << 4) +/* Left Channel Target Level: -24dBFS */ +#define AIC3204_LAGC1_LEVEL_24DBFS (7 << 4) +/* Left Channel Target Level Mask */ +#define AIC3204_LAGC1_LEVEL (7 << 4) +/* Left Channel Gain Hysteresis: Disabled */ +#define AIC3204_LAGC1_GAINHYST_DISABLED (0 << 0) +/* Left Channel Gain Hysteresis: 0.5dB */ +#define AIC3204_LAGC1_GAINHYST_0DB5 (1 << 0) +/* Left Channel Gain Hysteresis: 1dB */ +#define AIC3204_LAGC1_GAINHYST_1DB (2 << 0) +/* Left Channel Gain Hysteresis: 1.5dB */ +#define AIC3204_LAGC1_GAINHYST_1DB5 (3 << 0) +/* Left Channel Gain Hysteresis Mask */ +#define AIC3204_LAGC1_GAINHYST (3 << 0) + +/* Left Channel AGC Control Register 2 */ +#define AIC3204_LAGC2 AIC3204_PGREG(0, 87) +/* Left Channel AGC Hysteresis: 1dB */ +#define AIC3204_LAGC2_HYST_1DB (0 << 6) +/* Left Channel AGC Hysteresis: 2dB */ +#define AIC3204_LAGC2_HYST_2DB (1 << 6) +/* Left Channel AGC Hysteresis: 4dB */ +#define AIC3204_LAGC2_HYST_4DB (2 << 6) +/* Left Channel AGC Hysteresis: Disabled */ +#define AIC3204_LAGC2_HYST_DISABLED (3 << 6) +/* Left Channel AGC Hysteresis Mask */ +#define AIC3204_LAGC2_HYST (3 << 6) +/* Left Channel AGC Noise Gate Threshold Mask */ +#define AIC3204_LAGC2_NGATE (0x1f) +/* Left Channel AGC Noise Gate Threshold Shift */ +#define AIC3204_LAGC2_NGATE_SHIFT 1 + +/* TODO: Remaining AGC registers, DC measurement registers */ + +/* Power Configuration Register */ +#define AIC3204_POWER AIC3204_PGREG(1, 1) +/* Disable AVDD -> DVDD Link */ +#define AIC3204_POWER_AVDD_DVDD_UNLINK (1 << 3) + +/* LDO Control Register */ +#define AIC3204_LDO AIC3204_PGREG(1, 2) +/* DVDD LDO: 1.72V */ +#define AIC3204_LDO_DVDD_1V72 (0 << 6) +/* DVDD LDO: 1.67V */ +#define AIC3204_LDO_DVDD_1V67 (1 << 6) +/* DVDD LDO: 1.77V */ +#define AIC3204_LDO_DVDD_1V77 (2 << 6) +/* DVDD LDO Mask */ +#define AIC3204_LDO_DVDD (3 << 6) +/* AVDD LDO: 1.72V */ +#define AIC3204_LDO_AVDD_1V72 (0 << 4) +/* AVDD LDO: 1.67V */ +#define AIC3204_LDO_AVDD_1V67 (1 << 4) +/* AVDD LDO: 1.77V */ +#define AIC3204_LDO_AVDD_1V77 (2 << 4) +/* AVDD LDO Mask */ +#define AIC3204_LDO_AVDD (3 << 4) +/* Analog Blocks Enabled */ +#define AIC3204_LDO_ANALOG_ENABLED (0 << 3) +/* Analog Blocks Disabled */ +#define AIC3204_LDO_ANALOG_DISABLED (1 << 3) +/* Analog Block Status */ +#define AIC3204_LDO_ANALOG (1 << 3) +/* DVDD Over Current Detect */ +#define AIC3204_LDO_DVDD_OVERCURRENT (1 << 2) +/* AVDD Over Current Detect */ +#define AIC3204_LDO_AVDD_OVERCURRENT (1 << 1) +/* AVDD Powered Up */ +#define AIC3204_LDO_AVDD_UP (1 << 0) + +/* TODO: Playback Configuration Registers */ + +/* Output Driver Power Control Register */ +#define AIC3204_OUTDRV AIC3204_PGREG(1, 9) +/* Left Headphone Driver Up */ +#define AIC3204_OUTDRV_HPL_UP_SHIFT (5) +/* Right Headphone Driver Up */ +#define AIC3204_OUTDRV_HPR_UP_SHIFT (4) +/* Left Line Output Driver Up */ +#define AIC3204_OUTDRV_LOL_UP_SHIFT (3) +/* Right Line Output Driver Up */ +#define AIC3204_OUTDRV_LOR_UP_SHIFT (2) +/* Left Mixer Amplifier Up */ +#define AIC3204_OUTDRV_MAL_UP_SHIFT (1) +/* Right Mixer Amplifier Up */ +#define AIC3204_OUTDRV_MAR_UP_SHIFT (0) + +/* Common Mode Control Register */ +#define AIC3204_CMCTL AIC3204_PGREG(1, 10) +/* Full Chip Common Mode = 0.9V */ +#define AIC3204_CMCTL_FCCM_0V9 (0 << 6) +/* Full Chip Common Mode = 0.75V */ +#define AIC3204_CMCTL_FCCM_0V75 (1 << 6) +/* Full Chip Common Mode Mask */ +#define AIC3204_CMCTL_FCCM (1 << 6) +/* Headphone Output Common Mode = Full Chip Common Mode */ +#define AIC3204_CMCTL_HPCM_FCCM (0 << 4) +/* Headphone Output Common Mode = 1.25V */ +#define AIC3204_CMCTL_HPCM_1V25 (1 << 4) +/* Headphone Output Common Mode = 1.5V */ +#define AIC3204_CMCTL_HPCM_1V5 (2 << 4) +/* Headphone Output Common Mode = 1.65V */ +#define AIC3204_CMCTL_HPCM_1V65 (3 << 4) +/* Headphone Output Common Mode Mask */ +#define AIC3204_CMCTL_HPCM (3 << 4) +/* Line Output Common Mode = Full Chip Common Mode */ +#define AIC3204_CMCTL_LOCM_FCCM (0 << 3) +/* Line Output Common Mode = 1.65V from LDOin */ +#define AIC3204_CMCTL_LOCM_1V65LDOIN (1 << 3) +/* Line Output Common Mode Mask */ +#define AIC3204_CMCTL_LOCM (1 << 3) +/* Headphone Output Common Mode Source = AVDD */ +#define AIC3204_CMCTL_HPCMSRC_AVDD (0 << 1) +/* Headphone Output Common Mode Source = LDOin */ +#define AIC3204_CMCTL_HPCMSRC_LDOIN (1 << 1) +/* Headphone Output Common Mode Source Mask */ +#define AIC3204_CMCTL_HPCMSRC (1 << 1) +/* LDO Input Range = 1.5~1.95V */ +#define AIC3204_CMCTL_LDORANGE_1V5_1V95 (0 << 0) +/* LDO Input Range = 1.8~3.6V */ +#define AIC3204_CMCTL_LDORANGE_1V8_3V6 (1 << 0) +/* LDO Input Range Mask (effective when HPCMSRC = LDOIN) */ +#define AIC3204_CMCTL_LDORANGE (1 << 0) + +/* Over Current Protection Configuration Register TODO */ +#define AIC3204_OCP AIC3204_PGREG(1, 11) + +/* Left Headphone Driver Routing Register */ +#define AIC3204_HPLROUTE AIC3204_PGREG(1, 12) +/* Left Headphone Driver connects to LDAC+ */ +#define AIC3204_HPLROUTE_LDACP_SHIFT (3) +/* Left Headphone Driver connects to IN1L */ +#define AIC3204_HPLROUTE_IN1L_SHIFT (2) +/* Left Headphone Driver connects to MAL */ +#define AIC3204_HPLROUTE_MAL_SHIFT (1) +/* Left Headphone Driver connects to MAR */ +#define AIC3204_HPLROUTE_MAR_SHIFT (0) + +/* Right Headphone Driver Routing Register */ +#define AIC3204_HPRROUTE AIC3204_PGREG(1, 13) +/* Right Headphone Driver connects to LDAC- */ +#define AIC3204_HPRROUTE_LDACN_SHIFT (4) +/* Right Headphone Driver connects to RDAC+ */ +#define AIC3204_HPRROUTE_RDACP_SHIFT (3) +/* Right Headphone Driver connects to IN1L */ +#define AIC3204_HPRROUTE_IN1R_SHIFT (2) +/* Right Headphone Driver connects to MAL */ +#define AIC3204_HPRROUTE_MAR_SHIFT (1) +/* Right Headphone Driver connects to HPL */ +#define AIC3204_HPRROUTE_HPL_SHIFT (0) + +/* Left Line Output Routing Register */ +#define AIC3204_LOLROUTE AIC3204_PGREG(1, 14) +/* Left Line Output Driver connects to RDAC- */ +#define AIC3204_LOLROUTE_RDACN_SHIFT (4) +/* Left Line Output Driver connects to LDAC+ */ +#define AIC3204_LOLROUTE_LDACP_SHIFT (3) +/* Left Line Output Driver connects to MAL */ +#define AIC3204_LOLROUTE_MAL_SHIFT (1) +/* Left Line Output Driver connects to LOR */ +#define AIC3204_LOLROUTE_LOR_SHIFT (0) + +/* Right Line Output Routing Register */ +#define AIC3204_LORROUTE AIC3204_PGREG(1, 15) +/* Right Line Output Driver connects to RDAC+ */ +#define AIC3204_LORROUTE_RDACP_SHIFT (3) +/* Right Line Output Driver connects to MAR */ +#define AIC3204_LORROUTE_MAR_SHIFT (1) + +/* Left Headphone Gain Setting Register */ +#define AIC3204_HPLGAIN AIC3204_PGREG(1, 16) +/* Left Headphone Mute */ +#define AIC3204_HPLGAIN_MUTE_SHIFT (6) +/* Left Headphone Gain Mask */ +#define AIC3204_HPLGAIN_MASK (0x3f) + +/* Right Headphone Gain Setting Register */ +#define AIC3204_HPRGAIN AIC3204_PGREG(1, 17) +/* Right Headphone Mute */ +#define AIC3204_HPRGAIN_MUTE_SHIFT (6) +/* Right Headphone Gain Mask */ +#define AIC3204_HPRGAIN_MASK (0x3f) + +/* Left Line Output Gain Setting Register */ +#define AIC3204_LOLGAIN AIC3204_PGREG(1, 18) +/* Left Line Output Mute */ +#define AIC3204_LOLGAIN_MUTE_SHIFT (6) +/* Left Line Output Gain Mask */ +#define AIC3204_LOLGAIN_MASK (0x3f) + +/* Right Line Output Gain Setting Register */ +#define AIC3204_LORGAIN AIC3204_PGREG(1, 19) +/* Right Line Output Mute */ +#define AIC3204_LORGAIN_MUTE_SHIFT (6) +/* Right Line Output Gain Mask */ +#define AIC3204_LORGAIN_MASK (0x3f) + +/* Left Mixer Amplifier Gain Setting Register */ +#define AIC3204_MALGAIN AIC3204_PGREG(1, 24) +/* Left Mixer Amplifier Gain Mask */ +#define AIC3204_MALGAIN_MASK (0x3f) + +/* Right Mixer Amplifier Gain Setting Register */ +#define AIC3204_MARGAIN AIC3204_PGREG(1, 25) +/* Right Mixer Amplifier Gain Mask */ +#define AIC3204_MARGAIN_MASK (0x3f) + +#define AIC3204_MICBIAS AIC3204_PGREG(1, 51) + +#define AIC3204_LMICPGAPROUTE AIC3204_PGREG(1, 52) +#define AIC3204_LMICPGANROUTE AIC3204_PGREG(1, 54) + +#define AIC3204_RMICPGAPROUTE AIC3204_PGREG(1, 55) +#define AIC3204_RMICPGANROUTE AIC3204_PGREG(1, 57) + +#define AIC3204_LMICPGAVOL AIC3204_PGREG(1, 59) +#define AIC3204_RMICPGAVOL AIC3204_PGREG(1, 60) + +/* ADC Power Tune Mode Register */ +#define AIC3204_ADCPTM AIC3204_PGREG(1, 61) +/* ADC Power Tune Mode: PTM_R4 */ +#define AIC3204_ADCPTM_R4 (0x00) +/* ADC Power Tune Mode: PTM_R3 */ +#define AIC3204_ADCPTM_R3 (0x64) +/* ADC Power Tune Mode: PTM_R2 */ +#define AIC3204_ADCPTM_R2 (0xb6) +/* ADC Power Tune Mode: PTM_R1 */ +#define AIC3204_ADCPTM_R1 (0xff) + +/* Analogue Input Quick Charging Configuration Register */ +#define AIC3204_AIQC AIC3204_PGREG(1, 71) +/* Power up time: Default Value (whatever that is) */ +#define AIC3204_AIQC_PWRUP_DEFAULT (0x00) +/* Power up time: 3.1ms */ +#define AIC3204_AIQC_PWRUP_3M1S (0x31) +/* Power up time: 6.4ms */ +#define AIC3204_AIQC_PWRUP_6M4S (0x32) +/* Power up time: 1.6ms */ +#define AIC3204_AIQC_PWRUP_1M6S (0x33) +/* Power up time mask */ +#define AIC3204_AIQC_PWRUP (0x3f) + +/* Reference Power-Up Configuration Register */ +#define AIC3204_REFPU AIC3204_PGREG(1, 123) +/* Reference Power-Up delay: Slow */ +#define AIC3204_REFPU_DELAY_SLOW (0 << 0) +/* Reference Power-Up delay: 40ms */ +#define AIC3204_REFPU_DELAY_40MS (1 << 0) +/* Reference Power-Up delay: 80ms */ +#define AIC3204_REFPU_DELAY_80MS (2 << 0) +/* Reference Power-Up delay: 120ms */ +#define AIC3204_REFPU_DELAY_120MS (3 << 0) +/* Force power-up */ +#define AIC3204_REFPU_FORCE (1 << 2) + +/* TODO: Define the remaining registers ... this will do for now */ + +extern struct snd_soc_dai aic3204_dai; +extern struct snd_soc_codec_device soc_codec_dev_aic3204; + +#endif /* _AIC3204_H */ -- 1.6.4.4

2 7

[alsa-devel] [PATCH v3] alsa: ASoC: JZ4740: Add qi_lb60 board driver
by 1276924111-11158-27-git-send-email-lars＠metafoo.de 22 Jun '10

22 Jun '10

From: Lars-Peter Clausen <lars(a)metafoo.de> This patch adds ASoC support for the qi_lb60 board. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: alsa-devel(a)alsa-project.org -- Changes since v1 - Refer to AMP gpios always by their define - Do not try to set codecs format, since the set_fmt callback for the codec was dropped. Changes since v2 - The codec prefix was removed from jz4740 codec include file - This patch accidentally contained a few bits meant for ASoC JZ4740 platform support patch. Removed them. --- sound/soc/jz4740/Kconfig | 9 +++ sound/soc/jz4740/Makefile | 4 + sound/soc/jz4740/qi_lb60.c | 166 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 179 insertions(+), 0 deletions(-) create mode 100644 sound/soc/jz4740/qi_lb60.c diff --git a/sound/soc/jz4740/Kconfig b/sound/soc/jz4740/Kconfig index 27480f2..5351cba 100644 --- a/sound/soc/jz4740/Kconfig +++ b/sound/soc/jz4740/Kconfig @@ -12,3 +12,12 @@ config SND_JZ4740_SOC_I2S help Say Y if you want to use I2S protocol and I2S codec on Ingenic JZ4740 based boards. + +config SND_JZ4740_SOC_QI_LB60 + tristate "SoC Audio support for Qi LB60" + depends on SND_JZ4740_SOC && JZ4740_QI_LB60 + select SND_JZ4740_SOC_I2S + select SND_SOC_JZ4740_CODEC + help + Say Y if you want to add support for ASoC audio on the Qi LB60 board + a.k.a Qi Ben NanoNote. diff --git a/sound/soc/jz4740/Makefile b/sound/soc/jz4740/Makefile index 1be8d19..be873c1 100644 --- a/sound/soc/jz4740/Makefile +++ b/sound/soc/jz4740/Makefile @@ -7,3 +7,7 @@ snd-soc-jz4740-i2s-objs := jz4740-i2s.o obj-$(CONFIG_SND_JZ4740_SOC) += snd-soc-jz4740.o obj-$(CONFIG_SND_JZ4740_SOC_I2S) += snd-soc-jz4740-i2s.o +# Jz4740 Machine Support +snd-soc-qi-lb60-objs := qi_lb60.o + +obj-$(CONFIG_SND_JZ4740_SOC_QI_LB60) += snd-soc-qi-lb60.o diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c new file mode 100644 index 0000000..f15f491 --- /dev/null +++ b/sound/soc/jz4740/qi_lb60.c @@ -0,0 +1,166 @@ +/* + * Copyright (C) 2009, Lars-Peter Clausen <lars(a)metafoo.de> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/gpio.h> + +#include "../codecs/jz4740.h" +#include "jz4740-pcm.h" +#include "jz4740-i2s.h" + + +#define QI_LB60_SND_GPIO JZ_GPIO_PORTB(29) +#define QI_LB60_AMP_GPIO JZ_GPIO_PORTD(4) + +static int qi_lb60_spk_event(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *ctrl, int event) +{ + int on = 0; + if (event & SND_SOC_DAPM_POST_PMU) + on = 1; + else if (event & SND_SOC_DAPM_PRE_PMD) + on = 0; + + gpio_set_value(QI_LB60_SND_GPIO, on); + gpio_set_value(QI_LB60_AMP_GPIO, on); + + return 0; +} + +static const struct snd_soc_dapm_widget qi_lb60_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", qi_lb60_spk_event), + SND_SOC_DAPM_MIC("Mic", NULL), +}; + +static const struct snd_soc_dapm_route qi_lb60_routes[] = { + {"Mic", NULL, "MIC"}, + {"Speaker", NULL, "LOUT"}, + {"Speaker", NULL, "ROUT"}, +}; + +#define QI_LB60_DAIFMT (SND_SOC_DAIFMT_I2S | \ + SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBM_CFM) + +static int qi_lb60_codec_init(struct snd_soc_codec *codec) +{ + int ret; + struct snd_soc_dai *cpu_dai = codec->socdev->card->dai_link->cpu_dai; + + snd_soc_dapm_nc_pin(codec, "LIN"); + snd_soc_dapm_nc_pin(codec, "RIN"); + + ret = snd_soc_dai_set_fmt(cpu_dai, QI_LB60_DAIFMT); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cpu dai format: %d\n", ret); + return ret; + } + + snd_soc_dapm_new_controls(codec, qi_lb60_widgets, ARRAY_SIZE(qi_lb60_widgets)); + snd_soc_dapm_add_routes(codec, qi_lb60_routes, ARRAY_SIZE(qi_lb60_routes)); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link qi_lb60_dai = { + .name = "jz4740", + .stream_name = "jz4740", + .cpu_dai = &jz4740_i2s_dai, + .codec_dai = &jz4740_codec_dai, + .init = qi_lb60_codec_init, +}; + +static struct snd_soc_card qi_lb60 = { + .name = "QI LB60", + .dai_link = &qi_lb60_dai, + .num_links = 1, + .platform = &jz4740_soc_platform, +}; + +static struct snd_soc_device qi_lb60_snd_devdata = { + .card = &qi_lb60, + .codec_dev = &soc_codec_dev_jz4740_codec, +}; + +static struct platform_device *qi_lb60_snd_device; + +static int __init qi_lb60_init(void) +{ + int ret; + + qi_lb60_snd_device = platform_device_alloc("soc-audio", -1); + + if (!qi_lb60_snd_device) + return -ENOMEM; + + ret = gpio_request(QI_LB60_SND_GPIO, "SND"); + if (ret) { + pr_err("qi_lb60 snd: Failed to request SND GPIO(%d): %d\n", + QI_LB60_SND_GPIO, ret); + goto err_device_put; + } + + ret = gpio_request(QI_LB60_AMP_GPIO, "AMP"); + if (ret) { + pr_err("qi_lb60 snd: Failed to request AMP GPIO(%d): %d\n", + QI_LB60_AMP_GPIO, ret); + goto err_gpio_free_snd; + } + + gpio_direction_output(QI_LB60_SND_GPIO, 0); + gpio_direction_output(QI_LB60_AMP_GPIO, 0); + + platform_set_drvdata(qi_lb60_snd_device, &qi_lb60_snd_devdata); + qi_lb60_snd_devdata.dev = &qi_lb60_snd_device->dev; + + ret = platform_device_add(qi_lb60_snd_device); + if (ret) { + pr_err("qi_lb60 snd: Failed to add snd soc device: %d\n", ret); + goto err_unset_pdata; + } + + return 0; + +err_unset_pdata: + platform_set_drvdata(qi_lb60_snd_device, NULL); +/*err_gpio_free_amp:*/ + gpio_free(QI_LB60_AMP_GPIO); +err_gpio_free_snd: + gpio_free(QI_LB60_SND_GPIO); +err_device_put: + platform_device_put(qi_lb60_snd_device); + + return ret; +} +module_init(qi_lb60_init); + +static void __exit qi_lb60_exit(void) +{ + gpio_free(QI_LB60_AMP_GPIO); + gpio_free(QI_LB60_SND_GPIO); + platform_device_unregister(qi_lb60_snd_device); +} +module_exit(qi_lb60_exit); + +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_DESCRIPTION("ALSA SoC QI LB60 Audio support"); +MODULE_LICENSE("GPL v2"); -- 1.5.6.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Add Macbook 5,2 quirk
by Luke Yelavich 22 Jun '10

22 Jun '10

From: Luke Yelavich <luke.yelavich(a)canonical.com> BugLink: https://bugs.launchpad.net/bugs/463178 Set Macbook 5,2 (106b:4a00) hardware to use ALC885_MB5 Cc: <stable(a)kernel.org> Signed-off-by: Luke Yelavich <luke.yelavich(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f1ce7d7..b145306 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9486,6 +9486,7 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24), SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC885_IMAC91), SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5), + SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC885_MB5), /* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2, * so apparently no perfect solution yet */ -- 1.7.0.4

2 1

[alsa-devel] reading 1 second of audio at a time
by folkert 22 Jun '10

22 Jun '10

Hi, Am trying to achieve something trivial: reading 1 second of audio at a time, but something goes wrong: Sound 1277148008.835094 Sound 1277148008.838224 Sound 1277148010.834839 Sound 1277148010.838001 Sound 1277148012.834523 Sound 1277148012.837580 Sound 1277148014.835240 Sound 1277148014.838333 As you can see; audio is returned for 2 seconds, then it sleeps 2 secondsn, then 2 seconds of audio, etc. What I do is: if (snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) { } if (snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE) < 0) { } exact_rate = 44100; if (snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &exact_rate, 0) < 0) { error_exit("Error setting rate.\n"); } if (*rate != exact_rate) { fprintf(stderr, "The rate %d Hz is not supported by your hardware.\n ==> Using %d Hz instead.\n", *rate, exact_rate); *rate = exact_rate; } if (snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 2) < 0) { } int periods = 2; if (snd_pcm_hw_params_set_periods(pcm_handle, hwparams, periods, 0) < 0) { } if (snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, exact_rate * 2 * n_channels) < 0) { } if (snd_pcm_hw_params(pcm_handle, hwparams) < 0) { } return pcm_handle; } void get_audio_1s(snd_pcm_t *pcm_handle, short *buffer, int rate, int channels) { int samples = channels * rate, index = 0; while(samples > 0) { int cur = samples, pcmreturn; while ((pcmreturn = snd_pcm_readi(pcm_handle, &buffer[index], cur)) < 0) { snd_pcm_prepare(pcm_handle); printf(" underrun\n"); } index += pcmreturn; samples -= pcmreturn; } } So I call this get_audio_1s() with the samplerate (44100) and the number of channels (2). Then it seems to return the first second at once, then the second second of audio too, then sleeps for 2 seconds(!) and then repeats. Folkert van Heusden -- Ever wonder what is out there? Any alien races? Then please support the seti@home project: setiathome.ssl.berkeley.edu ---------------------------------------------------------------------- Phone: +31-6-41278122, PGP-key: 1F28D8AE, www.vanheusden.com

3 2

[alsa-devel] Question about snd_soc_add_controls
by Kuninori Morimoto 22 Jun '10

22 Jun '10

Dear very kind guys I would like to use snd_soc_add_controls for ak4642. But I'm not good at it now, so, for 1st step, I would like to add "Playback Volume". Playback Volume length of ak4642 chip is -115.0 dB to +12.0 dB step is 0.5 dB. and It can use mute. So, I created code like this ---------------------------------------------------------------------------- static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1); /* 0.5 dB step */ static const struct snd_kcontrol_new ak4642_snd_controls[] = { SOC_DOUBLE_R_TLV("Playback Volume", L_DVC, R_DVC, 0, 256, 0, out_tlv), }; ---------------------------------------------------------------------------- But I wonder #define DECLARE_TLV_DB_SCALE(name, min, step, mute) Does this "mute" mean that it can use mute ? But how to ? #define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) What is this "xinvert" ? Can you please teach me about above ? And where is these code used ? It seems very complex for me... Best regards -- Kuninori Morimoto

2 2

Re: [alsa-devel] snd_mixer_selem_get_playback_volume return value outside allowed range
by Raymond Yau 22 Jun '10

22 Jun '10

2010/5/25 Clemens Ladisch <clemens(a)ladisch.de> > James Courtier-Dutton wrote: > > On 25 May 2010 09:38, Clemens Ladisch <clemens(a)ladisch.de> wrote: > > > All the others seem to be mistakes: > > > > > > > soc/codecs/stac9766.c:... > > > > soc/codecs/wm8350.c:... > > > > soc/codecs/wm8400.c:... > > > > soc/codecs/wm8990.c:... > > > > Could this maybe explain why pulseaudio was having such problems with > > alsa with regards to gain controls???? > > Only if it happens on embedded hardware that uses these particular > codecs. > > > Regards, > Clemens > http://git.alsa-project.org/?p=alsa-utils.git;a=commit;h=c9b86f49a8a1a8c337… your patch did not fix the buggy driver (e.g. alsa-pulse plugin) which return volume outside the alllowed range amixer -D pulse Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 65536 Mono: Front Left: Playback 98304 [150%] [on] Front Right: Playback 98304 [150%] [on] Should snd_mixer_selem_get_playback_volume() perform range check ? tatic int get_volume_ops(snd_mixer_elem_t *elem, int dir, snd_mixer_selem_channel_id_t channel, long *value) { selem_none_t *s = snd_mixer_elem_get_private(elem); if (s->selem.caps & SM_CAP_GVOLUME) dir = SM_PLAY; if ((unsigned int) channel >= s->str[dir].channels) return -EINVAL; *value = s->str[dir].vol[channel]; + if (*value > s->str[dir].max) + return -EINVAL; + if (*value < s->str[dir].min) + return -EINVAL; return 0;

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[alsa-devel] [PATCH] ASoC: pandora: fix CLKX polarity
by Grazvydas Ignotas 21 Jun '10

21 Jun '10

After mass production started it was found that several boards exhibit noise problems during sound playback. After some investigation it was determined that CLKX polarity is set incorrectly, and even if most boards can tolerate the wrong setting, there are some that don't. Fix polarity setup in the board file. As the clock settings for input and output now match, merge in and out functions and structures to simplify code. Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- sound/soc/omap/omap3pandora.c | 36 ++++++++---------------------------- 1 files changed, 8 insertions(+), 28 deletions(-) diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 87ce842..9eecac1 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -43,12 +43,14 @@ static struct regulator *omap3pandora_dac_reg; -static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, unsigned int fmt) +static int omap3pandora_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; int ret; /* Set codec DAI configuration */ @@ -91,24 +93,6 @@ static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream, return 0; } -static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - return omap3pandora_cmn_hw_params(substream, params, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBS_CFS); -} - -static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - return omap3pandora_cmn_hw_params(substream, params, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); -} - static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { @@ -231,12 +215,8 @@ static int omap3pandora_in_init(struct snd_soc_codec *codec) return snd_soc_dapm_sync(codec); } -static struct snd_soc_ops omap3pandora_out_ops = { - .hw_params = omap3pandora_out_hw_params, -}; - -static struct snd_soc_ops omap3pandora_in_ops = { - .hw_params = omap3pandora_in_hw_params, +static struct snd_soc_ops omap3pandora_ops = { + .hw_params = omap3pandora_hw_params, }; /* Digital audio interface glue - connects codec <--> CPU */ @@ -246,14 +226,14 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { .stream_name = "HiFi Out", .cpu_dai = &omap_mcbsp_dai[0], .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], - .ops = &omap3pandora_out_ops, + .ops = &omap3pandora_ops, .init = omap3pandora_out_init, }, { .name = "TWL4030", .stream_name = "Line/Mic In", .cpu_dai = &omap_mcbsp_dai[1], .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], - .ops = &omap3pandora_in_ops, + .ops = &omap3pandora_ops, .init = omap3pandora_in_init, } }; -- 1.6.3.3

2 1

[alsa-devel] [PATCH] SOUND: usb/endpoint, fix dangling pointer use
by Jiri Slaby 21 Jun '10

21 Jun '10

Stanse found that in snd_usb_parse_audio_endpoints, there is a dangling pointer dereference. When snd_usb_parse_audio_format fails, fp is freed, and continue invoked. On the next loop, there is "fp && fp->altsetting == 1 && fp->channels == 1" test, but fp is set from the last iteration (but is bogus) and thus ilegally dereferenced. Set fp to NULL before "continue". Signed-off-by: Jiri Slaby <jslaby(a)suse.cz> Cc: Daniel Mack <daniel(a)caiaq.de> Cc: Clemens Ladisch <clemens(a)ladisch.de> Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 9593b91..6f6596c 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -427,6 +427,7 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) { kfree(fp->rate_table); kfree(fp); + fp = NULL; continue; } -- 1.7.1

3 2

[alsa-devel] redirecting audio to another desktop machine
by Viral Mehta 21 Jun '10

21 Jun '10

Hi list, I have an embedded device running linux kernel and a minimal filesystem. I want to access this device remotely which means I should be able to connect to the device just like we do Remote Desktop over ethernet. I am able to do this. I am able to send key-event and framebuffer data whenever my desktop machine connects to the device. But, I am not able to send audio data on remote side. I am really not sure which device file (/dev/snd/ ???) I should read and capture the data and should send across.... Or can I read the playback device ? I think NO. Do I need to create some kind of virtual sound device ? Any pointers/suggestions are appreciated. Thanks, Viral This Email may contain confidential or privileged information for the intended recipient (s) If you are not the intended recipient, please do not use or disseminate the information, notify the sender and delete it from your system. ______________________________________________________________________

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[alsa-devel] Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG no sound
by PChot 21 Jun '10

21 Jun '10

Hi all, i'm still having problems with my audio card CA0110 which uses snd-hda-intel. I'l having same problems as other that is there is no sound. After searching on net i find that people are having pretty much the same problem http://bbs.archlinux.org/viewtopic.php?id=92769 https://bugzilla.novell.com/show_bug.cgi?id=564324 http://forums.gentoo.org/viewtopic-t-811741.html?sid=0dec2bfdd962d29f54f294… The best example how is not working is as i start mplayer. The program runs but then there is no playing video. So i could be that is not snd-hda-intel but some connected codecs. Now i'm wondering is anyone working on this case? Who shall i contact that works on snd-hda-intel or on Creative cards? So i can contact them and try to help if possible. Please someone response. Thanks, Jan

1 0

Re: [alsa-devel] PROBLEM: SIS7019 stops recording after 42 min
by Raymond Yau 21 Jun '10

21 Jun '10

2010/6/21 David Dillow <dave(a)thedillows.org> > On Mon, 2010-06-21 at 08:47 +0800, Raymond Yau wrote: > > 2010/6/20 David Dillow <dave(a)thedillows.org> > > > > > Please keep me on the cc list, I'll see your message sooner in my Inbox > > > than my mailing list folders. > > Please don't trim the cc list when replying. If you inadvertently hit > reply instead of reply-to-all, my apologies. > > > AFAIK , pulse device did not support MMAP > > Thanks, that's a pretty good clue he's using the hardware device. > Unfortunately, that means either I have bug in my driver or he's got too > much system noise and hitting overruns. > > 17:41:30.646532 access("/usr/share/alsa/cards/SiS7019.conf", R_OK) = -1 ENOENT (No such file or directory) There is no aliase for SiS7019 entry in /usr/share/alsa/cards/aliases.conf

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Re: [alsa-devel] Problem on wm8974 alsa driver
by Mark Brown 20 Jun '10

20 Jun '10

On Sun, Jun 20, 2010 at 01:48:03AM +0800, jjinfo(a)nudt.edu.cn wrote: > By now, we can confirm that the digital signal has arrived the wm8974 > DAI interface. As per wm8974 manual, it is required to set some 8974 > registers(such as enabling DAC, MONOOUT,etc) so as to get analogue > signal. > However, the codec driver (wm8974.c) does not provide any routine to > set those necessary registers. My question is, where and how are these What makes you say that the driver has no support for these things? The driver provides automatic power management and application control for audio routing and gains. This control should be fairly complete. For example, the DAC power is controlled by line 221: SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8974_POWER3, 0, 0), This will be enabled when a playback is active and the DAC is connected to an output. > outpu-related registers of audio codec chip got correctly set up? Are > they set up by aplay via calling alsa library APIs? Must we provide > additional help functions to do such initialization? aplay only deals with streaming PCM data to the CODEC. You should use an application such as alsamixer or amixer to configure the CODEC into the desired state at runtime.

1 0

[alsa-devel] no audio output with TYAN S7025 motherboard and Realtek ALC262
by Bryan Green 20 Jun '10

20 Jun '10

Hi, I have several systems with a TYAN S7025 motherboard and Realtek ALC262 chipset. I have not been able to produce any audio output. Well, that is except for one of the systems: after a variety of tweeks and tests, I got sound working on one system, but I do not know what change enabled sound to start working, and I have not been able to reproduce the effect on the other systems. I have tested with kernels up to 2.6.33, and with ALSA version 1.0.22.1. For a non-working system, 'alsaconf' gives the following output: Unknown hardware: "HDA-Intel" "Realtek ALC262" "HDA:10ec0262,10ec0262,00100202" "0x8086" "0x3a3e" And dmesg gives the following: HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 HDA Intel 0000:00:1b.0: irq 89 for MSI/MSI-X HDA Intel 0000:00:1b.0: setting latency timer to 64 hda_codec: ALC262: BIOS auto-probing. autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) hp_outs=0 (0x0/0x0/0x0/0x0/0x0) mono: mono_out=0x0 dig-out=0x1e/0x0 inputs: mic=0x18, fmic=0x19, line=0x1a, fline=0x0, cd=0x1c, aux=0x0 dig-in=0x1f realtek: No valid SSID, checking pincfg 0x411111f0 for NID 0x1d realtek: Enable default setup for auto mode as fallback input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input8 For the working system, the dmesg output is the same except for two lines: autoconfig: line_outs=1 (0x1b/0x0/0x0/0x0/0x0) inputs: mic=0x18, fmic=0x19, line=0x1a, fline=0x0, cd=0x1c, aux=0x14 Is there a method for tweaking these values on the non-working system to match the working system, to see if that works? I've uploaded the alsa-info output from a non-working system here: http://www.alsa-project.org/db/?f=91057bfc12790688f1d257743e53d5d0ec2ce1e2 I've uploaded the alsa-info output from the working system here: http://www.alsa-project.org/db/?f=198420f14cd0062713636408640636e6d474cf94 Thanks, -bryan

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Re: [alsa-devel] Status of the Tascam US-122mkII???
by jordan johnston 20 Jun '10

20 Jun '10

hey sorry, gmail defaults to only you when i reply, EVEN when i am selecting "reply all". i just have to add the Cc: manually :| ( i wasn't taking anyone out of Cc: ) sorry, i wasn't trying "violate" the rules or anything else, okay? 1. i understand that Zen-kernel has nothing to do with alsa, other than sometimes alsa drivers for certain devices are available in Zen before they are in my distro and many others i have used. i was only explaining what Zen-kernel is and why i use it. because you asked... makes sense? 2. I am glad that you were able to give me an update on the status of these devices, as the documentation around the interweb, is out-dated, and thus very often misleading. 3. I am going to go ahead and read all of the documentation you posted links to, and i will see if i can figure out how it works. 4. I am also going to contact Tascam/Teac, as i am a paying customer - why not :) obviously they are unlikely to give a lot of information away, but maybe i can get someone who knows just how different(or similar) the implementation of the Us-122L vs. mkII really is. either way, i don't mind sitting on the phone for an hour or so / or through email to let them know they should support linux or at the very least work with ALSA a little. it sounds like a lot of these companies could have really good support under linux by A: providing info on the implementation and B: by handing a device or two, into the hands of the right people(developers/hackers)...what a shame! thanks again for all of your help. It has been a slice! jordan

1 0

Re: [alsa-devel] Status of the Tascam US-122mkII???
by Daniel Mack 20 Jun '10

20 Jun '10

*** Reply to ALL please! *** On Sat, Jun 19, 2010 at 06:51:58PM -0400, jordan johnston wrote: > On Sat, Jun 19, 2010 at 6:41 PM, Daniel Mack <daniel(a)caiaq.de> wrote: > > Well, you can assume that audio is transported over dedicated > > isochronous endpoints. The question is how audio parameters are set up. > > And you can find out by tracing the traffic and see what happens when > > you start or stop the stream, when you change the sample rate etc. > > > > now, here you are referring, to tracing in OSX, or Windows right? (like before.) Yes. > i may try that out, but as i am not versed in this stuff it may be > quite difficult > without better understanding. > > do you know of any really insightful information on the > subject(somewhere on the net). > im going to google around a bit too. Hmm, I can point you to the USB specs which gives you detailed explanation about the low-level communication on the bus: http://www.usb.org/developers/docs/usb_20_052510.zip And you can read other Linux drivers and see what they do. We have no clue what Tascam's engineers did when they designed the MK2 version, and how close they kept to the protocol specs of their earlier products. Maybe they just shifted some bytes around, maybe they completely started over. All left for you to find out :) > Ya, it was a gamble i had called the store earlier in the week, they > actually had the device i wanted. > however, i didn't end up going that day. so when i did come, it had > been sold. i then decided to take > a gamble on it. > > no big deal. - 30 day return policy, and they said it was fine that i > returned it, if i couldn't get it > supported under linux. It might take you more than 30 days to fully support it under Linux if you're going to do it yourself though ;) Daniel

1 0

Re: [alsa-devel] Status of the Tascam US-122mkII???
by Tobias Hansen 20 Jun '10

20 Jun '10

jordan johnston wrote: > sorry about that....no more re-posting... > Again, reply to all addressees of the mails, now you only replied to me. > don't revert yet!!! > > The card IS half working, i can use midi no problem with jack... > > i think the driver DOES work..... i think the problem may be in > .asoundrc or some other alsa settings. > > right now i am using .asoundrc that i had picked up from here: > > # got from http://www.nabble.com/tascam-us122l-and-ccrma-td24036395.html > # thanks to Karsten > > # The usb_stream plugin configuration > > pcm.!usb_stream { > @args [ CARD ] > @args.CARD { > type string > default "0" > } > > type usb_stream > > card $CARD > } > > ctl.!usb_stream { > @args [ CARD ] > @args.CARD { > type string > default "0" > } > > type hw > > card $CARD > } > > i think the problem is that alsa/jack don't know where to look for > both the inputs and outputs (as midi works great). > unfortunately, i have little to no experience with this, and am not > sure how to proceed. > > it turns out that this card DOES use echi_hcd (i didn't unbind > uchi_hcd this time). > > (dmesg) > usb 1-5: new high speed USB device using ehci_hcd and address 3 > usb 1-5: config 1 interface 1 altsetting 1 bulk endpoint 0x83 has > invalid maxpacket 9 > usb 1-5: New USB device found, idVendor=0644, idProduct=8021 > usb 1-5: New USB device strings: Mfr=1, Product=2, SerialNumber=3 > usb 1-5: Product: US-122 MKII > usb 1-5: Manufacturer: TASCAM > usb 1-5: SerialNumber: no serial number > > as you can see echi_hcd is being used and..... > > [ninez@ninez ~]$ ls /sys/bus/pci/drivers/ehci_hcd > 0000:00:1d.7 bind module new_id remove_id uevent unbind > > the card is found when you look for it! > > [ninez@ninez ~]$ cat /proc/asound/cards > 0 [Intel ]: HDA-Intel - HDA Intel > HDA Intel at 0xefffc000 irq 27 > 1 [US122L ]: USB US-122L - TASCAM US-122L > TASCAM US-122L (644:8021 if 0 at 001/003) > > when i start Qjackctl up: > > jackdmp 1.9.5 > Copyright 2001-2005 Paul Davis and others. > Copyright 2004-2009 Grame. > jackdmp comes with ABSOLUTELY NO WARRANTY > This is free software, and you are welcome to redistribute it > under certain conditions; see the file COPYING for details > JACK server starting in realtime mode with priority 89 > audio_reservation_init > Acquire audio card Audio0 > creating alsa driver ... hw:0|-|128|3|44100|0|0|nomon|swmeter|-|32bit > Using ALSA driver HDA-Intel running on card 0 - HDA Intel at > 0xefffc000 irq 27 *** using internal sound (bah!) > configuring for 44100Hz, period = 128 frames (2.9 ms), buffer = 3 periods > ALSA: final selected sample format for playback: 32bit integer little-endian > ALSA: use 3 periods for playback > Using port names patch v0.1 (07.04.2010) > port created: Midi-Through:midi/playback_1 > port created: Midi-Through:midi/capture_1 > port created: TASCAM-US-122L:midi/playback_1 > *** it uses the midi just fine (i tested it with Native > intruments, massive) > port created: TASCAM-US-122L:midi/capture_1 > 15:46:49.327 ALSA connection change. > 15:46:50.351 Server configuration saved to "/home/ninez/.jackdrc". > 15:46:50.352 Statistics reset. > 15:46:50.367 Client activated. > 15:46:50.369 JACK connection change. > 15:46:50.374 JACK connection graph change. > > if i set it to hw:1 jack can't start, nor does it matter if i am > duplex, playback(only) or capture(only).. > so i am convinced this driver can work, otherwise midi wouldn't work > at all. the problem is my configuration. > but i am not sure how to remedy this, any clues would help! (let's not > give up quite yet!) > > it's finding out how to tell it to use outputs and inputs properly. > > jordan > MIDI works because it is done by another driver. Follow these instructions to see if you can get the US122 MKII working like a US122L: http://www.premiumorange.com/la-page-web-of-phil/index.php?page=P030101#con… The interface is not hw:0 but usb_stream:1. Also the US122L works with both ehci-hcd and uhci-hcd, the US144 only works with uhci-hcd. Good luck, others tried it before and it didn't work. Greetings, Tobias

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Re: [alsa-devel] Status of the Tascam US-122mkII???
by Daniel Mack 19 Jun '10

19 Jun '10

Hi, On Sat, Jun 19, 2010 at 02:30:03PM -0400, jordan johnston wrote: > On Sat, Jun 19, 2010 at 2:09 PM, Daniel Mack <daniel(a)caiaq.de> wrote: > > > (please don't remove any addresses from Cc:) > > im not sure that i follow what you mean here, can you explain??? Always use your email client's "reply-to-all" function, and do not just send replies to me. > *** no i did not modify the driver to match this, how do i do that??? Sorry, I didn't check properly. The driver clearly notes the vendor/product ID of your device, that's why it is probed but fails. This was added in commit 2b6f6c0d11 ("ALSA: snd-usb-us122l: add product IDs of US-122MKII and US-144MKII"). Obviously, this patch wasn't tested with real hardware (this is also what the commit log says), so it's no surprise it doesn't work for you. I copied the author, and I'd vote for reverting the patch until the driver is modified to actually support what it claims to do. > I set it to USB1.1 because i was under the impression that it will > not work with echi_hcd, > am i correct?? Well, according to the lsusb dump, the device is high-speed, so you shouldn't need any tricks. > my problem is that there really isn't any documentation that's > up-to-date for this unit, the newest info i could find > was a thread about the patches being submitted in December of last > year. but i figure by now that Zen should be up to date > with the changes made, so i am still in the dark about this unit's status.. Who is Zen? I think the answer to your question is quite simple but unfortunate: TASCAM changed the protocol of the device in a newer version, and the existing ALSA driver became incompatible with that. So there are only two possibilites to fix this: either TASCAM will send patches to the driver (which I think is unrealistic), or someone starts over and reverse-engineers the protocol in order to make the driver support it. As you happen to have the hardware - would you volunteer? :) > Do you have contact information for him?? i am not sure how to go > about the next steps you mention, > but i would be more than willing to do what i can, if i was explained > the process. > like i said before i am not a good programmer, but i am decent enough > with unix/linux and i do have XCode > for OSX - if that's what i need for tracing? Well, starting developing on the Linux kernel should be possible for you then, and help is much appreciated :) There's a lot of documentation around, and there is this and many other lists to get help on specific topics. Just read Documentation/SubmittingPatches before you send :) For tracing, you would either need a hardware analyzer (which you probaly don't have) or a software tool. For Windows, tools like http://www.usblyzer.com/ exist, but I never tried them myself. Maybe give that one a try and analyze which packets go over the bus. As there is no documentation, there is no other way than guessing what they're doing. This can be really time-consuming and frustrating, unfortunately. Good luck :) Daniel

2 1

Re: [alsa-devel] Status of the Tascam US-122mkII???
by Daniel Mack 19 Jun '10

19 Jun '10

(please don't remove any addresses from Cc:) On Sat, Jun 19, 2010 at 01:58:02PM -0400, jordan johnston wrote: > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 0 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 0 > bInterfaceProtocol 0 > iInterface 0 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 0 > bInterfaceProtocol 0 > iInterface 0 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 1 > bNumEndpoints 4 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 0 > bInterfaceProtocol 0 > iInterface 0 I just wanted to know whether they switched to class compliance interfaces, but as you can see here, that's not the case. > you will notice it is using usb 1.1 (i unbinded the usb2, as i am told > it doesn't work with it) How did you do this? And you modified the driver to match this device ID, right? > usb 4-1: new full speed USB device using uhci_hcd and address 3 > usb 4-1: New USB device found, idVendor=0644, idProduct=8021 > usb 4-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 > usb 4-1: Product: US-122 MKII > usb 4-1: Manufacturer: TASCAM > usb 4-1: SerialNumber: no serial number > ALSA sound/usb/usx2y/usb_stream.c:603: status=-2 > ALSA sound/usb/usx2y/usb_stream.c:603: status=-2 > ALSA sound/usb/usx2y/us122l.c:365: us122l_start error -14 > snd-usb-us122l: probe of 4-1:1.1 failed with error -22 > usbcore: registered new interface driver snd-usb-us122l Sorry, I can't help here. Maybe Karsten, die author of this driver, has any clue (put to Cc: now). Otherwise, you would need to trace the USB traffic using TASCAM's driver for Windows or MacOSX, and then reverse-engineer the protocol ... HTH, Daniel

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[alsa-devel] Status of the Tascam US-122mkII???
by jordan johnston 19 Jun '10

19 Jun '10

hello, Recently, i have gotten my hands on a Tascam US-122mkII. i have done a lot of reading/googling and i can see that the device has been added to the snd-usb-us122l and even-cross referenced my own device to make sure the Id and such were there. Currently, i am unable to get sound out of the Unit. it does however, power up on boot. I have tried many things to get it working, and i have even gotten Jack to give me the option of picking this Device, however it shows errors and does not work :( **** I am curious about the status of these drivers?? **** does anyone have this unit working at all? **** and the steps they used to get it working. ( as much of the info i have read doesn't seem quite uptodate ) is there possibly firmware for this unit or do i use the -US122L firmware instead? for Just a bit of background, I am using Fedora 12 with Planet CCRMA. however, i do not using the CCRMA kernel, as it doesn't cut it for me. A: it doesn't support a lot of newer hardware, B: it doesn't use MSI-X on my machine(shared USB/soundcard) C: It's bloated... D: no BFS scheduler. Instead i am using Zen-kernel which tends to be a lot more up to date, and i have gotten much better performance from Zen than any other kernel on any distro i have ever used. now, I am more than willing to provide any information that might help.... this is what i know so far. dmesg: ALSA sound/usb/usx2y/usb_stream.c:603: status=-2 ALSA sound/usb/usx2y/usb_stream.c:603: status=-2 ALSA sound/usb/usx2y/us122l.c:365: us122l_start error -14 snd-usb-us122l: probe of 1-5:1.1 failed with error -22 i have tried googling these messages to no avail. i poked around through the driver, i've tried some of the trouble-shooting tips provided by the guy who originally got the older models to work, removing echi-hcd, or unbinding it, unplugging it, etc. but nothing seems to be making a difference. Then my thought was that i need a proper asound.conf or .asoundrc. because the soundcard seems to be available, when i unbind it and unplug/plug the unit back in. i even tried removing them to see if on reboot, that they might be reconfigured, but no luck. if anyone can provide help, that would be greatly appreciated. I have a 30-day return, so if i can't figure it out, i will be returning it for a FFADO approved/fully-supported device. however that being said, wouldn't it be nice to still take the opportunity to have this device better supported or help debug some issues?? while i am not a really good programmer, i think i could be useful, as i have this device in front of me and can provide useful information to developers, just let me know what you need! Jordan

2 1

[alsa-devel] Bug in alsa-lib or alsamixer and amixer
by Raymond Yau 19 Jun '10

19 Jun '10

2010/6/16 Colin Guthrie <gmane(a)colin.guthr.ie> > 'Twas brillig, and Raymond Yau at 15/06/10 00:43 did gyre and gimble: > > +12dB(400%) is even larger than the software gain 150% of PA > > Software gain of PA 150% = ~+11dB, so not as different as you imply. > > I've explained the cubic mapping already, so please don't use arbitrary, > differently calculated percentages when comparing things. It's like > comparing apples to oranges. > > Col > > When I changed the volume slider of the gnome sound applet ( select perference ) in Fedora 13 to maximum (i.e. PA 150% ) Seem to be bug in amixer and alsamixer alsamixer also change to 150% but press any key change back to 100% The volume range is only from 0 to 65536 why did alsa-lib allow alsa-pulse plugin to set it outside the range , alsamixer and amixer also display 150% ? amixer -D pulse Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 65536 Mono: Front Left: Playback 98304 [150%] [on] Front Right: Playback 98304 [150%] [on]

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[alsa-devel] Update on TLV320AIC3204 Driver
by Stuart Longland 19 Jun '10

19 Jun '10

Hi all, I've managed to get significantly further along with a driver for the TLV320AIC3204, to the point now that I'm starting to look at further things to tidy up for its inclusion into the kernel. Some issues I've faced however: (1) When using the SOC_DOUBLE_R_SX_TLV controls, I notice there's an odd interaction between the mute switch associated with the control, and its corresponding gain setting. Nothing changes in the actual registers (except the mute bit of course) but the displayed gain shoots up to infinity if the mute is toggled when the gain associated with that mute control is set below 0dB. When the gain is at 0dB or above, toggling the mute has no effect on the displayed gain setting. Has anyone noticed this before and what would be the cause? For reference; the controls are defined in lines 395..414 of tlv320aic3204.c. (2) On the TLV320AIC3204, the clocks used on the audio bus can either be sourced from the ADC clocks, or the DAC clocks. Naturally, before you see any clocks on the bus, you must first power up either the DAC or ADC (or both), and choose the appropriate source. I do this currently in the hw_params callback (right where I *used* to power up the DACs/ADCs, DAPM does this now). The issue I see, is what happens when a playback stream starts, followed later by a recording stream. All will be fine until the recording stream stops -- if the playback is still going, I fear the clocks will get shut off with the DAC, and the recording will come to a grinding halt. Duplex will be okay since both recording and playback are synchronised events in this case... but I can see the potential for this alternate case causing problems. Therefore, I ask: Is there a way to control this via DAPM, so that, when the component supplying the bit clocks is shut down, the serial interface clock source is switched automatically to the alternate source? (i.e. if DAC is shut down, switch to using the ADC clocks) For reference; the switching presently occurs in lines 917..924 of tlv320aic3204.c. (3) The sysfs interface of my driver still remains. I had a look at using the ASoC debugfs interface, however there are a *lot* of registers on the 'AIC3204 in a sparse memory map. When I try to inspect the registers via /sys/kernel/debug/asoc/tlv320aic3204.0-0018/codec_reg ... I notice there are more registers in the device than are accessible via this interface: karo ~ # tail /sys/kernel/debug/asoc/tlv320aic3204.0-0018/codec_reg tlv320aic3204-i2c 0-0018: aic3204_write: pg 0 reg 0[0000] <= 01 tlv320aic3204-i2c 0-0018: aic3204_write: pg 0 reg 0[0000] <= 02 tlv320aic3204-i2c 0-0018: aic3204_write: pg 0 reg 0[0000] <= 03 tlv320aic3204-i2c 0-0018: aic3204_write: pg 0 reg 0[0000] <= 00 tlv320aic3204-i2c 0-0018: aic3204_write: pg 0 reg 0[0000] <= 01 tlv320aic3204-i2c 0-0018: aic3204_write: pg 0 reg 0[0000] <= 02 tlv320aic3204-i2c 0-0018: aic3204_write: pg 0 reg 0[0000] <= 03 1a7: 0 1a8: 0 1a9: 0 1aa: 0 1ab: 0 1ac: 0 1ad: 0 1ae: 0 1af: 0 1b0: karo ~ # Notice it stops at 0x01b0 (corresponding to page 3, register 48), with an incomplete line. There were more registers, than there was buffer space to write the file. Is there a way to increase this buffer? Or alternatively, should I perhaps port my sysfs interface to debugfs, and modify it to augment the existing debugfs interface? The driver is still using the registration model that was used in kernel 2.6.34. I've managed to backport the ALSA tree to 2.6.28 for our project (since that's what's officially supported by Ka-Ro) and so far, it's working, although things are still quite crude. Since my last contact I've managed to figure out some aspects of DAPM, for instance the DACs and ADCs are powered up and down by DAPM, whereas I used to do this when setting the PLL manually. I cache more of the registers now, I tried doing something "clever" and skipping the pages of registers that aren't used and trying to cache just the pages that were meaningful, but I noticed that the rest of ALSA more or less assumed the register address used with cache was the real address. Digging around on the mailing list about how to handle a sparse register map lead me to the solution of just biting the bullet and caching all 80-odd pages. Wasteful on memory, but I won't miss anything. The driver still has an initialisation script facility, whereby custom register settings can be made at init. I'm still undecided as to whether to keep this or not... it's not the "done thing", but if I leave it there, it allows for custom adaptive filtering coefficients and other parameters to be set by the machine driver, which may be useful in some applications. Is it worth cleaning this up, or should it be ditched before the driver is submitted for inclusion? The driver is accessible online at <http://www.longlandclan.yi.org/~stuartl/asoc/> along with some comments about how to add it to your kernel sources. I'd appreciate any feedback or advice on how the driver can be improved. Regards, -- Stuart Longland (aka Redhatter, VK4MSL) .'''. Gentoo Linux/MIPS Cobalt and Docs Developer '.'` : . . . . . . . . . . . . . . . . . . . . . . .'.' http://dev.gentoo.org/~redhatter :.' I haven't lost my mind... ...it's backed up on a tape somewhere.

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Re: [alsa-devel] [PATCH] ASoC: Add new TI TLV320AIC3204 CODEC driver
by Liam Girdwood 19 Jun '10

19 Jun '10

On Fri, 2010-06-18 at 13:57 +1000, Stuart Longland wrote: > The TLV320AIC3204 is a low-power stereo audio CODEC capable of sample > rates of up to 192kHz. This driver implements basic functionality, > using I²C for the control channel. > > The audio interface supports many data bus formats, including I²S > master/slave, DSP, left/right justified and TDM. > > What works: > - Playback at various bitrates up to 96kHz > - Recording at various bitrates up to 96kHz > - Mixer interface > - PLL generation of CODEC clocks from MCLK > > What could work better: > - DAPM > > What isn't tested: > - Audio interfaces other than I²S > - PLL with clocks other than ~12MHz > - Mono recording/playback > - 192kHz recording/playback > > What isn't implemented: > - SPI interface support > - PLL without fractional divider (would allow <10MHz clocks) > - Clock sources other than MCLK > - Adaptive filtering > - AGC > - Headset detection, JACK framework > > Signed-off-by: Stuart Longland <redhatter(a)gentoo.org> Just had a quick check and the register caching needs addressed. I agree with your comments, I don't think we really want to cache all 16K of codec registers here as we will probably only ever use a handful of them. I think a smaller lookup table containing only the registers that we care about will do. Fwiw, a generic ASoC lookup table would be best as this could be used by other codecs with large register maps too. The table should be ordered (for quick lookup) and also contain a readable/writeable/volatile flag for each register. Thanks Liam -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

3 4

[alsa-devel] Alsa 1.0.23 problems with VIA VT1708S
by William King 19 Jun '10

19 Jun '10

A few different issues with this system, but for devel and debugging I have 3 laptops of the exact same hardware. Asus k50AB laptops with the VIA VT1708S sound card. I am documenting as best as I can on this ubuntu bug ticket: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/595606 Currently the issues are: 1. Front mic does not work. No audio with any tests, even arecord test. 2. headphone jack does not work. When headphones are plugged in, the sound continues to come out of the speakers, and no sound out of the headphones. I am able to test patches easily. I have just installed from source 1.0.23 release.

1 0

Re: [alsa-devel] [Alsa-user] Crash 2.6.32.x, Echo 3G, Alsa, SCSI
by Giuliano Pochini 18 Jun '10

18 Jun '10

On Thu, 17 Jun 2010 08:11:06 -0400 "F. Heitkamp" <heitkamp(a)ameritech.net> wrote: > >> My goal is to be able to plug in a MIDI keyboard, or USB guitar > >> interface and play along with songs etc. > >> > >> I plugged in a MIDI keyboard, "/proc/asound/cards" showed the keyboard > >> was recognized. The keyboard has a USB and MIDI interface. > >> I am not sure if they both work at the same time. I had the MIDI cable > >> plugged into the Gina3G, but I could not tell if the keyboard was > >> recognized. > >> I compiled csound5. When I engaged it the system immediately froze and > >> started spitting out: > >> [...] > > Ok, I compiled csound, but I don't know how to use it. How can I reproduce > > the problem ? > > > Sorry I took so long to get back to you. > It turns out Csound is not related to the problem. > > In this case I am only using kernel 2.6.33.5-rt23-amd64, alsa lib 1.0.2, > but 2.6.34 does it too. > I bring up firefox. Web pages attempt to play sound. The kernel > begans doing: > > [ 2879.766331] allocate_pipes: ch=1 int=2 > [ 2879.766332] allocate_pipes: channel already open > [ 2879.766333] allocate_pipes(1) err=-11 > [ 2879.766334] pcm_hw_freed > [ 2.821190] pcm_analog_out_open cs=0 oc=2 r=44100 > [ 2879.821266] allocate_pipes: ch=1 int=2 > [ 2879.821267] allocate_pipes: channel already open > > ,infinitely. The system is unusable at this point and I have to > Alt-SysRq-k to get back in. > > Let me know what other information to provide. Perhaps is it related with the OSS emulation ? Do you know if the browser plays the sound itself or if it is using an external program or plugin ? -- Giuliano.

1 0

[alsa-devel] Cannot find
by subapriya sakthivel 18 Jun '10

18 Jun '10

Hay guys, I have download eclair branch code using below commands. repo init -u git://android.git.kernel.org/platform/manifest.git -b eclair repo sync But inside Hardware folder, i cannot find alsa_sound folder. Can anyone help me on this? Am i chekout from correct branch? If so, where can i get this folder. Thanks for your help in advance. -Priya

3 2

[PATCH] soc/imx: clean Kconfig
by Eric Bénard 18 Jun '10

18 Jun '10

Signed-off-by: Eric Bénard <eric(a)eukrea.com> --- sound/soc/imx/Kconfig | 10 ++-------- 1 files changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 6ef57e0..35e6cf6 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -8,22 +8,17 @@ config SND_IMX_SOC Say Y or M if you want to add support for codecs attached to the i.MX SSI interface. -config SND_MXC_SOC_SSI - tristate - config SND_MXC_SOC_WM1133_EV1 tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted" depends on SND_IMX_SOC && MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL select SND_SOC_WM8350 - select SND_MXC_SOC_SSI help Enable support for audio on the i.MX31ADS with the WM1133-EV1 PMIC board with WM8835x fitted. config SND_SOC_PHYCORE_AC97 tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards" - depends on MACH_PCM043 || MACH_PCA100 - select SND_MXC_SOC_SSI + depends on SND_IMX_SOC && (MACH_PCM043 || MACH_PCA100) select SND_SOC_WM9712 help Say Y if you want to add support for SoC audio on Phytec phyCORE @@ -31,8 +26,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 bool "Eukrea TLV320" - depends on MACH_EUKREA_MBIMX27_BASEBOARD || MACH_EUKREA_MBIMXSD_BASEBOARD - select SND_IMX_SOC + depends on SND_IMX_SOC && (MACH_EUKREA_MBIMX27_BASEBOARD || MACH_EUKREA_MBIMXSD_BASEBOARD) select SND_SOC_TLV320AIC23 help Enable I2S based access to the TLV320AIC23B codec attached -- 1.6.3.3 --------------020001010209060103040907--

1 0

Re: [alsa-devel] Contents of Alsa-devel digest...
by Junhwa Rhee 18 Jun '10

18 Jun '10

On Jun 18, 2010, at 11:38 AM, alsa-devel-request(a)alsa-project.org wrote: > Send Alsa-devel mailing list submissions to > alsa-devel(a)alsa-project.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > or, via email, send a message with subject or body 'help' to > alsa-devel-request(a)alsa-project.org > > You can reach the person managing the list at > alsa-devel-owner(a)alsa-project.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Alsa-devel digest..." > > > Today's Topics: > > 1. Re: [PATCH 2/5] ALSA: usb-audio: unify UAC macros and struct > names (Daniel Mack) > 2. Re: Timing Info (Clemens Ladisch) > 3. Re: [PATCH] ASoC: pandora: fix CLKX polarity (Mark Brown) > 4. [PATCH] ALSA: hda - add ideapad model for Conexant 5051 codec > (Herton Ronaldo Krzesinski) > 5. Re: [PATCH] ALSA: hda - add ideapad model for Conexant 5051 > codec (Takashi Iwai) > 6. Re: snd_mixer_selem_get_playback_volume return value outside > allowed range (Raymond Yau) > 7. Re: [PATCH] eukrea-tlv320: add support for our i.MX25 board > (Mark Brown) > 8. SOC_DOUBLE_R_SX_TLV (was Update on TLV320AIC3204 Driver) > (Stuart Longland) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 17 Jun 2010 17:26:26 +0200 > From: Daniel Mack <daniel(a)caiaq.de> > Subject: Re: [alsa-devel] [PATCH 2/5] ALSA: usb-audio: unify UAC > macros and struct names > To: Alex Lee <alexlee188(a)gmail.com> > Cc: Takashi Iwai <tiwai(a)suse.de>, "alsa-devel(a)alsa-project.org" > <alsa-devel(a)alsa-project.org>, "clemens(a)ladisch.de" > <clemens(a)ladisch.de> > Message-ID: <20100617152626.GL17833(a)buzzloop.caiaq.de> > Content-Type: text/plain; charset=us-ascii > > (hint, hint: don't top-post :)) > > On Thu, Jun 17, 2010 at 08:21:22AM -0700, Alex Lee wrote: >> It will not help existing codes, I agree. But it may make writing new UAC2 >> specific code easier? > > Not as long as most of the code stays hybrid for both versions. If there > were versions of these enum values for both UAC1 and UAC2, you would > need to choose one of them in the parser codes. Which one you go for > doesn't actually make any difference as they're both the same, but the > code would read as if it was specific to any version. > > I'd say this would add more to the confusion, and not ease any. > > > Daniel > > > > ------------------------------ > > Message: 2 > Date: Thu, 17 Jun 2010 17:36:59 +0200 > From: Clemens Ladisch <clemens(a)ladisch.de> > Subject: Re: [alsa-devel] Timing Info > To: Paul Dugas <paul(a)dugasenterprises.com> > Cc: alsa-devel(a)alsa-project.org > Message-ID: <4C1A411B.2060301(a)ladisch.de> > Content-Type: text/plain; charset=us-ascii > > Paul Dugas wrote: >> On Sun, Jun 13, 2010 at 10:10 AM, Clemens Ladisch <clemens(a)ladisch.de> wrote: >>> Set SND_PCM_TSTAMP_ENABLE to generate a timestamp whenever the hardware >>> position is updated. Call snd_pcm_status to read the position and the >>> corresponding timestamp. >> >> I've re-read this a number of times and have been fiddling with code >> that dumps the status at various points but I'm still not grasping the >> concept here. Can comeone point me to an explanation of the delay, >> avail, and avail_max values? > > The avail value tells you how many frames are available, i.e., can be > written, in the output buffer. The largest value that this had since > the last snd_pcm_status call is avail_max. > > The delay value is the time that will elapse before the next frame that > you write into the buffer will actually be output by the hardware. This > is not only the time until the hardware reads this frame out of the > buffer (which could be computed from avail) but also any processing > after that. > > > HTH > Clemens > > > ------------------------------ > > Message: 3 > Date: Thu, 17 Jun 2010 16:50:01 +0100 > From: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > Subject: Re: [alsa-devel] [PATCH] ASoC: pandora: fix CLKX polarity > To: Grazvydas Ignotas <notasas(a)gmail.com>, alsa-devel(a)alsa-project.org > Cc: lrg(a)slimlogic.co.uk > Message-ID: <u6g5khs6pbrq63fngu2ncjvv.1276789801383(a)email.android.com> > Content-Type: text/plain; charset=utf-8 > > Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > > Probably best to add a CC to stable too. Sorry about the rubbish formatting, on my phone and it can't do any better. > > Grazvydas Ignotas <notasas(a)gmail.com> wrote: > >> After mass production started it was found that several boards exhibit >> noise problems during sound playback. After some investigation it was >> determined that CLKX polarity is set incorrectly, and even if most boards >> can tolerate the wrong setting, there are some that don't. >> >> Fix polarity setup in the board file. As the clock settings for input and >> output now match, merge in and out functions and structures to simplify >> code. >> >> Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> >> --- >> sound/soc/omap/omap3pandora.c | 36 ++++++++---------------------------- >> 1 files changed, 8 insertions(+), 28 deletions(-) >> >> diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c >> index 87ce842..9eecac1 100644 >> --- a/sound/soc/omap/omap3pandora.c >> +++ b/sound/soc/omap/omap3pandora.c >> @@ -43,12 +43,14 @@ >> >> static struct regulator *omap3pandora_dac_reg; >> >> -static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream, >> - struct snd_pcm_hw_params *params, unsigned int fmt) >> +static int omap3pandora_hw_params(struct snd_pcm_substream *substream, >> + struct snd_pcm_hw_params *params) >> { >> struct snd_soc_pcm_runtime *rtd = substream->private_data; >> struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; >> struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; >> + int fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | >> + SND_SOC_DAIFMT_CBS_CFS; >> int ret; >> >> /* Set codec DAI configuration */ >> @@ -91,24 +93,6 @@ static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream, >> return 0; >> } >> >> -static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream, >> - struct snd_pcm_hw_params *params) >> -{ >> - return omap3pandora_cmn_hw_params(substream, params, >> - SND_SOC_DAIFMT_I2S | >> - SND_SOC_DAIFMT_IB_NF | >> - SND_SOC_DAIFMT_CBS_CFS); >> -} >> - >> -static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream, >> - struct snd_pcm_hw_params *params) >> -{ >> - return omap3pandora_cmn_hw_params(substream, params, >> - SND_SOC_DAIFMT_I2S | >> - SND_SOC_DAIFMT_NB_NF | >> - SND_SOC_DAIFMT_CBS_CFS); >> -} >> - >> static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w, >> struct snd_kcontrol *k, int event) >> { >> @@ -231,12 +215,8 @@ static int omap3pandora_in_init(struct snd_soc_codec *codec) >> return snd_soc_dapm_sync(codec); >> } >> >> -static struct snd_soc_ops omap3pandora_out_ops = { >> - .hw_params = omap3pandora_out_hw_params, >> -}; >> - >> -static struct snd_soc_ops omap3pandora_in_ops = { >> - .hw_params = omap3pandora_in_hw_params, >> +static struct snd_soc_ops omap3pandora_ops = { >> + .hw_params = omap3pandora_hw_params, >> }; >> >> /* Digital audio interface glue - connects codec <--> CPU */ >> @@ -246,14 +226,14 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { >> .stream_name = "HiFi Out", >> .cpu_dai = &omap_mcbsp_dai[0], >> .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], >> - .ops = &omap3pandora_out_ops, >> + .ops = &omap3pandora_ops, >> .init = omap3pandora_out_init, >> }, { >> .name = "TWL4030", >> .stream_name = "Line/Mic In", >> .cpu_dai = &omap_mcbsp_dai[1], >> .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], >> - .ops = &omap3pandora_in_ops, >> + .ops = &omap3pandora_ops, >> .init = omap3pandora_in_init, >> } >> }; >> -- >> 1.6.3.3 >> > > ------------------------------ > > Message: 4 > Date: Thu, 17 Jun 2010 14:15:06 -0300 > From: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> > Subject: [alsa-devel] [PATCH] ALSA: hda - add ideapad model for > Conexant 5051 codec > To: alsa-devel(a)alsa-project.org > Cc: Takashi Iwai <tiwai(a)suse.de> > Message-ID: <201006171415.07101.herton(a)mandriva.com.br> > Content-Type: Text/Plain; charset="us-ascii" > > Lenovo IdeaPad Y430 has an additional subwoofer connected at pin 0x1b, > which isn't muted when headphone is plugged in. This adds additional > support to the extra subwoofer via new ideapad model. > > Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> > --- > Documentation/sound/alsa/HD-Audio-Models.txt | 1 + > sound/pci/hda/patch_conexant.c | 20 ++++++++++++++++++++ > 2 files changed, 21 insertions(+), 0 deletions(-) > > diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt > b/Documentation/sound/alsa/HD-Audio-Models.txt > index 1d38b0d..84e81ad 100644 > --- a/Documentation/sound/alsa/HD-Audio-Models.txt > +++ b/Documentation/sound/alsa/HD-Audio-Models.txt > @@ -282,6 +282,7 @@ Conexant 5051 > hp HP Spartan laptop > hp-dv6736 HP dv6736 > hp-f700 HP Compaq Presario F700 > + ideapad Lenovo IdeaPad laptop > lenovo-x200 Lenovo X200 laptop > toshiba Toshiba Satellite M300 > > diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c > index 2bf2cb5..54f7419 100644 > --- a/sound/pci/hda/patch_conexant.c > +++ b/sound/pci/hda/patch_conexant.c > @@ -1632,6 +1632,11 @@ static void cxt5051_update_speaker(struct hda_codec > *codec) > pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; > snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, > pinctl); > + /* on ideapad there is an aditional speaker (subwoofer) to mute */ > + if (spec->ideapad) > + snd_hda_codec_write(codec, 0x1b, 0, > + AC_VERB_SET_PIN_WIDGET_CONTROL, > + pinctl); > } > > /* turn on/off EAPD (+ mute HP) as a master switch */ > @@ -1888,6 +1893,13 @@ static void cxt5051_init_mic_port(struct hda_codec > *codec, hda_nid_t nid, > #endif > } > > +static struct hda_verb cxt5051_ideapad_init_verbs[] = { > + /* Subwoofer */ > + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, > + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, > + { } /* end */ > +}; > + > /* initialize jack-sensing, too */ > static int cxt5051_init(struct hda_codec *codec) > { > @@ -1917,6 +1929,7 @@ enum { > CXT5051_LENOVO_X200, /* Lenovo X200 laptop, also used for Advanced Mini > Dock 250410 */ > CXT5051_F700, /* HP Compaq Presario F700 */ > CXT5051_TOSHIBA, /* Toshiba M300 & co */ > + CXT5051_IDEAPAD, /* Lenovo IdeaPad Y430 */ > CXT5051_MODELS > }; > > @@ -1927,6 +1940,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = { > [CXT5051_LENOVO_X200] = "lenovo-x200", > [CXT5051_F700] = "hp-700", > [CXT5051_TOSHIBA] = "toshiba", > + [CXT5051_IDEAPAD] = "ideapad", > }; > > static struct snd_pci_quirk cxt5051_cfg_tbl[] = { > @@ -1938,6 +1952,7 @@ static struct snd_pci_quirk cxt5051_cfg_tbl[] = { > CXT5051_LAPTOP), > SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), > SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200), > + SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo IdeaPad", CXT5051_IDEAPAD), > {} > }; > > @@ -1999,6 +2014,11 @@ static int patch_cxt5051(struct hda_codec *codec) > spec->mixers[0] = cxt5051_toshiba_mixers; > spec->auto_mic = AUTO_MIC_PORTB; > break; > + case CXT5051_IDEAPAD: > + spec->init_verbs[spec->num_init_verbs++] = > + cxt5051_ideapad_init_verbs; > + spec->ideapad = 1; > + break; > } > > return 0; > -- > 1.7.1 > > > > ------------------------------ > > Message: 5 > Date: Thu, 17 Jun 2010 20:39:20 +0200 > From: Takashi Iwai <tiwai(a)suse.de> > Subject: Re: [alsa-devel] [PATCH] ALSA: hda - add ideapad model for > Conexant 5051 codec > To: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> > Cc: alsa-devel(a)alsa-project.org > Message-ID: <s5hk4pxld0n.wl%tiwai(a)suse.de> > Content-Type: text/plain; charset=US-ASCII > > At Thu, 17 Jun 2010 14:15:06 -0300, > Herton Ronaldo Krzesinski wrote: >> >> Lenovo IdeaPad Y430 has an additional subwoofer connected at pin 0x1b, >> which isn't muted when headphone is plugged in. This adds additional >> support to the extra subwoofer via new ideapad model. >> >> Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> > > Thanks, applied now. > > > Takashi > >> --- >> Documentation/sound/alsa/HD-Audio-Models.txt | 1 + >> sound/pci/hda/patch_conexant.c | 20 ++++++++++++++++++++ >> 2 files changed, 21 insertions(+), 0 deletions(-) >> >> diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt >> b/Documentation/sound/alsa/HD-Audio-Models.txt >> index 1d38b0d..84e81ad 100644 >> --- a/Documentation/sound/alsa/HD-Audio-Models.txt >> +++ b/Documentation/sound/alsa/HD-Audio-Models.txt >> @@ -282,6 +282,7 @@ Conexant 5051 >> hp HP Spartan laptop >> hp-dv6736 HP dv6736 >> hp-f700 HP Compaq Presario F700 >> + ideapad Lenovo IdeaPad laptop >> lenovo-x200 Lenovo X200 laptop >> toshiba Toshiba Satellite M300 >> >> diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c >> index 2bf2cb5..54f7419 100644 >> --- a/sound/pci/hda/patch_conexant.c >> +++ b/sound/pci/hda/patch_conexant.c >> @@ -1632,6 +1632,11 @@ static void cxt5051_update_speaker(struct hda_codec >> *codec) >> pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; >> snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, >> pinctl); >> + /* on ideapad there is an aditional speaker (subwoofer) to mute */ >> + if (spec->ideapad) >> + snd_hda_codec_write(codec, 0x1b, 0, >> + AC_VERB_SET_PIN_WIDGET_CONTROL, >> + pinctl); >> } >> >> /* turn on/off EAPD (+ mute HP) as a master switch */ >> @@ -1888,6 +1893,13 @@ static void cxt5051_init_mic_port(struct hda_codec >> *codec, hda_nid_t nid, >> #endif >> } >> >> +static struct hda_verb cxt5051_ideapad_init_verbs[] = { >> + /* Subwoofer */ >> + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, >> + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, >> + { } /* end */ >> +}; >> + >> /* initialize jack-sensing, too */ >> static int cxt5051_init(struct hda_codec *codec) >> { >> @@ -1917,6 +1929,7 @@ enum { >> CXT5051_LENOVO_X200, /* Lenovo X200 laptop, also used for Advanced Mini >> Dock 250410 */ >> CXT5051_F700, /* HP Compaq Presario F700 */ >> CXT5051_TOSHIBA, /* Toshiba M300 & co */ >> + CXT5051_IDEAPAD, /* Lenovo IdeaPad Y430 */ >> CXT5051_MODELS >> }; >> >> @@ -1927,6 +1940,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = { >> [CXT5051_LENOVO_X200] = "lenovo-x200", >> [CXT5051_F700] = "hp-700", >> [CXT5051_TOSHIBA] = "toshiba", >> + [CXT5051_IDEAPAD] = "ideapad", >> }; >> >> static struct snd_pci_quirk cxt5051_cfg_tbl[] = { >> @@ -1938,6 +1952,7 @@ static struct snd_pci_quirk cxt5051_cfg_tbl[] = { >> CXT5051_LAPTOP), >> SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), >> SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200), >> + SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo IdeaPad", CXT5051_IDEAPAD), >> {} >> }; >> >> @@ -1999,6 +2014,11 @@ static int patch_cxt5051(struct hda_codec *codec) >> spec->mixers[0] = cxt5051_toshiba_mixers; >> spec->auto_mic = AUTO_MIC_PORTB; >> break; >> + case CXT5051_IDEAPAD: >> + spec->init_verbs[spec->num_init_verbs++] = >> + cxt5051_ideapad_init_verbs; >> + spec->ideapad = 1; >> + break; >> } >> >> return 0; >> -- >> 1.7.1 >> > > > ------------------------------ > > Message: 6 > Date: Fri, 18 Jun 2010 07:39:07 +0800 > From: Raymond Yau <superquad.vortex2(a)gmail.com> > Subject: Re: [alsa-devel] snd_mixer_selem_get_playback_volume return > value outside allowed range > To: ALSA Development Mailing List <alsa-devel(a)alsa-project.org> > Message-ID: > <AANLkTin7y2JzkaCYb4p69uDfWOo4cTa85in0Q3l7RwbD(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > 2010/6/17 Clemens Ladisch <clemens(a)ladisch.de> > >> Raymond Yau wrote: >>> Should snd_mixer_selem_get_playback_volume() perform range check ? >> >> No, it's the responsibility of the control implementation to return >> valid values (and to check that values that are being set are valid). >> >> >> Regards, >> Clemens >> > > If it is the responsibility of the driver/external plugin to keep the value > inside the allowed range > > your patch actually help those buggy driver/external plugin hide the bug > only > > > ------------------------------ > > Message: 7 > Date: Fri, 18 Jun 2010 01:56:37 +0100 > From: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > Subject: Re: [alsa-devel] [PATCH] eukrea-tlv320: add support for our > i.MX25 board > To: Eric B?nard <eric(a)eukrea.com> > Cc: alsa-devel(a)alsa-project.org, s.hauer(a)pengutronix.de, > lrg(a)slimlogic.co.uk > Message-ID: <20100618005637.GA3333(a)opensource.wolfsonmicro.com> > Content-Type: text/plain; charset=iso-8859-1 > > On Thu, Jun 17, 2010 at 03:44:01PM +0200, Eric B?nard wrote: >> * tdm slot has to be configured to get sound working on i.MX25 >> >> Signed-off-by: Eric B?nard <eric(a)eukrea.com> > > Applied but I had to hand apply the Kconfig update, please check that it > applied correctly. > > > ------------------------------ > > Message: 8 > Date: Fri, 18 Jun 2010 12:38:10 +1000 > From: Stuart Longland <redhatter(a)gentoo.org> > Subject: [alsa-devel] SOC_DOUBLE_R_SX_TLV (was Update on TLV320AIC3204 > Driver) > To: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > Cc: Takashi Iwai <tiwai(a)suse.de>, alsa-devel(a)alsa-project.org, Eric > B??nard <eric(a)eukrea.com>, Liam Girdwood <lrg(a)slimlogic.co.uk> > Message-ID: <20100618023810.GS7759(a)www.longlandclan.yi.org> > Content-Type: text/plain; charset=us-ascii > > On Tue, Jun 15, 2010 at 03:11:10PM +1000, Stuart Longland wrote: >> When a read is performed, this structure points to this function as the >> means for reading the register (defined in sound/soc-core.c): >> >> /** >> * snd_soc_get_volsw_2r_sx - double with tlv and variable data size >> * mixer get callback >> * @kcontrol: mixer control >> * @uinfo: control element information >> * >> * Returns 0 for success. >> */ >> int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol, >> struct snd_ctl_elem_value *ucontrol) >> { >> struct soc_mixer_control *mc = >> (struct soc_mixer_control *)kcontrol->private_value; >> struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); >> unsigned int mask = (1<<mc->shift)-1; >> int min = mc->min; >> int val = snd_soc_read(codec, mc->reg) & mask; >> int valr = snd_soc_read(codec, mc->rreg) & mask; >> >> ucontrol->value.integer.value[0] = ((val & 0xff)-min); >> ucontrol->value.integer.value[1] = ((valr & 0xff)-min); >> return 0; >> } >> EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r_sx); > > I've figured it out now... and there'll be a patch to fix it. > > The mute <--> gain interaction was pure coincidence as it turns out. > There's no interaction between the two, just that when I toggle the > mute, the gain gets re-read, and hence it *appeared* to interact. > > Notice the value calculated for the channel is: > > output = ( regval & 0xff ) - minimum > > In my case, my minimum is -6. Suppose I set my driver gain to -1dB, in > hexadecimal, 0x3f. Since no sign extension is done, (regval & 0xff) = > 63. Adding 6 to this (double negation) you get 69... which is greater > than the maximum gain of 29. Hence the scale shoots up to maximum. > > The fix, is to change the above so that they mask the result; thus > wrapping around the zero point like so: > > output = ( ( regval & 0xff ) - minimum ) & mask > > This fixes the problem that I observe, and shouldn't break the other > user of the control. A patch to fix this issue is coming. > -- > Stuart Longland (aka Redhatter, VK4MSL) .'''. > Gentoo Linux/MIPS Cobalt and Docs Developer '.'` : > . . . . . . . . . . . . . . . . . . . . . . .'.' > http://dev.gentoo.org/~redhatter :.' > > I haven't lost my mind... > ...it's backed up on a tape somewhere. > > > ------------------------------ > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > End of Alsa-devel Digest, Vol 40, Issue 80 > ******************************************

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[alsa-devel] [PATCH] ALSA: hda - add ideapad model for Conexant 5051 codec
by Herton Ronaldo Krzesinski 17 Jun '10

17 Jun '10

Lenovo IdeaPad Y430 has an additional subwoofer connected at pin 0x1b, which isn't muted when headphone is plugged in. This adds additional support to the extra subwoofer via new ideapad model. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_conexant.c | 20 ++++++++++++++++++++ 2 files changed, 21 insertions(+), 0 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 1d38b0d..84e81ad 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -282,6 +282,7 @@ Conexant 5051 hp HP Spartan laptop hp-dv6736 HP dv6736 hp-f700 HP Compaq Presario F700 + ideapad Lenovo IdeaPad laptop lenovo-x200 Lenovo X200 laptop toshiba Toshiba Satellite M300 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 2bf2cb5..54f7419 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1632,6 +1632,11 @@ static void cxt5051_update_speaker(struct hda_codec *codec) pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl); + /* on ideapad there is an aditional speaker (subwoofer) to mute */ + if (spec->ideapad) + snd_hda_codec_write(codec, 0x1b, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pinctl); } /* turn on/off EAPD (+ mute HP) as a master switch */ @@ -1888,6 +1893,13 @@ static void cxt5051_init_mic_port(struct hda_codec *codec, hda_nid_t nid, #endif } +static struct hda_verb cxt5051_ideapad_init_verbs[] = { + /* Subwoofer */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5051_init(struct hda_codec *codec) { @@ -1917,6 +1929,7 @@ enum { CXT5051_LENOVO_X200, /* Lenovo X200 laptop, also used for Advanced Mini Dock 250410 */ CXT5051_F700, /* HP Compaq Presario F700 */ CXT5051_TOSHIBA, /* Toshiba M300 & co */ + CXT5051_IDEAPAD, /* Lenovo IdeaPad Y430 */ CXT5051_MODELS }; @@ -1927,6 +1940,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = { [CXT5051_LENOVO_X200] = "lenovo-x200", [CXT5051_F700] = "hp-700", [CXT5051_TOSHIBA] = "toshiba", + [CXT5051_IDEAPAD] = "ideapad", }; static struct snd_pci_quirk cxt5051_cfg_tbl[] = { @@ -1938,6 +1952,7 @@ static struct snd_pci_quirk cxt5051_cfg_tbl[] = { CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200), + SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo IdeaPad", CXT5051_IDEAPAD), {} }; @@ -1999,6 +2014,11 @@ static int patch_cxt5051(struct hda_codec *codec) spec->mixers[0] = cxt5051_toshiba_mixers; spec->auto_mic = AUTO_MIC_PORTB; break; + case CXT5051_IDEAPAD: + spec->init_verbs[spec->num_init_verbs++] = + cxt5051_ideapad_init_verbs; + spec->ideapad = 1; + break; } return 0; -- 1.7.1

2 1

Re: [alsa-devel] [PATCH] ASoC: pandora: fix CLKX polarity
by Mark Brown 17 Jun '10

17 Jun '10

Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Probably best to add a CC to stable too. Sorry about the rubbish formatting, on my phone and it can't do any better. Grazvydas Ignotas <notasas(a)gmail.com> wrote: >After mass production started it was found that several boards exhibit >noise problems during sound playback. After some investigation it was >determined that CLKX polarity is set incorrectly, and even if most boards >can tolerate the wrong setting, there are some that don't. > >Fix polarity setup in the board file. As the clock settings for input and >output now match, merge in and out functions and structures to simplify >code. > >Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> >--- > sound/soc/omap/omap3pandora.c | 36 ++++++++---------------------------- > 1 files changed, 8 insertions(+), 28 deletions(-) > >diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c >index 87ce842..9eecac1 100644 >--- a/sound/soc/omap/omap3pandora.c >+++ b/sound/soc/omap/omap3pandora.c >@@ -43,12 +43,14 @@ > > static struct regulator *omap3pandora_dac_reg; > >-static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream, >- struct snd_pcm_hw_params *params, unsigned int fmt) >+static int omap3pandora_hw_params(struct snd_pcm_substream *substream, >+ struct snd_pcm_hw_params *params) > { > struct snd_soc_pcm_runtime *rtd = substream->private_data; > struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; > struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; >+ int fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | >+ SND_SOC_DAIFMT_CBS_CFS; > int ret; > > /* Set codec DAI configuration */ >@@ -91,24 +93,6 @@ static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream, > return 0; > } > >-static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream, >- struct snd_pcm_hw_params *params) >-{ >- return omap3pandora_cmn_hw_params(substream, params, >- SND_SOC_DAIFMT_I2S | >- SND_SOC_DAIFMT_IB_NF | >- SND_SOC_DAIFMT_CBS_CFS); >-} >- >-static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream, >- struct snd_pcm_hw_params *params) >-{ >- return omap3pandora_cmn_hw_params(substream, params, >- SND_SOC_DAIFMT_I2S | >- SND_SOC_DAIFMT_NB_NF | >- SND_SOC_DAIFMT_CBS_CFS); >-} >- > static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w, > struct snd_kcontrol *k, int event) > { >@@ -231,12 +215,8 @@ static int omap3pandora_in_init(struct snd_soc_codec *codec) > return snd_soc_dapm_sync(codec); > } > >-static struct snd_soc_ops omap3pandora_out_ops = { >- .hw_params = omap3pandora_out_hw_params, >-}; >- >-static struct snd_soc_ops omap3pandora_in_ops = { >- .hw_params = omap3pandora_in_hw_params, >+static struct snd_soc_ops omap3pandora_ops = { >+ .hw_params = omap3pandora_hw_params, > }; > > /* Digital audio interface glue - connects codec <--> CPU */ >@@ -246,14 +226,14 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { > .stream_name = "HiFi Out", > .cpu_dai = &omap_mcbsp_dai[0], > .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], >- .ops = &omap3pandora_out_ops, >+ .ops = &omap3pandora_ops, > .init = omap3pandora_out_init, > }, { > .name = "TWL4030", > .stream_name = "Line/Mic In", > .cpu_dai = &omap_mcbsp_dai[1], > .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], >- .ops = &omap3pandora_in_ops, >+ .ops = &omap3pandora_ops, > .init = omap3pandora_in_init, > } > }; >-- >1.6.3.3 >

1 0

[alsa-devel] Timing Info
by Paul Dugas 17 Jun '10

17 Jun '10

I'm using snd_pcm_readi() to collect samples and am wondering if/how I can get accurate timestamps for them. Seems like the snd_pcm_status() gives me this info but I'm unclear on the specifics. Can someone point me toward the correct field in the status structure that I should be using and what sample that timestamp corresponds to? Does snd_pcm_status_get_tstamp() get me the timestamp for the first frame returned by the next snd_pcm_readi() call? P

2 6

Re: [alsa-devel] [PATCH 2/5] ALSA: usb-audio: unify UAC macros and struct names
by Daniel Mack 17 Jun '10

17 Jun '10

(hint, hint: don't top-post :)) On Thu, Jun 17, 2010 at 08:21:22AM -0700, Alex Lee wrote: > It will not help existing codes, I agree. But it may make writing new UAC2 > specific code easier? Not as long as most of the code stays hybrid for both versions. If there were versions of these enum values for both UAC1 and UAC2, you would need to choose one of them in the parser codes. Which one you go for doesn't actually make any difference as they're both the same, but the code would read as if it was specific to any version. I'd say this would add more to the confusion, and not ease any. Daniel

1 0

[alsa-devel] error code 130: io_open - ALSA open error
by S. Aguinaga 17 Jun '10

17 Jun '10

Hello Folks, I tried finding what this error code means, but my searches come up empty. Would any of you point me to the location where these error codes are described or shoot me an email with a brief explanation. // Sal Aguinaga // Northwestern University

2 1

Re: [alsa-devel] [PATCH 2/5] ALSA: usb-audio: unify UAC macros and struct names
by Daniel Mack 17 Jun '10

17 Jun '10

On Wed, Jun 16, 2010 at 01:31:17PM -0700, Alex Lee wrote: > It will be painful, but what about defining a whole new set of > UAC2_xxx constants even if it is the same value as UAC1 ? That doesn't really help much, as the code using these macros is hybrid, for both versions. Just grep for them in the sources to see what I mean :) Daniel > On 16-Jun-2010, at 1:21 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > >At Wed, 16 Jun 2010 19:40:44 +0200, > >Daniel Mack wrote: > >> > >>On Wed, Jun 16, 2010 at 07:34:49PM +0200, Takashi Iwai wrote: > >>>At Wed, 16 Jun 2010 17:57:28 +0200, > >>>Daniel Mack wrote: > >>>> > >>>>diff --git a/include/linux/usb/audio.h b/include/linux/usb/audio.h > >>>>index c51200c..a54b825 100644 > >>>>--- a/include/linux/usb/audio.h > >>>>+++ b/include/linux/usb/audio.h > >>>>@@ -39,8 +39,8 @@ > >>>>#define UAC_MIXER_UNIT 0x04 > >>>>#define UAC_SELECTOR_UNIT 0x05 > >>>>#define UAC_FEATURE_UNIT 0x06 > >>>>-#define UAC_PROCESSING_UNIT_V1 0x07 > >>>>-#define UAC_EXTENSION_UNIT_V1 0x08 > >>>>+#define UAC1_PROCESSING_UNIT 0x07 > >>>>+#define UAC1_EXTENSION_UNIT 0x08 > >>> > >>>So now we have mixed prefix here, UAC_ and UAC1_. > >>>Isn't it a bit confusing, too? > >>> > >>>Honestly, I have no much preference about this name-ruling. > >>>But it's of course better if it's stabilized :) > >> > >>Well yeah, I hate that too, especially as it is a matter of taste > >>eventually. > > > >Indeed, it's just a matter of taste. > > > >>However, the idea is: things that are common for both UAC1 > >>and UAC2 are prefixed with UAC_, and only those things that are > >>special > >>get a number suffix. Which is the case in the block you quoted above. > > > >Yeah, that I understood. It's just that I feel something not clear > >around this... It might be simply because of the salad I ate today, > >though. But I'd like to hear opinions of others before merging. > >If nothing comes up, I'm willing to apply as is. > > > >>(This perticular detail is really the greatest unnecessary > >>confusion in > >>the UAC2 spec, btw. They just drop one enumeration value and shuffled > >>two others around for no obvious reason. Now we have to live > >>with that.) > > > >There are always enough examples how to behave rude :) > > > > > >thanks, > > > >Takashi

1 0

[alsa-devel] [PATCH] alsa: riptide: don't use own hex_to_bin() method
by Andy Shevchenko 17 Jun '10

17 Jun '10

From: Andy Shevchenko <ext-andriy.shevchenko(a)nokia.com> Signed-off-by: Andy Shevchenko <ext-andriy.shevchenko(a)nokia.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- sound/pci/riptide/riptide.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index ad44626..59d7996 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -97,6 +97,7 @@ #include <linux/gameport.h> #include <linux/device.h> #include <linux/firmware.h> +#include <linux/kernel.h> #include <asm/io.h> #include <sound/core.h> #include <sound/info.h> @@ -667,13 +668,12 @@ static u32 atoh(const unsigned char *in, unsigned int len) unsigned char c; while (len) { + int value; + c = in[len - 1]; - if ((c >= '0') && (c <= '9')) - sum += mult * (c - '0'); - else if ((c >= 'A') && (c <= 'F')) - sum += mult * (c - ('A' - 10)); - else if ((c >= 'a') && (c <= 'f')) - sum += mult * (c - ('a' - 10)); + value = hex_to_bin(c); + if (value >= 0) + sum += mult * value; mult *= 16; --len; } -- 1.6.6.1

2 1

[alsa-devel] [PATCH] Get rid of incorrect "long" types and casts.
by linux＠audioscience.com 17 Jun '10

17 Jun '10

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> These give incorrect results for index wrap on 64 bit. Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- sound/pci/asihpi/hpi6205.c | 22 +++++++++++----------- 1 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index e89991e..3b44134 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -941,11 +941,11 @@ static void outstream_host_buffer_free(struct hpi_adapter_obj *pao, } -static long outstream_get_space_available(struct hpi_hostbuffer_status +static u32 outstream_get_space_available(struct hpi_hostbuffer_status *status) { - return status->size_in_bytes - ((long)(status->host_index) - - (long)(status->dSP_index)); + return status->size_in_bytes - (status->host_index - + status->dSP_index); } static void outstream_write(struct hpi_adapter_obj *pao, @@ -954,7 +954,7 @@ static void outstream_write(struct hpi_adapter_obj *pao, struct hpi_hw_obj *phw = pao->priv; struct bus_master_interface *interface = phw->p_interface_buffer; struct hpi_hostbuffer_status *status; - long space_available; + u32 space_available; if (!phw->outstream_host_buffer_size[phm->obj_index]) { /* there is no BBM buffer, write via message */ @@ -1007,7 +1007,7 @@ static void outstream_write(struct hpi_adapter_obj *pao, } space_available = outstream_get_space_available(status); - if (space_available < (long)phm->u.d.u.data.data_size) { + if (space_available < phm->u.d.u.data.data_size) { phr->error = HPI_ERROR_INVALID_DATASIZE; return; } @@ -1018,7 +1018,7 @@ static void outstream_write(struct hpi_adapter_obj *pao, && hpios_locked_mem_valid(&phw->outstream_host_buffers[phm-> obj_index])) { u8 *p_bbm_data; - long l_first_write; + u32 l_first_write; u8 *p_app_data = (u8 *)phm->u.d.u.data.pb_data; if (hpios_locked_mem_get_virt_addr(&phw-> @@ -1248,9 +1248,9 @@ static void instream_start(struct hpi_adapter_obj *pao, hw_message(pao, phm, phr); } -static long instream_get_bytes_available(struct hpi_hostbuffer_status *status) +static u32 instream_get_bytes_available(struct hpi_hostbuffer_status *status) { - return (long)(status->dSP_index) - (long)(status->host_index); + return status->dSP_index - status->host_index; } static void instream_read(struct hpi_adapter_obj *pao, @@ -1259,9 +1259,9 @@ static void instream_read(struct hpi_adapter_obj *pao, struct hpi_hw_obj *phw = pao->priv; struct bus_master_interface *interface = phw->p_interface_buffer; struct hpi_hostbuffer_status *status; - long data_available; + u32 data_available; u8 *p_bbm_data; - long l_first_read; + u32 l_first_read; u8 *p_app_data = (u8 *)phm->u.d.u.data.pb_data; if (!phw->instream_host_buffer_size[phm->obj_index]) { @@ -1272,7 +1272,7 @@ static void instream_read(struct hpi_adapter_obj *pao, status = &interface->instream_host_buffer_status[phm->obj_index]; data_available = instream_get_bytes_available(status); - if (data_available < (long)phm->u.d.u.data.data_size) { + if (data_available < phm->u.d.u.data.data_size) { phr->error = HPI_ERROR_INVALID_DATASIZE; return; } -- 1.7.0.4

2 1

Re: [alsa-devel] [PATCH] ASoC: Add max9888 CODEC driver
by Mark Brown 17 Jun '10

17 Jun '10

On Wed, Jun 16, 2010 at 01:40:57PM -0500, jesse.marroquin(a)maxim-ic.com wrote: > From: Jesse Marroquin <jesse.marroquin(a)maxim-ic.com> > > This patch adds max9888 CODEC driver. So, the major comment I have here is that this driver feels very much like it's fighting with the standard APIs for things rather than working with them more than it should - there's an awful lot of open coded stuff that I can't see a real need for here (and if there is a need it should be factored out of the driver), and some things that are just straight reimplementations of standard kernel functionality. More detailed comments as I go through: > select SND_SOC_WM9712 if SND_SOC_AC97_BUS > select SND_SOC_WM9713 if SND_SOC_AC97_BUS > + select SND_SOC_MAX9888 if I2C > help Keep the entries in this file and the Makefile sorted. > +#define AUDIO_NAME "MAX9888" > +#define MAX9888_DRIVER_VERSION "1.0" In general version numbers in the kernel code are bad karma - they tend to just bitrot with the version number of the kernel being the only signifgcant bit. > +#ifdef MAX9888_DEBUG > +#define dprintk(x...) printk(KERN_WARNING x) > +#else > +#define dprintk(x...) > +#endif > + > +#define err(format, arg...) \ > + printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) > + > +#define info(format, arg...) \ > + printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) > + > +#define warn(format, arg...) \ > + printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) This stuff should all be using the standard dev_ macros (or pr_ ones where no device is available). Those give more informative output too. > +/* > + * Equalizer DSP filter coefficients generated from the evkit software tool > + */ So, what happens if a user goes and uses this tool to generate their own register settings? The way the software is structured it's going to be fairly involved for them to add additional configurations. This all looks like it ought to be implemented in a much more generic fashion with a data table being parsed by the driver to give the coefficient settings - at the minute pretty much everywhere the coefficient settings are handled the set of possible settings is manually coded. This would then mean that platform data could be supplied adding additional settings to or replacing the default ones provided by the driver. > +/* > + * Dynamic high pass excursion limiter filter coefficients. > + * The coefficients define the user programmable frequency response for the > + * excursion limiter. > + * Use the PC evkit software to generate the coefficients and put it here. > + */ > +static unsigned int param_ex_resp_a[] = { 0, 0, 0, 0, 0 }; > +static unsigned int param_ex_resp_b[] = { 0, 0, 0, 0, 0 }; > +static unsigned int param_ex_resp_c[] = { 0, 0, 0, 0, 0 }; This needs to be platform data, users should not have to modify the driver code in order to use the device since obviously that's not sustainable when the driver is shared by multiple boards. > +/* > + * Read the MAX9888 register space > + */ > +static unsigned int max9888_read(struct snd_soc_codec *codec, unsigned int reg) > +{ This looks like you can just use snd_soc_set_cache_io() and use the standard register I/O functions. > +/* > + * The INx1 and INx2 PGA's share a power control signal. Extra ' here. > + * This function OR's the two power events to keep an unpowered INx > + * from turning off it's counterpart. > + * The control names are used to identify the PGA. Some more comments about how you've actually implemented this might be a little clearer. One thing you might wish to consider which would simplify this workaround is to have the individual PGAs have no power management and use a SND_SOC_DAPM_SUPPLY for the actual register control. This will mean that you don't need to open code anything, DAPM will do the power up for you. It does mean the sequencing will be wrong but since these are input PGAs you'll probably not experience any negative side effects. Actually, looking further down the driver where this is used I'm confused about what you're doing here since all the users also have their own individual power bits. > + */ > +int max9888_pga_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, > + int event) > +{ > + struct snd_soc_codec *codec = w->codec; > + unsigned int val; > + unsigned int pga; > + unsigned int mask; > + static int power_state; This should be in the CODEC private data rather than a static - if someone wants to put two of the chips down on one board no need to make problems for them! > + if (w->reg != M9888_REG_4A_PWR_EN_IN) > + return -EINVAL; I'd upgrade this to a BUG_ON(), if the driver is calling this with the wrong register it's buggy. > + /* Turn on, avoiding unnecessary writes */ > + val = max9888_read(codec, w->reg); > + > + if (!(val & (1 << w->shift))) { > + val |= (1 << w->shift); > + max9888_write(codec, w->reg, val | MBEN); You're looking for snd_soc_update_bits() here. > +/* > + * Define a few static eq settings > + */ > +static const char *max9888_eq[] = { > + "voice", "music", "flat", "treble reduce", "loudness" }; The names should all be capitalised to work well in UIs. > +static int max9888_eq1_value = MAX9888_EQ_FLAT; > +static int max9888_eq2_value = MAX9888_EQ_FLAT; Again, these should be CODEC private data rather than static. > +static const char *max9888_ex_mode[] = { > + "disable", > + "100Hz", > + "400Hz", > + "600Hz", > + "800Hz", > + "1000Hz", > + "200-400Hz", > + "400-600Hz", > + "400-800Hz", > + "user-400Hz", > + "user-600Hz", > + "user-800Hz", > + "user-1000Hz" If the user has specified modes I'd rather expect they'd just override the standard settings. Conversely, if the user hasn't specified modes I'd not expect the driver to offer to use them. > +/* > + * Functions and control structures for loading > + * static coefficients into the bi-quad filter. > + * > + * As it is the user has a choice of three filters - a, b, or c. > + */ > +static const char *max9888_ex_resp[] = { "a", "b", "c" }; If there's no more meaningful name for the options it'd be nice to at least include a comment explaining why we're just calling them A, B and C. > +static int max9888_ex1_resp_value = MAX9888_BIQUAD_A; > +static int max9888_ex2_resp_value = MAX9888_BIQUAD_A; Same comment as before about use of private data. > + > +/* > + * Control mic1 analog amplifier circuit on/off, and gain > + * Typical setting: register value of 0x7F for 30dB mic gain > + */ > +void m9888_mic1_gain_control(struct snd_soc_codec *codec, unsigned int gain, > + unsigned int power_on) I really can't tell why this has been open coded - you probably just need an appropraite TLV table. If it needs to be open coded please document why. > + /* power off the amplifier */ > + if (power_on == 0) { Is the power really controlled by the gain? If so, why isn't this hooked in to DAPM? > +/* > + * Set headphone output volume and mute setting > + */ > +void headphone_set_volume(struct snd_soc_codec *codec, u8 vol, u8 mute) Similar comments about open coding for this and all the other set_volume() implementations - they all look even more standard than the microphone PGAs so you should be able to code them in a much more standard fashion. > +/* > + * configure speaker ALC > + */ > +void m9888_speaker_ALC_control(struct snd_soc_codec *codec) > +{ > + /* compressor enable */ > + max9888_write(codec, M9888_REG_41_SPKALC_COMP, (0 << 7) | > + (7 << 4) | (0 << 3) | (0 << 0)); This and several of the other functions look like they can only ever set one value - what's the purpose of these functions? > +static const char *max9888_hp_spk_mute[] = { "disable", "enable" }; > + > +static const struct soc_enum max9888_hp_spk_mute_enum[] = { > + SOC_ENUM_SINGLE_EXT(2, max9888_hp_spk_mute), > +}; This is not how ALSA implements mute controls - you should have a control with a name "Speaker Switch" (something Switch, anyway) almost certainly using one of the standard register access macros. See other CODEC drivers for examples. The same thing applies to all your other mute controls, all this mute handling can probably be replaced with a single line of code for each of the mutes. This will also have the advantage of allowing individual control of the channels. > +static const struct soc_enum max9888_enum[] = { > + SOC_ENUM_SINGLE(M9888_REG_3A_LVL_RECEIVER, 7, 2, max9888_rx_mute), > + SOC_ENUM_SINGLE(M9888_REG_18_DAI1_FILTERS, 7, 2, max9888_fltr_mode), > + SOC_ENUM_SINGLE(M9888_REG_18_DAI1_FILTERS, 3, 2, max9888_highrate), > + SOC_ENUM_SINGLE(M9888_REG_18_DAI1_FILTERS, 0, 6, max9888_dai1_fltr), > + SOC_ENUM_SINGLE(M9888_REG_18_DAI1_FILTERS, 4, 6, max9888_dai1_fltr), > + SOC_ENUM_SINGLE(M9888_REG_20_DAI2_FILTERS, 3, 2, max9888_highrate), > + SOC_ENUM_SINGLE(M9888_REG_20_DAI2_FILTERS, 0, 2, max9888_dc_block), > + SOC_ENUM_SINGLE(M9888_REG_40_SPKDHP_THRESHOLD, 0, 8, > + max9888_exlimit_threshold) > +}; Define individual static variables for each of these, indexing into a table is very error prone and brings no benefit. > + /* analog output levels */ > + SOC_DOUBLE_R("hp_vol", M9888_REG_38_LVL_HEADPHONE_L, > + M9888_REG_39_LVL_HEADPHONE_R, 0, 31, 0), This should be "Headphone Volume". Pretty much all your controls have naming problems - you should try to follow the guidlines in the kernel in Documentation/sound/alsa/ControlNames.txt. This will allow applications to parse them and render them appropriately. Since there's so many errors here I'll not point out further ones. > + SND_SOC_DAPM_PGA_E("INA2 Input", M9888_REG_4A_PWR_EN_IN, > + 7, 0, NULL, 0, max9888_pga_event, > + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), > + > + SND_SOC_DAPM_PGA_E("INB1 Input", M9888_REG_4A_PWR_EN_IN, > + 6, 0, NULL, 0, max9888_pga_event, > + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), Looking at this in conjunction with the event above things seem really odd - you've got this event which implements shared power bits for something, but all of these controls also have individually managed power bits that are handled in the normal fashion. > +/* > + * Set bias level > + */ > +static int max9888_set_bias_level(struct snd_soc_codec *codec, > + enum snd_soc_bias_level level) > +{ > + switch (level) { > + case SND_SOC_BIAS_ON: > + case SND_SOC_BIAS_PREPARE: > + case SND_SOC_BIAS_STANDBY: > + max9888_write(codec, M9888_REG_4C_PWR_SYS, SHDN|JDWK); You probably want to use snd_soc_update_bits() here (or just make _ON and _PREPARE noops). > + break; > + case SND_SOC_BIAS_OFF: > + max9888_write(codec, M9888_REG_4C_PWR_SYS, JDWK); It's a bit odd that you're enabling something in _OFF, especially something that's enabled when the device is active - what does JDWK do? > +int max9888_mute(struct snd_soc_dai *dai, int mute) > +{ > + struct snd_soc_codec *codec = dai->codec; > + > + if (mute) > + max9888_set_bias_level(codec, SND_SOC_BIAS_OFF); > + else > + max9888_set_bias_level(codec, SND_SOC_BIAS_ON); > + return 0; > +} This is wrong - the bias management function should be managing the global bias levels for the CODEC and the digital mute function should be muting the digital audio interface. One or both of these functions is doing the wrong thing. Note in particular that bypass paths need to work while the DAI is muted. > + case SND_SOC_DAIFMT_CBS_CFM: > + /* codec clk slave & frm master */ > + case SND_SOC_DAIFMT_CBM_CFS: > + /* codec clk master & frame slave */ > + info("Clock mode unsupported"); > + return -1; You should return a real error code here, not a random negative number, and the error message should be an error not an info statement. > + /* I2S or TDM */ > + if ((fmt & SND_SOC_DAIFMT_I2S) == SND_SOC_DAIFMT_I2S) > + max9888->reg_14_val &= ~(1 << 2); > + else > + max9888->reg_14_val |= (1 << 2); I don't think that comparison does what you wanted - look at what the value of SND_SOC_DAIFMT_I2S is for one thing. > + max9888_write(codec, M9888_REG_14_DAI1_FORMAT, > + max9888->reg_14_val); > + > + max9888_write(codec, M9888_REG_15_DAI1_CLOCK, > + (0 << 6) | (0 << 0)); > + > + max9888_write(codec, M9888_REG_16_DAI1_IOCFG, > + (1 << 6) | > + (0 << 5) | > + (0 << 4) | > + (0 << 3) | (0 << 2) | (1 << 1) | (1 << 0)); > + > + max9888_write(codec, M9888_REG_17_DAI1_TDM, > + (0 << 6) | (1 << 4) | (0 << 0)); These statements unconditionally write some fixed magic numbers to some registers. That's a bit odd - if nothing else I'd expect that any static configuration would be done once on system startup. > + * Setup DAI2 format > + */ > +static int max9888_dai2_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) Similar comments apply to the above; it looks awfully like these can be generalised. I also note that while you have two DAIs here there's no mention of them in your DAPM configuration - you have everything wired directly to the DAC and ADC. I'd not expect this to work terribly well if someone tries to use both DAIs (see below for some more on this). > + } else { > + info("Invalid master clock frequency %u", freq); > + return 1; dev_err() and return a proper error code. > + /* these registers writes take effect with SHDN cycle */ > + if (max9888_read(codec, M9888_REG_4C_PWR_SYS) & 0x80) { > + max9888_write(codec, M9888_REG_4C_PWR_SYS, 0x00); > + max9888_write(codec, M9888_REG_4C_PWR_SYS, 0x80); ...and they do...? > +/* DAI Voice mode 2 */ > + {.name = "max9888 DAI1", > + .id = 2, > + .playback = { > + .stream_name = "Voice Playback", > + .channels_min = 1, > + .channels_max = 1, > + .rates = MAX9888_RATES, > + .formats = MAX9888_FORMATS,}, So, Voice Playback isn't actually wired up to anything. This means that when you try to start playback from a voice stream nothing will actually happen to power up any of the output paths, which probably isn't what's wanted. > + .capture = { > + .stream_name = "Capture", > + .channels_min = 1, > + .channels_max = 1, > + .rates = MAX9888_RATES, > + .formats = MAX9888_FORMATS,}, This, on the other hand, uses the same name as the primary DAI so I'd expect some confusion in DAPM if both are simultaneously active. > + /* Jack detection function only works well on Revision C */ > + if (max9888->rev_version == MAX9888_REVC) { > + max9888_write(codec, M9888_REG_0F_IRQ_ENABLE, IJDET); > + max9888_write(codec, M9888_REG_49_CFG_JACKDET, > + JDETEN | JDEB_200ms); > + max9888_write(codec, M9888_REG_4C_PWR_SYS, JDWK); > + } I see no support for jack detection in the driver... > + max9888_write(codec, M9888_REG_22_MIX_ADC_LEFT, MIX_MIC1 | MIX_INA1); > + max9888_write(codec, M9888_REG_23_MIX_ADC_RIGHT, MIX_MIC2 | MIX_INA1); > + max9888_write(codec, M9888_REG_21_MIX_DAC, > + MIX_DAI1L_TO_DACL | MIX_DAI2L_TO_DACL | MIX_DAI1R_TO_DACR > + | MIX_DAI2R_TO_DACR); > + max9888_write(codec, M9888_REG_29_MIX_SPEAKER, > + MIX_LDAC_TO_SPL | MIX_RDAC_TO_SPR); > + max9888_write(codec, M9888_REG_27_MIX_HEADPHONE, > + MIX_LDAC_TO_HPL | MIX_RDAC_TO_HPR); > + > + headphone_set_volume(codec, 0x1f, 0); > + speaker_set_volume(codec, 0x1f, 0); > + m9888_mic1_gain_control(codec, 20, 1); > + m9888_mic2_gain_control(codec, 20, 1); These should all use the chip defaults and allow the user to configure things appropriately for their own system via the ALSA controls. The driver has to work with all boards using the device which means that it's not possible to pick sane default setups. > +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) Is I2C control really optional? > +/* > + * gumstix i2c codec control layer > + */ Cut'n'paste. > +#define IJDET (1<<1) This needs to be namespaced. > +#define M9888_REG_21_MIX_DAC 0x21 > +#define MIX_DAI1L_TO_DACL (1<<7) > +#define MIX_DAI1R_TO_DACL (1<<6) > +#define MIX_DAI2L_TO_DACL (1<<5) > +#define MIX_DAI2R_TO_DACL (1<<4) > +#define MIX_DAI1L_TO_DACR (1<<3) > +#define MIX_DAI1R_TO_DACR (1<<2) > +#define MIX_DAI2L_TO_DACR (1<<1) > +#define MIX_DAI2R_TO_DACR (1<<0) This too; all the individual register bit definitions in this header have this problem. > +#define HIGH_BYTE(w) ((w >> 8) & 0x00ff) > +#define LOW_BYTE(w) (w & 0x00ff) I suspect there are standard defines for these. If not there possibly ought to be (and you have a namespace issue if defining locally).

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[alsa-devel] ASoC updates for 2.6.36
by Mark Brown 16 Jun '10

16 Jun '10

The following changes since commit 67884e215bf24ad9c244cc8b9eb3c243e85385a1: Merge branch 'for-2.6.35' into for-2.6.36 (2010-06-15 11:55:35 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Tweak to the WM2000 driver and a fix for the error reporting issues with volsw_2r_sx. Mark Brown (2): ASoC: Default WM2000 ANC and speaker to enabled ASoC: Pay attention to write errors in volsw_2r_sx sound/soc/codecs/wm2000.c | 2 ++ sound/soc/soc-core.c | 6 ++---- 2 files changed, 4 insertions(+), 4 deletions(-)

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Re: [alsa-devel] [PATCH v2] soc/sh: Fix I2C dependency for SND_FSI_AK4642 and SND_FSI_DA7210
by Mark Brown 16 Jun '10

16 Jun '10

On Tue, Jun 15, 2010 at 05:38:55PM +0200, Peter Huewe wrote: > From: Peter Huewe <peterhuewe(a)gmx.de> > > The config option SND_FSI_AK4642 selects SND_SOC_AK4642 which in turn > enables the compilation of ak4642.c - however this codec uses I2C to > communicate with the HW. > Same applies to DA7210. Applied, thanks.

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[alsa-devel] [PATCH RFC 0/3] WM97xx AC97 GPIO support
by Manuel Lauss 16 Jun '10

16 Jun '10

Here's 3 patches which implement simple AC97 GPIO support to the WM97xx touch drivers and the Alchemy (au1x) AC97 ASoC implementation. Tested on Au1200 hardware with a WM9712 codec: According to Mark Brown the WM9712 only sets GPIOs when data is sent through slot 12, which prompted me to implement this. Thanks, Manuel Lauss Manuel Lauss (3): sound: ac97: Add GPIO manipulation hooks input: wm97xx: set GPIOs with AC97 GPIO busops if implemented sound: asoc: au1x: implement AC97 GPIO access drivers/input/touchscreen/wm97xx-core.c | 5 +++- include/sound/ac97_codec.h | 2 + sound/soc/au1x/psc-ac97.c | 39 +++++++++++++++++++++++++++++- sound/soc/au1x/psc.h | 2 + 4 files changed, 45 insertions(+), 3 deletions(-) -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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Re: [alsa-devel] [PATCH v2] soc/sh: Fix I2C dependency for SND_FSI_AK4642 and SND_FSI_DA7210
by Paul Mundt 16 Jun '10

16 Jun '10

On Tue, Jun 15, 2010 at 05:38:55PM +0200, Peter Huewe wrote: > The config option SND_FSI_AK4642 selects SND_SOC_AK4642 which in turn > enables the compilation of ak4642.c - however this codec uses I2C to > communicate with the HW. > Same applies to DA7210. > It uses I2C, but it doesn't matter which I2C master it's interfaced with. A simple depends on I2C will suffice for this purpose just fine.

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[alsa-devel] snd_pcm_wait won't return
by Dennis Borgmann 16 Jun '10

16 Jun '10

Dear alsa-devel list! I already posted this on alsa-users, but this list doesn't seem to be that crowded so I will give it a try in here... I am using the alsa-lib in one of my projects and within this project, there is a call for snd_pcm_wait() on a machine, that plays sound. The call itself looks like this: snd_pcm_wait( pcm_handle, 1000 ); A few steps before, I had the timeout set to unlimited (-1), which sometimes did not return at all. Now, it sometimes returns after the timeout, but in result I cannot write anything to the soundcard anymore. This error is reproducable if there are many underruns in a row. Since this program runs in a network environment and receives its data from a datastream, I sometimes cannot avoid those underruns. So, what could be the error, that leads to alsa not responding anymore? I can only revitalize the system after such an error by rebooting the whole system (well, at least I did not find another solution like snd_pcm_prepare or snd_pcm_recover). Which additional informations do you need? What is there, that I can provide in addition? Could this be an alsa-bug? Kind regards, Dennis

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Re: [alsa-devel] multiple definitions of 'to_psc_dma_stream' in kernel-2.6.34
by Grant Likely 15 Jun '10

15 Jun '10

On Sat, Jun 12, 2010 at 2:56 PM, Jon Smirl <jonsmirl(a)gmail.com> wrote: > Grant, is this something you did? Please try the attached fix. g. > > commit 1d8222e8df07ce4f86fb7fa80b02bdee03b57985 > Author: Grant Likely <grant.likely(a)secretlab.ca> > Date: Sat Nov 7 01:34:31 2009 -0700 > > ASoC/mpc5200: add to_psc_dma_stream() helper > > Move the resolving of the psc_dma_stream pointer to a helper function > to reduce duplicate code > > Signed-off-by: Grant Likely <grant.likely(a)secretlab.ca> > Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> > Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > > > On Sat, Jun 12, 2010 at 4:30 PM, John Hilmar Linkhorst > <John.Linkhorst(a)rwth-aachen.de> wrote: >> Dear Jon, >> >> while compiling the kernel for my efika machine I ran into the following error: >> >> LD sound/soc/fsl/built-in.o >> sound/soc/fsl/mpc5200_psc_ac97.o: In function `to_psc_dma_stream': >> mpc5200_psc_ac97.c:(.text+0x0): multiple definition of `to_psc_dma_stream' >> sound/soc/fsl/mpc5200_dma.o:mpc5200_dma.c:(.text+0x0): first defined here >> sound/soc/fsl/efika-audio-fabric.o: In function `to_psc_dma_stream': >> efika-audio-fabric.c:(.text+0x0): multiple definition of `to_psc_dma_stream' >> sound/soc/fsl/mpc5200_dma.o:mpc5200_dma.c:(.text+0x0): first defined here >> make[3]: *** [sound/soc/fsl/built-in.o] Error 1 >> make[2]: *** [sound/soc/fsl] Error 2 >> make[1]: *** [sound/soc] Error 2 >> make: *** [sound] Error 2 >> >> could you help me solving this problem? >> >> best regards >> John Linkhorst >> > > > > -- > Jon Smirl > jonsmirl(a)gmail.com > -- Grant Likely, B.Sc., P.Eng. Secret Lab Technologies Ltd.

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Re: [alsa-devel] [PATCH] soc/sh: Fix I2C dependency for SND_FSI_AK4642
by Mark Brown 15 Jun '10

15 Jun '10

On Tue, Jun 15, 2010 at 04:49:07PM +0200, Peter Hüwe wrote: > So maybe even move the dependency up to SND_SOC_SH4_FSI > since both codecs rely on I2C? No, while the two boards in mainline currently may need I2C obviously other board designs with other control interfaces are possible so it should be on the individual boards.

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[alsa-devel] ASoC updates for 2.6.36
by Mark Brown 15 Jun '10

15 Jun '10

The following changes since commit 9eb34302681d3f6cf0b186aae31ba08cbd5f22fb: Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2010-06-07 18:38:56 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Mostly minor cleanups plus the fix from 2.6.35. Grant Likely (1): ASoC: Remove unused header from MPC5200 PSC driver Kuninori Morimoto (5): ASoC: header cleanup for FSI ASoC: header cleanup for FSI-AK4642 ASoC: header cleanup for FSI-DA7210 ASoC: header cleanup for ak4642 ASoC: header cleanup for da7210 Mark Brown (2): Merge branch 'ep93xx' into for-2.6.36 Merge branch 'for-2.6.35' into for-2.6.36 Ryan Mallon (2): EP93xx: Add i2s core support ASoC: EP93xx: Add Snapper CL15 i2s audio support Sudhakar Rajashekhara (1): ASoC: DaVinci: Fix McASP hardware FIFO configuration Takashi Iwai (1): Merge branch 'for-2.6.35' of git://git.kernel.org/.../broonie/sound-2.6 into fix/asoc Wan ZongShun (1): ASoC: NUC900: patch for fix build error arch/arm/mach-ep93xx/clock.c | 67 ++++++++++- arch/arm/mach-ep93xx/core.c | 67 ++++++++++ arch/arm/mach-ep93xx/include/mach/ep93xx-regs.h | 10 ++ arch/arm/mach-ep93xx/include/mach/platform.h | 3 + arch/arm/mach-ep93xx/snappercl15.c | 1 + sound/soc/codecs/ak4642.c | 8 -- sound/soc/codecs/da7210.c | 9 -- sound/soc/davinci/davinci-mcasp.c | 2 - sound/soc/ep93xx/Kconfig | 9 ++ sound/soc/ep93xx/Makefile | 3 + sound/soc/ep93xx/snappercl15.c | 150 +++++++++++++++++++++++ sound/soc/fsl/mpc5200_psc_i2s.c | 1 - sound/soc/fsl/mpc5200_psc_i2s.h | 12 -- sound/soc/nuc900/nuc900-audio.c | 2 +- sound/soc/nuc900/nuc900-pcm.c | 2 +- sound/soc/sh/fsi-ak4642.c | 9 -- sound/soc/sh/fsi-da7210.c | 9 -- sound/soc/sh/fsi.c | 9 -- 18 files changed, 311 insertions(+), 62 deletions(-) create mode 100644 sound/soc/ep93xx/snappercl15.c delete mode 100644 sound/soc/fsl/mpc5200_psc_i2s.h

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[alsa-devel] ASoC update for 2.6.35
by Mark Brown 15 Jun '10

15 Jun '10

The following changes since commit e854df613fe934c94a0b39eccb4104e72ccbbded: Merge branch 'for-2.6.35' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc (2010-06-02 14:17:44 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Sudhakar Rajashekhara (1): ASoC: DaVinci: Fix McASP hardware FIFO configuration There is a fix for duplicate enables of the DMA, the changelog is detailed. sound/soc/davinci/davinci-mcasp.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-)

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[alsa-devel] [PATCH 0/6] ASoC: header cleanup
by Kuninori Morimoto 15 Jun '10

15 Jun '10

Dear Mark These are header cleanup patches Kuninori Morimoto (5): ASoC: header cleanup for FSI ASoC: header cleanup for FSI-AK4642 ASoC: header cleanup for FSI-DA7210 ASoC: header cleanup for ak4642 ASoC: header cleanup for da7210 Not so important Best regards -- Kuninori Morimoto

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Re: [alsa-devel] [PATCH] ASoC: DaVinci: Fix McASP hardware FIFO configuration
by Mark Brown 15 Jun '10

15 Jun '10

On Fri, Jun 11, 2010 at 07:24:51PM +0530, Sudhakar Rajashekhara wrote: > On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral > has FIFO support. This FIFO provides additional data buffering. It > also provides tolerance to variation in host/DMA controller response > times. More details of the FIFO operation can be found at Applied, thanks.

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Re: [alsa-devel] [PATCH] soc/sh: Fix I2C dependency for SND_FSI_AK4642
by Mark Brown 14 Jun '10

14 Jun '10

On Mon, Jun 14, 2010 at 05:39:41PM +0200, Peter Huewe wrote: > bool "FSI-AK4642 sound support" > depends on SND_SOC_SH4_FSI > select SND_SOC_AK4642 > + select I2C I'd have thought a depends was more appropriate. Arguably you want to depend on the SH I2C driver since the driver is useless without the relevant I2C controller.

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[alsa-devel] [PATCH] ASoC: DaVinci: Fix McASP hardware FIFO configuration
by Sudhakar Rajashekhara 14 Jun '10

14 Jun '10

On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral has FIFO support. This FIFO provides additional data buffering. It also provides tolerance to variation in host/DMA controller response times. More details of the FIFO operation can be found at http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber=spr… Existing sequence of steps for audio playback/capture are: a. DMA configuration b. McASP configuration (configures and enables FIFO) c. Start DMA d. Start McASP (enables FIFO) During McASP configuration, while FIFO was being configured, FIFO was being enabled in davinci_hw_common_param() function of sound/soc/davinci/davinci-mcasp.c file. This generated a transmit DMA event, which gets serviced when DMA is started. https://patchwork.kernel.org/patch/84611/ patch clears the DMA events before starting DMA, which is the right thing to do. But this resulted in a state where DMA was waiting for an event from McASP (after step c above), but the event which was already there, has got cleared (because of step b above). The fix is not to enable the FIFO during McASP configuration as FIFO was being enabled as part of McASP start. Signed-off-by: Sudhakar Rajashekhara <sudhakar.raj(a)ti.com> --- Applies to latest of Linus's tree. sound/soc/davinci/davinci-mcasp.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 79f0f4a..d395509 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -612,7 +612,6 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) NUMDMA_MASK); mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); - mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); } if (dev->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) { @@ -623,7 +622,6 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) NUMDMA_MASK); mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); - mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); } } -- 1.5.6

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Re: [alsa-devel] [PATCH 3/4] ALSA: usb-audio: parse UAC2 sample rate ranges correctly
by Daniel Mack 14 Jun '10

14 Jun '10

On Fri, Jun 11, 2010 at 10:59:44PM +0800, Alex Lee wrote: > If the device has a variable clock, it may report a RES of only 1. So you > may end up with thousands (and even millions) of sample rates. It might be > better to have separate code to cater to discrete clocks vs variable clocks. Just for the records: this is exactly what Mac OS X does, and it will comsume 100% CPU when a device with such descriptors is plugged in. So even though this is now sanely handled by ALSA, it really shouldn't be done that way by implementors. Daniel

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Re: [alsa-devel] [PATCH 1/1] ALSA: hda - Fix line-in for mb5 model MacBook (Pro) 5, 1 / 5, 2
by Takashi Iwai 14 Jun '10

14 Jun '10

At Mon, 14 Jun 2010 16:31:21 +0930, Alex Murray wrote: > > Takashi - There is no incompatibilty, both are the same - the mb5 model > is used for all MacBook (Pro) 5,1 / 5,2 models, the only differnce > between the Pro and non-Pro is the Pro has the subwoofer (LFE) which the > non-Pro lacks. > > You can see here > http://ubuntuforums.org/showpost.php?p=9444153&postcount=3 for a copy > of /proc/asound/card0/codec#0 on a MacBook 5,1 (i.e. not Pro) which > confirms the line-out is 0x7 for that model as well - I've tested this > directly on my MacBook Pro 5,1 - so I am 100% confident this is correct > for both Pro and non-Pro models, unless you have evidence otherwise? OK, then it's all green. I applied the patch now. Thanks for a quick response! Takashi

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Re: [alsa-devel] [PATCH 1/1] ALSA: hda - Fix line-in for mb5 model MacBook (Pro) 5, 1 / 5, 2
by Takashi Iwai 14 Jun '10

14 Jun '10

At Mon, 14 Jun 2010 12:08:43 +0930, murray.alex(a)gmail.com wrote: > > From: Alex Murray <murray.alex(a)gmail.com> > > The line-in input is 0x7 not 0x2 for MacBook (Pro) 5,1 / 5,2 models > > Signed-off-by: Alex Murray <murray.alex(a)gmail.com> MB5 != MBP5, so I suspect this might be rather the incompatibility between them... Takashi > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index 17d4548..007f96b 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -6948,7 +6948,7 @@ static struct hda_input_mux mb5_capture_source = { > .num_items = 3, > .items = { > { "Mic", 0x1 }, > - { "Line", 0x2 }, > + { "Line", 0x7 }, > { "CD", 0x4 }, > }, > }; > @@ -7469,8 +7469,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = { > HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), > HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT), > HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT), > - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), > - HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), > + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT), > + HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT), > HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), > HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), > HDA_CODEC_VOLUME("Line Boost", 0x15, 0x00, HDA_INPUT), > @@ -7853,10 +7853,9 @@ static struct hda_verb alc885_mb5_init_verbs[] = { > {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, > {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, > > - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, > - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, > - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, > - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, > + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0x1)}, > + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x7)}, > + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x4)}, > { } > }; > > -- > 1.7.0.4 >

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Re: [alsa-devel] ALSA's jack reporting framework map to Android framework
by Mark Brown 14 Jun '10

14 Jun '10

On 12 Jun 2010, at 23:36, Mike Lockwood wrote: > There is no requirement to use the switch driver for the headset > detect. At the time we did this we couldn't find an existing standard > and we had already written the switch driver for something else, so it > was an easy solution at the time. But if there is something better > available now I don't see any reason not to use it. In situations like this I'd always recommend discussing upstream - there may be something you've missed, or if there isn't then it means whatever you come up with can be adopted as a standard by everyone. > I don't know what the current thinking is about ALSA and android - > someone on our media team would need to address that part of the > thread. But if using ALSA requires GPL/LGPL code in userspace that > would probably be a deal breaker right there. Obviously there's already the option of using ALSA with the regular libraries (which are LGPLed) in Android which a bunch of vendors are using already - there's existing drivers for a range of CPUs and CODECs and many vendors are already shipping other Linux based OSs on their devices outside of Android which use things like the standard audio stack and so don't have a problem with LGPL in userspace. Discussions with Brian Swetland (CCed in so he can correct me if I'm off base here) suggested that a direct to kernel option bypassing the standard libraries might be developed under a MITish license to provide as a default. This is obviously much less work than developing an entirely new audio stack would be, and unless plugins are in use the userspace portion is very thin.

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Re: [alsa-devel] Contents of Alsa-devel digest..
by Junhwa Rhee 14 Jun '10

14 Jun '10

Re: Contents of Alsa-devel digest..

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[alsa-devel] BUG: cx88 can't find device struct
by Timothy E. Krantz 13 Jun '10

13 Jun '10

Hi, I started looking into this problem a while back and gave up. Now I decided to try again. I am running MythDora 12.23 and would like to update the alsa driver to 1.0.23 so that I can use HDMI audio out on my nvidia GT220 card. So I download the alsa-driver-1.0.23.tar.bz file, unpack it, ./configure and make and make install. When I reboot I can verify the new alsa driver by cat/proc/asound/version and it is indeed 1.0.23. If I then try to record an analog station I get: Jun 9 14:40:19 mythdora kernel: BUG: cx88 can't find device struct. Can't proceed with open Jun 9 14:40:19 mythdora kernel: cx88[1]: Calling XC5000 callback Jun 9 14:40:19 mythdora kernel: BUG: cx88 can't find device struct. Can't proceed with open Hmmm, ok the alsa driver implementation updated some kernel header file perhaps, so I download the latest v4l-dvb tree and compile it and install it and reboot. Same thing. No sound on analog recordings. I am sure I am missing some important step or something. Any help would be greatly appreciated! Tim

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Re: [alsa-devel] Alsa-devel Digest, Vol 40, Issue 58
by Tobias Schneider 13 Jun '10

13 Jun '10

----- Ursprüngliche Nachricht ----- Von: alsa-devel-request(a)alsa-project.org Gesendet: Sonntag, 13. Juni 2010 12:00 An: alsa-devel(a)alsa-project.org Betreff: Alsa-devel Digest, Vol 40, Issue 58 Send Alsa-devel mailing list submissions to alsa-devel(a)alsa-project.org To subscribe or unsubscribe via the World Wide Web, visit http://mailman.alsa-project.org/mailman/listinfo/alsa-devel or, via email, send a message with subject or body 'help' to alsa-devel-request(a)alsa-project.org You can reach the person managing the list at alsa-devel-owner(a)alsa-project.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Alsa-devel digest..." Today's Topics: 1. Re: ALSA's jack reporting framework map to Android framework (Mark Brown) 2. Re: [PATCH] ASoC: Remove unused header (Mark Brown) 3. Re: Timing Info (Paul Dugas) ---------------------------------------------------------------------- Message: 1 Date: Sat, 12 Jun 2010 18:05:52 +0100 From: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Subject: Re: [alsa-devel] ALSA's jack reporting framework map to Android framework To: "Harsha, Priya" <priya.harsha(a)intel.com> Cc: "alsa-devel(a)alsa-project.org" <alsa-devel(a)alsa-project.org>, Mike Lockwood <lockwood(a)android.com> Message-ID: <20100612170552.GA3110(a)sirena.org.uk> Content-Type: text/plain; charset=us-ascii On Sat, Jun 12, 2010 at 09:28:34AM +0530, Harsha, Priya wrote: Please fix your MUA to word wrap your messages, it makes them more legible and easier to reply to. > The ALSA's jack reporting framework reports jacks at > /dev/input/eventx. But the Android framework expects the jack reporting > to be done in /sys/.../switch/h2w. That ... is simply 'class' - the Android kernel includes a new sysfs class for switches, CCing in Mike Lockwood who wrote it in case he's got any input from the Android side. > Any suggestions as to how the ALSA's jack reporting framework to > Android framework? Are there any audio drivers that have mapped this? The obvious thing to do here seems to be to update the Android framework to be able to use the standard kernel interfaces. I obviously can't speak for Google but it seems like the sort of patch they'd be willing to accept, especially if you retain the ability to use the non-standard extensions that are currently being used. I guess an alternative would be to have the Android kernels add an adaptor in the input subsystem (for various reasons not all jack sense goes through ALSA) so that input subsystem switches get represented via the Android switch ABI, though that would increase the amount of out of tree code that needs to be carried and so seems like the wrong direction to be heading. ------------------------------ Message: 2 Date: Sat, 12 Jun 2010 18:07:15 +0100 From: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Subject: Re: [alsa-devel] [PATCH] ASoC: Remove unused header To: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: Grant Likely <grant.likely(a)secretlab.ca>, Herb Peyerl <hpeyerl(a)beer.org>, alsa-devel(a)alsa-project.org, linux-kernel(a)vger.kernel.org Message-ID: <20100612170715.GB3110(a)sirena.org.uk> Content-Type: text/plain; charset=us-ascii On Fri, Jun 11, 2010 at 10:50:40AM +0100, Liam Girdwood wrote: > On Thu, 2010-06-10 at 17:54 -0600, Grant Likely wrote: > > The header contains an extern that isn't used by anything. Remove. > > Signed-off-by: Grant Likely <grant.likely(a)secretlab.ca> > Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> Applied, thanks. ------------------------------ Message: 3 Date: Sat, 12 Jun 2010 17:59:27 -0400 From: Paul Dugas <paul(a)dugasenterprises.com> Subject: Re: [alsa-devel] Timing Info To: alsa-devel(a)alsa-project.org Message-ID: <AANLkTik3ycq3POxHRG_bc0jO-KByZVvirVJ-X9-bySY_(a)mail.gmail.com> Content-Type: text/plain; charset=UTF-8 On Fri, Jun 11, 2010 at 6:00 PM, Paul Dugas <paul(a)dugasenterprises.com> wrote: > I'm using snd_pcm_readi() to collect samples and am wondering if/how I > can get accurate timestamps for them. ?Seems like the snd_pcm_status() > gives me this info but I'm unclear on the specifics. ?Can someone > point me toward the correct field in the status structure that I > should be using and what sample that timestamp corresponds to? > > Does snd_pcm_status_get_tstamp() get me the timestamp for the first > frame returned by the next snd_pcm_readi() call? I know now get_tstamp() isn't what I'm looking for after reading some prior postings and other web sites. Can't seem to find a discussion of timing in the ALSA docs themselves. Maybe I'm being dense. Looks like some processing of the trigger timestamp is what I need but I'm not following the logic of "triggers" in the simple use of snd_pcm_readi() that I've got. Before the first readi() call, the trigger time is 0. The second time, it's pretty close to the wall time for the first call. Perhaps that's the timestamp of the first sample of the first period I read and I need to add the period_time each time? Seems like that'd accrue error when recording over longer periods of time (which my application does). What's the "timestamp mode". Does it generate a timestamp each period? How would I access that timestamp? Thanks in advance for any assistance, Paul ------------------------------ _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel End of Alsa-devel Digest, Vol 40, Issue 58 ******************************************

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[alsa-devel] [PATCH] ASoC: Remove unused header
by Grant Likely 12 Jun '10

12 Jun '10

The header contains an extern that isn't used by anything. Remove. Signed-off-by: Grant Likely <grant.likely(a)secretlab.ca> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Jon Smirl <jonsmirl(a)gmail.com> Cc: Herb Peyerl <hpeyerl(a)beer.org> --- sound/soc/fsl/mpc5200_psc_i2s.c | 1 - sound/soc/fsl/mpc5200_psc_i2s.h | 12 ------------ 2 files changed, 0 insertions(+), 13 deletions(-) delete mode 100644 sound/soc/fsl/mpc5200_psc_i2s.h diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 4f455bd..676841c 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -16,7 +16,6 @@ #include <asm/mpc52xx_psc.h> -#include "mpc5200_psc_i2s.h" #include "mpc5200_dma.h" /** diff --git a/sound/soc/fsl/mpc5200_psc_i2s.h b/sound/soc/fsl/mpc5200_psc_i2s.h deleted file mode 100644 index ce55e07..0000000 --- a/sound/soc/fsl/mpc5200_psc_i2s.h +++ /dev/null @@ -1,12 +0,0 @@ -/* - * Freescale MPC5200 PSC in I2S mode - * ALSA SoC Digital Audio Interface (DAI) driver - * - */ - -#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__ -#define __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__ - -extern struct snd_soc_dai psc_i2s_dai[]; - -#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_I2S_H__ */

3 2

[alsa-devel] ALSA's jack reporting framework map to Android framework
by Harsha, Priya 12 Jun '10

12 Jun '10

Hi all, The ALSA's jack reporting framework reports jacks at /dev/input/eventx. But the Android framework expects the jack reporting to be done in /sys/.../switch/h2w. Any suggestions as to how the ALSA's jack reporting framework to Android framework? Are there any audio drivers that have mapped this? Any kind of input would be really helpful Thanks, Harsha

2 1

[alsa-devel] ac97 Master and PCM volume control
by Raymond Yau 12 Jun '10

12 Jun '10

https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/581650 >> Master ranges from -46.5dB to 0dB but PCM ranges from -34.5dB to +12dB, Daniel T Chen <https://launchpad.net/%7Ecrimsun> wrote on 2010-05-18: #1<https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/581650/comments/1> This is not a pulse bug but a hardware one I wonder why you said this is a hardware bug since this is ac97 specification maximum the total gain will be +12dB, not 0dB as indicated. Did you read those ac97 2.x specification ?

1 0

[alsa-devel] extra interrupt on plug insertion undoes mux selection
by Peter M. Petrakis 12 Jun '10

12 Jun '10

Hi All, Trying my hand at snd quirking, I'm working a Dell Latitude E4310. Here's what I'm seeing. There's a headset jack (HP/MIC combo) on the LHS, when I insert a headset I see the following in the logs. Note: port 0xA is the mic jack, 0xB is hp jack, and 0x11 in internal mic. Jun 11 18:17:45 ubuntu-laptop kernel: [ 193.978394] XXX stac92xx_unsol_event: codec subsystem id 10280410 Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361330] XXX stac92xx_unsol_event: event - 6 nid - 0a. Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361335] XXX stac92xx_unsol_event: event - 6 MIC Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361337] XXX stac92xx_mic_detect: ENTER Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361340] XXX get_pin_presence:4422 ENTER, NID 0a Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361434] XXX get_pin_presence:4429 EXIT 1, got pin sense Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361439] XXX stac92xx_mic_detect: HERE 4662 Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361441] XXX stac92xx_mic_detect: HERE 4676 Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361444] XXX stac92xx_mic_detect:4678 mux_nid:0017 index:6 Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361449] XXX stac92xx_mic_detect: EXIT Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361451] XXX stac92xx_unsol_event: genevent - 6 Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361532] XXX get_pin_presence:4422 ENTER, NID 0a Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361620] XXX get_pin_presence:4429 EXIT 1, got pin sense Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361628] XXX stac92xx_unsol_event: codec subsystem id 10280410 Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361631] XXX stac92xx_unsol_event: event - 4 nid - 0b. Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361634] XXX stac92xx_unsol_event: event - 4 HP or LO Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361636] XXX stac92xx_hp_detect: ENTER Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361638] XXX get_pin_presence:4422 ENTER, NID 0b Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361717] XXX get_pin_presence:4429 EXIT 1, got pin sense Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361720] XXX stac92xx_hp_detect:4522 probe_pin:0 hp_pin:000b Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361807] XXX stac92xx_hp_detect: pinctl c0 Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361810] XXX stac92xx_hp_detect: HERE 4534 Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361812] XXX stac92xx_hp_detect: HERE 4537 Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361906] XXX stac92xx_hp_detect: HERE 4569 Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361996] XXX stac92xx_hp_detect: EXIT Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.361998] XXX stac92xx_unsol_event: genevent - 4 Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.362067] XXX get_pin_presence:4422 ENTER, NID 0b Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.362163] XXX get_pin_presence:4429 EXIT 1, got pin sense Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.362168] XXX stac92xx_unsol_event: codec subsystem id 10280410 Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.735203] XXX stac92xx_unsol_event: event - 6 nid - 0a. Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.735206] XXX stac92xx_unsol_event: event - 6 MIC Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.735209] XXX stac92xx_mic_detect: ENTER Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.735212] XXX get_pin_presence:4422 ENTER, NID 0a Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.735311] XXX get_pin_presence:4433 EXIT 0, no pin sense Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.735315] XXX stac92xx_mic_detect: HERE 4666 Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.735318] XXX stac92xx_mic_detect: HERE 4676 Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.735320] XXX stac92xx_mic_detect:4678 mux_nid:0017 index:4 Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.735325] XXX stac92xx_mic_detect: EXIT Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.735327] XXX stac92xx_unsol_event: genevent - 6 Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.735410] XXX get_pin_presence:4422 ENTER, NID 0a Jun 11 18:17:45 ubuntu-laptop kernel: [ 201.735497] XXX get_pin_presence:4433 EXIT 0, no pin sense I've tested against kernels 2.6.32 and 2.6.35rc2 with the same results. The mux is indeed back where it started after the plug is fully inserted. Node 0x17 [Audio Selector] wcaps 0x300d0d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x0f 0x0f] Power: setting=D0, actual=D0 Connection: 7 0x0c 0x0e 0x0f 0x1b 0x11* 0x12 0x0a and I can push it around with hda-verb to get the desired result... The question is why is it doing this? To make things weirder, I can get input from the external mic even if it isn't selected. This is my first attempt at making codecs "do the right thing" so I could very well be missing something obvious. What else am I missing here? Thanks. Peter

1 0

Re: [alsa-devel] [PATCH 3/4] ALSA: usb-audio: parse UAC2 sample rate ranges correctly
by Daniel Mack 11 Jun '10

11 Jun '10

On Fri, Jun 11, 2010 at 10:59:44PM +0800, Alex Lee wrote: > If the device has a variable clock, it may report a RES of only 1. So you > may end up with thousands (and even millions) of sample rates. It might be > better to have separate code to cater to discrete clocks vs variable clocks. AFAIK, ALSA cannot cope with that yet. Am I right? Daniel > On Fri, Jun 11, 2010 at 9:33 PM, Daniel Mack <daniel(a)caiaq.de> wrote: > > > A device may report its supported sample rates in ranges rather than in > > discrete triplets. The code used to only parse the MIN field instead of > > properly paying attention to the MAX and RES values. > > > > Signed-off-by: Daniel Mack <daniel(a)caiaq.de> > > Reported-by: Alex Lee <alexlee188(a)gmail.com> > > --- > > sound/usb/format.c | 80 > > ++++++++++++++++++++++++++++++++++++++++----------- > > 1 files changed, 62 insertions(+), 18 deletions(-) > > > > diff --git a/sound/usb/format.c b/sound/usb/format.c > > index 8eccf17..e946db7 100644 > > --- a/sound/usb/format.c > > +++ b/sound/usb/format.c > > @@ -206,6 +206,47 @@ static int parse_audio_format_rates_v1(struct > > snd_usb_audio *chip, struct audiof > > } > > > > /* > > + * Helper function to walk the array of sample rate triplets reported by > > + * the device. The problem is that we need to parse whole array first to > > + * get to know how many sample rates we have to expect. > > + * Then fp->rate_table can be allocated and filled. > > + */ > > +static int parse_uac2_sample_rate_range(struct audioformat *fp, int > > nr_triplets, > > + const unsigned char *data) > > +{ > > + int i, nr_rates = 0; > > + > > + fp->rates = fp->rate_min = fp->rate_max = 0; > > + > > + for (i = 0; i < nr_triplets; i++) { > > + int min = combine_quad(&data[2 + 12 * i]); > > + int max = combine_quad(&data[6 + 12 * i]); > > + int res = combine_quad(&data[10 + 12 * i]); > > + int rate; > > + > > + if ((max < 0) || (min < 0) || (res < 0) || (max < min)) > > + continue; > > + > > + for (rate = min; rate <= max; rate += res) { > > + if (fp->rate_table) > > + fp->rate_table[nr_rates] = rate; > > + if (!fp->rate_min || rate < fp->rate_min) > > + fp->rate_min = rate; > > + if (!fp->rate_max || rate > fp->rate_max) > > + fp->rate_max = rate; > > + fp->rates |= snd_pcm_rate_to_rate_bit(rate); > > + nr_rates++; > > + > > + /* avoid endless loop */ > > + if (res == 0) > > + break; > > + } > > + } > > + > > + return nr_rates; > > +} > > + > > +/* > > * parse the format descriptor and stores the possible sample rates > > * on the audioformat table (audio class v2). > > */ > > @@ -215,7 +256,7 @@ static int parse_audio_format_rates_v2(struct > > snd_usb_audio *chip, > > { > > struct usb_device *dev = chip->dev; > > unsigned char tmp[2], *data; > > - int i, nr_rates, data_size, ret = 0; > > + int nr_triplets, data_size, ret = 0; > > int clock = snd_usb_clock_find_source(chip, chip->ctrl_intf, > > fp->clock); > > > > if (clock < 0) { > > @@ -237,8 +278,8 @@ static int parse_audio_format_rates_v2(struct > > snd_usb_audio *chip, > > goto err; > > } > > > > - nr_rates = (tmp[1] << 8) | tmp[0]; > > - data_size = 2 + 12 * nr_rates; > > + nr_triplets = (tmp[1] << 8) | tmp[0]; > > + data_size = 2 + 12 * nr_triplets; > > data = kzalloc(data_size, GFP_KERNEL); > > if (!data) { > > ret = -ENOMEM; > > @@ -259,26 +300,29 @@ static int parse_audio_format_rates_v2(struct > > snd_usb_audio *chip, > > goto err_free; > > } > > > > - fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL); > > + /* Call the triplet parser, and make sure fp->rate_table is NULL. > > + * We just use the return value to know how many sample rates we > > + * will have to deal with. */ > > + kfree(fp->rate_table); > > + fp->rate_table = NULL; > > + fp->nr_rates = parse_uac2_sample_rate_range(fp, nr_triplets, data); > > + > > + if (fp->nr_rates == 0) { > > + snd_printk(KERN_ERR "%s(): unable to parse sample rate > > triplets\n", > > + __func__); > > + ret = -EINVAL; > > + goto err_free; > > + } > > + > > + fp->rate_table = kmalloc(sizeof(int) * fp->nr_rates, GFP_KERNEL); > > if (!fp->rate_table) { > > ret = -ENOMEM; > > goto err_free; > > } > > > > - fp->nr_rates = 0; > > - fp->rate_min = fp->rate_max = 0; > > - > > - for (i = 0; i < nr_rates; i++) { > > - int rate = combine_quad(&data[2 + 12 * i]); > > - > > - fp->rate_table[fp->nr_rates] = rate; > > - if (!fp->rate_min || rate < fp->rate_min) > > - fp->rate_min = rate; > > - if (!fp->rate_max || rate > fp->rate_max) > > - fp->rate_max = rate; > > - fp->rates |= snd_pcm_rate_to_rate_bit(rate); > > - fp->nr_rates++; > > - } > > + /* Call the triplet parser again, but this time, fp->rate_table is > > + * allocated, so the rates will be stored */ > > + parse_uac2_sample_rate_range(fp, nr_triplets, data); > > > > err_free: > > kfree(data); > > -- > > 1.7.1 > > > >

4 11

[alsa-devel] usb-audio: some UAC2 fixes
by Daniel Mack 11 Jun '10

11 Jun '10

Hi, here are four small patches that fix some minor bugs in the UAC2 implementation. Thanks to Alex Lee for reporting them. Don't know if it's possible to get them into 2.6.35 as they're no regressions in respect to 2.6.34 (I followed the recent discussions on LKML). However, they fix real bugs and I would also label them stable(a)kernel.org if 2.6.35 would have already been released. So .. I don't know :) I have a number of more patches that do cleanups and implement more functions, but they can certainly go into 2.6.36 and they're not finished yet, so I'm cooking them for some more time. Thanks, Daniel [PATCH 1/4] ALSA: usb-audio: add check for faulty clock in parse_audio_format_rates_v2() [PATCH 2/4] ALSA: usb-audio: fix control messages for USB_RECIP_INTERFACE [PATCH 3/4] ALSA: usb-audio: parse UAC2 sample rate ranges correctly [PATCH 4/4] ALSA: usb-audio: fix UAC2 control value queries

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[PATCH] ALSA: usb-audio: fix UAC2 control value queries
by Daniel Mack 11 Jun '10

11 Jun '10

For RANGE requests, we should only query as much bytes as we're in face interested in. For CUR requests, we shouldn't confuse the firmware with an overlong request but just ask for 2 bytes. This might need fixing in the future as it's not entirely clear when to dispatch 1-byte, 2-byte and 4-byte request blocks. For now, we assume everything is coded in 16bit - this works for all firmware implementations I've seen. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Reported-by: Alex Lee <alexlee188(a)gmail.com> --- sound/usb/mixer.c | 19 ++++++++++++++----- 1 files changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index a060d00..6939d0f 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -297,20 +297,27 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) { - unsigned char buf[14]; /* enough space for one range of 4 bytes */ + unsigned char buf[2 + 3*sizeof(__u16)]; /* enough space for one range */ unsigned char *val; - int ret; + int ret, size; __u8 bRequest; - bRequest = (request == UAC_GET_CUR) ? - UAC2_CS_CUR : UAC2_CS_RANGE; + if (request == UAC_GET_CUR) { + bRequest = UAC2_CS_CUR; + size = sizeof(__u16); + } else { + bRequest = UAC2_CS_RANGE; + size = sizeof(buf); + } + + memset(buf, 0, sizeof(buf)); ret = snd_usb_ctl_msg(cval->mixer->chip->dev, usb_rcvctrlpipe(cval->mixer->chip->dev, 0), bRequest, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, cval->mixer->ctrlif | (cval->id << 8), - buf, sizeof(buf), 1000); + buf, size, 1000); if (ret < 0) { snd_printk(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", @@ -318,6 +325,8 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v return ret; } + /* FIXME: how should we handle multiple triplets here? */ + switch (request) { case UAC_GET_CUR: val = buf; -- 1.7.1

1 0

[alsa-devel] [PATCH 0/3] EP93xx: Add i2s audio support
by Ryan Mallon 11 Jun '10

11 Jun '10

Repost of the i2s audio series for ep93xx. Patch 2/3 is unchanged. The core support has been modified to move setting of the SPOL and ORIDE bits of the i2sclkdiv register, as well as the i2s pin strapping configuration, to a new function called ep93xx_i2s_acquire which is called from the audio board drivers. Ryan Mallon (3): EP93xx: Add i2s core support EP93xx: Add i2s audio driver EP93xx: Add Snapper CL15 i2s audio support arch/arm/mach-ep93xx/clock.c | 67 +++- arch/arm/mach-ep93xx/core.c | 67 +++ arch/arm/mach-ep93xx/include/mach/ep93xx-regs.h | 10 + arch/arm/mach-ep93xx/include/mach/platform.h | 3 + arch/arm/mach-ep93xx/snappercl15.c | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/ep93xx/Kconfig | 18 + sound/soc/ep93xx/Makefile | 11 + sound/soc/ep93xx/ep93xx-i2s.c | 487 +++++++++++++++++++++++ sound/soc/ep93xx/ep93xx-i2s.h | 18 + sound/soc/ep93xx/ep93xx-pcm.c | 319 +++++++++++++++ sound/soc/ep93xx/ep93xx-pcm.h | 22 + sound/soc/ep93xx/snappercl15.c | 150 +++++++ 14 files changed, 1174 insertions(+), 1 deletions(-) create mode 100644 sound/soc/ep93xx/Kconfig create mode 100644 sound/soc/ep93xx/Makefile create mode 100644 sound/soc/ep93xx/ep93xx-i2s.c create mode 100644 sound/soc/ep93xx/ep93xx-i2s.h create mode 100644 sound/soc/ep93xx/ep93xx-pcm.c create mode 100644 sound/soc/ep93xx/ep93xx-pcm.h create mode 100644 sound/soc/ep93xx/snappercl15.c

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[PATCH] ALSA: usb-audio: parse UAC2 sample rate ranges correctly
by Daniel Mack 11 Jun '10

11 Jun '10

A device may report its supported sample rates in ranges rather than in discrete triplets. The code used to only parse the MIN field instead of properly paying attention to the MAX and RES values. Also, handle RES values of 1 correctly and announce a continous sample rate range in this case. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Reported-by: Alex Lee <alexlee188(a)gmail.com> --- sound/usb/format.c | 92 +++++++++++++++++++++++++++++++++++++++++---------- 1 files changed, 74 insertions(+), 18 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index 8eccf17..30364ab 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -206,6 +206,60 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof } /* + * Helper function to walk the array of sample rate triplets reported by + * the device. The problem is that we need to parse whole array first to + * get to know how many sample rates we have to expect. + * Then fp->rate_table can be allocated and filled. + */ +static int parse_uac2_sample_rate_range(struct audioformat *fp, int nr_triplets, + const unsigned char *data) +{ + int i, nr_rates = 0; + + fp->rates = fp->rate_min = fp->rate_max = 0; + + for (i = 0; i < nr_triplets; i++) { + int min = combine_quad(&data[2 + 12 * i]); + int max = combine_quad(&data[6 + 12 * i]); + int res = combine_quad(&data[10 + 12 * i]); + int rate; + + if ((max < 0) || (min < 0) || (res < 0) || (max < min)) + continue; + + /* + * for ranges with res == 1, we announce a continuous sample + * rate range, and this function should return 0 for no further + * parsing. + */ + if (res == 1) { + fp->rate_min = min; + fp->rate_max = max; + fp->rates = SNDRV_PCM_RATE_CONTINUOUS; + return 0; + } + + for (rate = min; rate <= max; rate += res) { + if (fp->rate_table) + fp->rate_table[nr_rates] = rate; + if (!fp->rate_min || rate < fp->rate_min) + fp->rate_min = rate; + if (!fp->rate_max || rate > fp->rate_max) + fp->rate_max = rate; + fp->rates |= snd_pcm_rate_to_rate_bit(rate); + + nr_rates++; + + /* avoid endless loop */ + if (res == 0) + break; + } + } + + return nr_rates; +} + +/* * parse the format descriptor and stores the possible sample rates * on the audioformat table (audio class v2). */ @@ -215,7 +269,7 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, { struct usb_device *dev = chip->dev; unsigned char tmp[2], *data; - int i, nr_rates, data_size, ret = 0; + int nr_triplets, data_size, ret = 0; int clock = snd_usb_clock_find_source(chip, chip->ctrl_intf, fp->clock); if (clock < 0) { @@ -237,8 +291,8 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, goto err; } - nr_rates = (tmp[1] << 8) | tmp[0]; - data_size = 2 + 12 * nr_rates; + nr_triplets = (tmp[1] << 8) | tmp[0]; + data_size = 2 + 12 * nr_triplets; data = kzalloc(data_size, GFP_KERNEL); if (!data) { ret = -ENOMEM; @@ -259,26 +313,28 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, goto err_free; } - fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL); + /* Call the triplet parser, and make sure fp->rate_table is NULL. + * We just use the return value to know how many sample rates we + * will have to deal with. */ + kfree(fp->rate_table); + fp->rate_table = NULL; + fp->nr_rates = parse_uac2_sample_rate_range(fp, nr_triplets, data); + + if (fp->nr_rates == 0) { + /* SNDRV_PCM_RATE_CONTINUOUS */ + ret = 0; + goto err_free; + } + + fp->rate_table = kmalloc(sizeof(int) * fp->nr_rates, GFP_KERNEL); if (!fp->rate_table) { ret = -ENOMEM; goto err_free; } - fp->nr_rates = 0; - fp->rate_min = fp->rate_max = 0; - - for (i = 0; i < nr_rates; i++) { - int rate = combine_quad(&data[2 + 12 * i]); - - fp->rate_table[fp->nr_rates] = rate; - if (!fp->rate_min || rate < fp->rate_min) - fp->rate_min = rate; - if (!fp->rate_max || rate > fp->rate_max) - fp->rate_max = rate; - fp->rates |= snd_pcm_rate_to_rate_bit(rate); - fp->nr_rates++; - } + /* Call the triplet parser again, but this time, fp->rate_table is + * allocated, so the rates will be stored */ + parse_uac2_sample_rate_range(fp, nr_triplets, data); err_free: kfree(data); -- 1.7.1

1 0

Re: [alsa-devel] [sdr-widget] UAC2 sampling freq setting
by Daniel Mack 11 Jun '10

11 Jun '10

(Moving this discussion from sdr-widget to alsa-devel. Can't Cc: the original mailing list as it forbids cross-posting) On Thu, Jun 10, 2010 at 09:36:48PM -0700, Alex Lee wrote: > I studied the driver source again and it becomes apparent that the > driver only looks at the MIN value of the triplet (MIN, MAX, RES). It > does not take into consideration of the MAX and the RES in each > triplet. For samplerates it does that, yes. > The driver calculates its own MIN and MAX values by going through all > the triplets in the clock source. In the present firmware, we only > have ONE triplet in each clock source, with: > > # of triplets = 1 (2 byte value) > MIN 48000 > MAX 96000 > RES 48000 Ah, that's a corner case I didn't see. I'll fix this. > in accordance with the UAC2 specs. The present driver code expects: > > # of triplets = 2 (2 byte value) > triplet 1: > > MIN 48000 MAX 48000 RES 00000 > > triplet 2: > > MIN 96000 MAX 96000 RES 00000 Yes, that's right. All firmware code I've seen so far also has a triplet for each discrete sample rate. > Thus the present driver cannot decode the 1st case above correctly, > and only get the 48khz value. > > I will update the firmware to cater to the 2nd case above and we > should have the 96khz back :-) Ok, fair fix. I'll make sure future versions of the driver also support the other approach. Thanks, Daniel

1 0

[alsa-devel] HDMI audio with NVIDIA NVS 3100M
by Jakub Schmidtke 11 Jun '10

11 Jun '10

Hi. I am having problems getting the HDMI audio working with NVIDIA NVS 3100M card (this is a card in Lenovo T410 laptop). When I'm not using any options for the snd-hda-intel module, I have 4 controls in alsamixer: S/PDIF >S/PDIF 1 S/PDIF 2 S/PDIF 3. Playing to any of the 4 available devices doesn't produce any sound through TV. In a thread about GT2XX cards ( http://forum.xbmc.org/showthread.php?p=527386 ) there was a solution - adding "options snd-hda-intel enable_msi=0 probe_mask=0xffff,0xfff2" to modprobe's config. When I am using this option I have only one S/PDIF output, but I am still not getting any sound through TV. I have run alsa-info.sh script and uploaded the results (this is without any options for the modprobe): http://www.alsa-project.org/db/?f=7514a673a73aca75d58c3e426c59d0bf6ca3dfc8 This is info from lspci about this card: 01:00.0 VGA compatible controller: nVidia Corporation NVIDIA NVS 3100M (rev a2) 01:00.0 0300: 10de:0a6c (rev a2) 01:00.1 Audio device: nVidia Corporation High Definition Audio Controller (rev a1) 01:00.1 0403: 10de:0be3 (rev a1) Any hints? Maybe I can debug or check something else? I am ok with modifying and building kernel modules, I just have no idea where to start... Thanks!

1 0

[alsa-devel] Speaker auto-mute doesn't work on ALC269
by David Song 10 Jun '10

10 Jun '10

When a headphone is plugged, I can hear sound from both speaker and headphone. This issue exists on ASUS EeePC S101 and EeePC 1005PE which both have ALC269. I'm testing with the latest kernel. Any regression here?

2 1

Re: [alsa-devel] [PATCH] NUC900: patch for fix build error
by Liam Girdwood 10 Jun '10

10 Jun '10

On Thu, 2010-06-10 at 10:40 +0800, Wan ZongShun wrote: > This patch is to change 'auido.h' to 'audio.h' for > fixing nuc900 alsa driver build error. > > > Signed-off-by: Wan ZongShun <mcuos.com(a)gmail.com> Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

2 1

[alsa-devel] New release of amidicat program, connecting MIDI to stdin/out
by Josh Lehan 10 Jun '10

10 Jun '10

Hello. On the mailing list a few months ago, mentioned a program I had written, then called "midicat". Found out there's another product by that name. So, the new name of the program is "amidicat". The "a" prefix also clarifies that it's an ALSA program. It's a program that connects the ALSA MIDI sequencer to standard input and output. Among other things, this is fun for prototyping quick MIDI one-liners on the command line, such as "echo 903C7F | amidicat" to play Middle C. I have made a new release of the program, which is basically unchanged, except for a few bug fixes (thanks to Valgrind for discovering a minor memory leak). The biggest improvement is that there's now a manual page and a DocBook documentation page. Here's the webpage for amidicat, if interested: http://www.krellan.com/amidicat/ The download is there. Have fun, and thanks for reading this. Josh Lehan

2 2

[alsa-devel] [PATCH v3 0/5] OMAP/ASoC: McBSP: FIFO handling related fixes
by Peter Ujfalusi 10 Jun '10

10 Jun '10

Hello, Changes since RFC v2: - Comments added explaining the changes, and documenting the API in patch2 - In patch5: - Variable names changed in the hw_rule function - hw_rule function got renamed - The snd_pcm_hw_rule_add calls moved so it anly got set on OMAP3 Intro mail from the previous series: This series aims to correct how the McBSP FIFO is viewed, and handled. Introduction of the problem: OMAP McBSP FIFO is word structured: McBSP2 has 1024 + 256 = 1280 word long buffer, McBSP1,3,4,5 has 128 word long buffer This means, that the size of the FIFO depends on the McBSP word size configuration. For example on McBSP3: 16bit samples: size is 128 * 2 = 256 bytes 32bit samples: size is 128 * 4 = 512 bytes It is simpler to place constraint for buffer and period based on channels. McBSP3 as example again (16 or 32 bit samples): 1 channel (mono): size is 128 frames (128 words) 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words) 4 channels: size is 128 / 4 = 32 frames (4 * 32 words) Since now the McBSP codec supports not only 16bit samples (32biut has been recently added), the FIFO size handling is no longer correct, since it has been hard wired for 16bit word length. The series changes how the users of McBSP are configuring the FIFO: It used to be 0 based (0 meant 1 word threshold). After this series users can configure the threshold in 1 base mode (1 means 1 word threshold). The platform code now provides the _full_ size of the FIFO in words, instead of the already limited value used in the past. In ASoC omap-mcbsp code hw_rule based constraint refinement is going to be used instead of the hardwired static constraint, which was correct only in case of 16bit word length. The hw_rule is refining the minimum buffer size based on the channel number going to be used by the coming stream. In case of threshold mode additional hw_rule refines the maximum allowed period size. The series are generated agains Takashi's sound-2.6: topic/asoc branch. CCing also Eduardo, and Eero since they have worked on the original FIFO/threshold implementation. All commetns and testers are welcome! Peter --- Peter Ujfalusi (5): OMAP: McBSP: Function to query the FIFO size OMAP2: McBSP: Change the way how the FIFO is handled OMAP2: McBSP: Use the port's buffer_size when calculating tx delay ASoC: omap-mcbsp: Save, and use wlen for threshold configuration ASoC: omap-mcbsp: Place correct constraints for streams arch/arm/mach-omap2/mcbsp.c | 10 ++-- arch/arm/plat-omap/include/plat/mcbsp.h | 2 + arch/arm/plat-omap/mcbsp.c | 51 ++++++++++----- sound/soc/omap/omap-mcbsp.c | 113 ++++++++++++++++++++++++------- 4 files changed, 129 insertions(+), 47 deletions(-)

5 12

[alsa-devel] External samplerate changes, UAC2 clock topologies
by Daniel Mack 10 Jun '10

10 Jun '10

Hi, when I implemented the basics for parsing the UAC2 clock functions, I came across some principal questions I would like to discuss. As I wrote in the commit log, UAC2 supports a number of clock units which can be combined to complex topologies. The single unit types are - clock sources (can be fixed, variable, programmable and synced to the USB clock) - clock switches, which let the host decide which one out of many input pins appears on its output pin - clock multipliers, which have a numerator and a denominator, and both can be read-only or read-write, respectively A clock source has controls for its frequency and its validity, a clock selector has a control to control its current state, and a clock multiplier has controls for its numerator and denominator The driver is currently able to walk the graph of connected units in order to find CLOCK_SOURCE descriptors as end-leafs. This is mandatory as all requests for sample rate queries have to be sent to this unit type. This raises the first question, as the driver can only know which sample rates are supported in the currently selected combination of selectors. Once the user changes any of them (which is currently already possible as clock selectors are exported as mixer controls), everything can change. And this can even happen while the stream is active, because there's currently no check about whether the clock selector is part of the actively used path. The second issue I see is that a clock can lose its validity. A real-life example is an external S/PDIF connected device which provides a clock and which is suddenly disconnected. Firmwares are expected to notify the host about such cases, and these messages are trivial to dispatch. However, I wonder how the driver should react on this. From a user's perspective, it would be best to just make the driver find another clock path which reports a valid clock source endpoint, changes the sample rate accordingly and continues streaming. There would be a gap in the stream of course, but at least it would not kill the applications or require major exception handling in userspace. I wonder which approaches are actually possible to implement, which details in the ALSA core would need to be extended, and so on. Any oppinions? Has this been done before for any other audio hardware supported by ALSA? Thanks, Daniel

5 6

[alsa-devel] Mixer volume/decibel mapping
by Sebastian H. 10 Jun '10

10 Jun '10

Hello everyone I'm working on an ALSA-Mixer application for C++/Qt4 and stumbled over volume and decibel mapping for sliders. My question: Is there always an 1 to 1 mapping from volume to decibel and vice versa or can there be decibel values in between two volume steps? Regards, Sebastian H.

7 38

[alsa-devel] snd_mixer handle in two threads
by Sebastian H. 09 Jun '10

09 Jun '10

Hello again This is still about the alsamixer-qt4 application I working on. This time I wonder if a snd_mixer_t handle created with snd_mixer_open ( ... ) can be used in two threads concurrently. One thread should be the main Qt4 GUI thread and the second thread should only poll ( ... ) blocking for ALSA events on the shared snd_mixer_t handle in accompany with snd_mixer_poll_descriptors_count ( ... ) snd_mixer_poll_descriptors ( ... ) snd_mixer_poll_descriptors_revents ( ... ) The second thread then will post Qt4 signals to the main thread on occassion. Does this sound feasible? This approach seems neccessary as I want to avoid timed polls and any integration of linux poll() in the Qt4 mail loop seems to boil down to timed polls. Regards, Sebastian H.

1 1

[alsa-devel] Analog Devices AD1984B HP TouchSmart iq504t No Mic Input
by Brandon Ros 09 Jun '10

09 Jun '10

http://www.alsa-project.org/db/?f=053c6a4c3bc19b4992e3c98d0336a375067c8db3 Despite various models used in /etc/modprobe.d/modprobe.conf (touchsmart, auto, laptop, mobile, thinkpad) and various alsamixer settings, I am unable to get input from the two microphones mounted on my monitor as seen in this picture: http://h10025.www1.hp.com/ewfrf-JAVA/Doc/images/c01481326.jpg Playback works fine.

1 0

[alsa-devel] Alsa + PXA + Kernel 2.6.33 resume issue
by Shane Volpe 09 Jun '10

09 Jun '10

I have upgraded an embedded system (PXA270) to the 2.6.33 kernel (from 2.6.23). Now when I run an audio program (ogg123) and put the system into standby then wake the system up the audio does not work (continue playing) after resume until I terminate ogg123 and start again. After restarting ogg123 the audio works fine with out reseting the system or changing anything else on the system. The system is using the UCB1400 AC97 codec and just as added information when the audio is locked up, after resume, the touchscreen on the system still works which is also controlled by the UCB1400 audio codec. I have also tried several other audio players and the results are the same. Has anyone else seen this bug and does anyone have suggestions on further tests to perform to narrow down were the bug may reside? Regards, Shane

1 0

[alsa-devel] Open source UAC2 sound card for UAC2 driver testing
by Alex Lee 09 Jun '10

09 Jun '10

Hi all, We are developing an open source high speed USB sound card. Please see: http://code.google.com/p/sdr-widget/ for details. With the alpha-1 hardware, we are doing development and testing of the UAC2 firmware against the latest alsa driver here. We have already demonstrated 96khz 24bit stereo capture with UAC2 and we will proceed to 192khz soon. Alpha-1 has an ENOB of at least 20 and we hope alpha-2 will have an ENOB of 22 or better. Alpha-2 kits are available for order now. (Note that these are prototype kits for experimenters only.) With both the UAC2 driver and the sound card firmware being open source, we hope to expedite full UAC2 spec compliance. We are already working closely with Daniel Mack on the testing. Alex

2 1

Re: [alsa-devel] [PATCH v2 4/4]AsoC/nuc900: fix a typo and rename the header file
by Liam Girdwood 09 Jun '10

09 Jun '10

On Wed, 2010-06-02 at 14:05 +0800, Wan ZongShun wrote: > Fix a '#include "nuc900-audio.h' typo, I think it should be 'audio'. > At the same time, this patch renames the 'nuc900-auido.h' file to > 'nuc900-audio.h'. > > Signed-off-by: Wan ZongShun<mcuos.com(a)gmail.com> > --- > sound/soc/nuc900/nuc900-ac97.c | 2 +- > .../soc/nuc900/{nuc900-auido.h => nuc900-audio.h} | 0 > 2 files changed, 1 insertions(+), 1 deletions(-) > rename sound/soc/nuc900/{nuc900-auido.h => nuc900-audio.h} (100%) > > diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c > index db7fc13..caa7c90 100644 > --- a/sound/soc/nuc900/nuc900-ac97.c > +++ b/sound/soc/nuc900/nuc900-ac97.c > @@ -25,7 +25,7 @@ > > #include <mach/mfp.h> > > -#include "nuc900-auido.h" > +#include "nuc900-audio.h" > > static DEFINE_MUTEX(ac97_mutex); > struct nuc900_audio *nuc900_ac97_data; > diff --git a/sound/soc/nuc900/nuc900-auido.h b/sound/soc/nuc900/nuc900-audio.h > similarity index 100% > rename from sound/soc/nuc900/nuc900-auido.h > rename to sound/soc/nuc900/nuc900-audio.h I've just tried to build NUC900 as part of the ASoC multi-component work and it looks like you didn't build this patch before sending it. The old header name is still used in nuc900-audio.c and nuc900-pcm.c. Can you send an incremental patch to fix this. Thanks Liam -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

2 1

[alsa-devel] [PATCH] cs46xx: convert to firmware loader
by Krzysztof Helt 08 Jun '10

08 Jun '10

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The cs46xx uses unlicensed binary blob in the cs46xx_image.h file. Convert this file to binary firmware and use kernel firmare loader. The DSP binary contains 3 parts which are loaded at offsets 0x0, 0x10000 and 0x20000. I have converted this image into 3 binary images cs46xx_0.fw, cs46xx_1.fw and cs46xx_2.fw. This issue was reported as the Linux kernel bug #10750. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- I don't know who else should be informed about this patch. firmware/Makefile | 2 + firmware/WHENCE | 13 + firmware/cs46xx/cs46xx_0.fw.ihex | 769 +++++++++ firmware/cs46xx/cs46xx_1.fw.ihex | 897 ++++++++++ firmware/cs46xx/cs46xx_2.fw.ihex | 1793 ++++++++++++++++++++ include/sound/cs46xx.h | 2 +- sound/pci/Kconfig | 1 + sound/pci/cs46xx/cs46xx.c | 9 +- sound/pci/cs46xx/cs46xx_image.h | 3468 -------------------------------------- sound/pci/cs46xx/cs46xx_lib.c | 60 +- sound/pci/cs46xx/cs46xx_lib.h | 4 - 11 files changed, 3525 insertions(+), 3493 deletions(-) create mode 100644 firmware/cs46xx/cs46xx_0.fw.ihex create mode 100644 firmware/cs46xx/cs46xx_1.fw.ihex create mode 100644 firmware/cs46xx/cs46xx_2.fw.ihex delete mode 100644 sound/pci/cs46xx/cs46xx_image.h diff --git a/firmware/Makefile b/firmware/Makefile index 243409f..b49c61f 100644 --- a/firmware/Makefile +++ b/firmware/Makefile @@ -84,6 +84,8 @@ fw-shipped-$(CONFIG_SCSI_QLOGIC_1280) += qlogic/1040.bin qlogic/1280.bin \ qlogic/12160.bin fw-shipped-$(CONFIG_SCSI_QLOGICPTI) += qlogic/isp1000.bin fw-shipped-$(CONFIG_SMCTR) += tr_smctr.bin +fw-shipped-$(CONFIG_SND_CS46XX) += cs46xx/cs46xx_0.fw cs46xx/cs46xx_1.fw \ + cs46xx/cs46xx_2.fw fw-shipped-$(CONFIG_SND_KORG1212) += korg/k1212.dsp fw-shipped-$(CONFIG_SND_MAESTRO3) += ess/maestro3_assp_kernel.fw \ ess/maestro3_assp_minisrc.fw diff --git a/firmware/WHENCE b/firmware/WHENCE index e8e550f..e904fe9 100644 --- a/firmware/WHENCE +++ b/firmware/WHENCE @@ -26,6 +26,19 @@ Licence: Allegedly GPLv2+, but no source visible. Marked: -------------------------------------------------------------------------- +Driver: cs46xx -- Cirrus Logic (Sound Fusion) CS46xx audio device + +File: cs46xx/cs46xx_0.fw +File: cs46xx/cs46xx_1.fw +File: cs46xx/cs46xx_2.fw + +Licence: Unknown + +This is the content of the cs46xx_image.h file which contained +a DSP code without any license. + +-------------------------------------------------------------------------- + Driver: korg1212 -- Korg 1212 IO audio device File: korg/k1212.dsp diff --git a/firmware/cs46xx/cs46xx_0.fw.ihex b/firmware/cs46xx/cs46xx_0.fw.ihex new file mode 100644 index 0000000..0289edb --- /dev/null +++ b/firmware/cs46xx/cs46xx_0.fw.ihex @@ -0,0 +1,769 @@ +:1000000000000000000000000000000000000000F0 +:1000100000000000000000000000000000000000E0 +:10002000000000000000000063010000000000006C +:1000300000000000000000000000000000000000C0 +:1000400000000000000000000000000000000000B0 +:1000500000000000000000000000000000000000A0 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+:106FA00000000000000000000000000000000000E1 +:106FB00000000000000000000000000000000000D1 +:106FC00000000000000000000000000000000000C1 +:106FD00000000000000000000000000000000000B1 +:106FE00000000000000000000000000000000000A1 +:106FF0000000000000000000000000000000000091 +:00000001FF diff --git a/include/sound/cs46xx.h b/include/sound/cs46xx.h index e3005a6..89edf87 100644 --- a/include/sound/cs46xx.h +++ b/include/sound/cs46xx.h @@ -1739,7 +1739,7 @@ int snd_cs46xx_pcm_iec958(struct snd_cs46xx *chip, int device, struct snd_pcm ** int snd_cs46xx_pcm_center_lfe(struct snd_cs46xx *chip, int device, struct snd_pcm **rpcm); int snd_cs46xx_mixer(struct snd_cs46xx *chip, int spdif_device); int snd_cs46xx_midi(struct snd_cs46xx *chip, int device, struct snd_rawmidi **rmidi); -int snd_cs46xx_start_dsp(struct snd_cs46xx *chip); +int snd_cs46xx_start_dsp(struct snd_card *card); int snd_cs46xx_gameport(struct snd_cs46xx *chip); #endif /* __SOUND_CS46XX_H */ diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index e7a8cd0..00b3066 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -241,6 +241,7 @@ config SND_CS4281 config SND_CS46XX tristate "Cirrus Logic (Sound Fusion) CS4280/CS461x/CS462x/CS463x" + select FW_LOADER select SND_RAWMIDI select SND_AC97_CODEC help diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index 767fa7f..ed659ce 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -36,6 +36,12 @@ MODULE_AUTHOR("Jaroslav Kysela <perex(a)perex.cz>"); MODULE_DESCRIPTION("Cirrus Logic Sound Fusion CS46XX"); MODULE_LICENSE("GPL"); +#ifndef CONFIG_SND_CS46XX_NEW_DSP +MODULE_FIRMWARE("cs46xx/cs46xx_0.fw"); +MODULE_FIRMWARE("cs46xx/cs46xx_1.fw"); +MODULE_FIRMWARE("cs46xx/cs46xx_2.fw"); +#endif + MODULE_SUPPORTED_DEVICE("{{Cirrus Logic,Sound Fusion (CS4280)}," "{Cirrus Logic,Sound Fusion (CS4610)}," "{Cirrus Logic,Sound Fusion (CS4612)}," @@ -129,7 +135,8 @@ static int __devinit snd_card_cs46xx_probe(struct pci_dev *pci, snd_card_free(card); return err; } - if ((err = snd_cs46xx_start_dsp(chip)) < 0) { + err = snd_cs46xx_start_dsp(card); + if (err < 0) { snd_card_free(card); return err; } diff --git a/sound/pci/cs46xx/cs46xx_image.h b/sound/pci/cs46xx/cs46xx_image.h deleted file mode 100644 index dc93f62..0000000 --- a/sound/pci/cs46xx/cs46xx_image.h +++ /dev/null @@ -1,3468 +0,0 @@ -struct BA1struct { - struct { - unsigned long offset; - unsigned long size; - } memory[BA1_MEMORY_COUNT]; - u32 map[BA1_DWORD_SIZE]; -}; - - -static struct BA1struct BA1Struct = { -{{ 0x00000000, 0x00003000 },{ 0x00010000, 0x00003800 },{ 0x00020000, 0x00007000 }}, -{0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000163,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00200040,0x00008010,0x00000000, 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-0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000, -0x00000000,0x00000000,0x00000000,0x00000000} - }; diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index aad3708..11a8207 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -53,7 +53,9 @@ #include <linux/slab.h> #include <linux/gameport.h> #include <linux/mutex.h> - +#ifndef CONFIG_SND_CS46XX_NEW_DSP +#include <linux/firmware.h> +#endif #include <sound/core.h> #include <sound/control.h> @@ -360,22 +362,41 @@ int snd_cs46xx_clear_BA1(struct snd_cs46xx *chip, #else /* old DSP image */ -#include "cs46xx_image.h" - -int snd_cs46xx_download_image(struct snd_cs46xx *chip) +int snd_cs46xx_download_image(struct snd_card *card) { - int idx, err; - unsigned long offset = 0; + struct snd_cs46xx *chip = card->private_data; + int err; + const struct firmware *fw_0 = NULL, *fw_1 = NULL, *fw_2 = NULL; - for (idx = 0; idx < BA1_MEMORY_COUNT; idx++) { - if ((err = snd_cs46xx_download(chip, - &BA1Struct.map[offset], - BA1Struct.memory[idx].offset, - BA1Struct.memory[idx].size)) < 0) - return err; - offset += BA1Struct.memory[idx].size >> 2; - } - return 0; + err = request_firmware(&fw_0, "cs46xx/cs46xx_0.fw", card->dev); + if (err < 0) { + printk(KERN_ERR "cs46xx: Error loading cs46xx_0.fw"); + goto cleanup0; + } + err = request_firmware(&fw_1, "cs46xx/cs46xx_1.fw", card->dev); + if (err < 0) { + printk(KERN_ERR "cs46xx: Error loading cs46xx_1.fw"); + goto cleanup1; + } + err = request_firmware(&fw_2, "cs46xx/cs46xx_2.fw", card->dev); + if (err < 0) { + printk(KERN_ERR "cs46xx: Error loading cs46xx_2.fw"); + goto cleanup2; + } + err = snd_cs46xx_download(chip, (u32 *)fw_0->data, 0x0, fw_0->size); + if (err < 0) + goto cleanup2; + err = snd_cs46xx_download(chip, (u32 *)fw_1->data, 0x10000, fw_1->size); + if (err < 0) + goto cleanup2; + err = snd_cs46xx_download(chip, (u32 *)fw_2->data, 0x20000, fw_2->size); +cleanup2: + release_firmware(fw_2); +cleanup1: + release_firmware(fw_1); +cleanup0: + release_firmware(fw_0); + return err; } #endif /* CONFIG_SND_CS46XX_NEW_DSP */ @@ -3060,8 +3081,9 @@ static void cs46xx_enable_stream_irqs(struct snd_cs46xx *chip) snd_cs46xx_poke(chip, BA1_CIE, tmp); /* capture interrupt enable */ } -int __devinit snd_cs46xx_start_dsp(struct snd_cs46xx *chip) +int __devinit snd_cs46xx_start_dsp(struct snd_card *card) { + struct snd_cs46xx *chip = card->private_data; unsigned int tmp; /* * Reset the processor. @@ -3107,7 +3129,7 @@ int __devinit snd_cs46xx_start_dsp(struct snd_cs46xx *chip) return -EIO; #else /* old image */ - if (snd_cs46xx_download_image(chip) < 0) { + if (snd_cs46xx_download_image(card) < 0) { snd_printk(KERN_ERR "image download error\n"); return -EIO; } @@ -3664,7 +3686,7 @@ int snd_cs46xx_resume(struct pci_dev *pci) for (i = 0; i < ARRAY_SIZE(saved_regs); i++) snd_cs46xx_pokeBA0(chip, saved_regs[i], chip->saved_regs[i]); #else - snd_cs46xx_download_image(chip); + snd_cs46xx_download_image(card); #endif #if 0 diff --git a/sound/pci/cs46xx/cs46xx_lib.h b/sound/pci/cs46xx/cs46xx_lib.h index b518949..3613e70 100644 --- a/sound/pci/cs46xx/cs46xx_lib.h +++ b/sound/pci/cs46xx/cs46xx_lib.h @@ -49,10 +49,6 @@ #define SCB_ON_PARENT_NEXT_SCB 1 #define SCB_ON_PARENT_SUBLIST_SCB 2 -/* 3*1024 parameter, 3.5*1024 sample, 2*3.5*1024 code */ -#define BA1_DWORD_SIZE (13 * 1024 + 512) -#define BA1_MEMORY_COUNT 3 - /* * common I/O routines */ -- 1.6.4 ---------------------------------------------------------------------- Doladuj telefon w promocji! Sprawdz >>> http://linkint.pl/f2708

2 2

[alsa-devel] [PATCH 1/5]alsa:patch_realtek.c Add SSID table for MacBookAir2, 1
by Justin P. Mattock 08 Jun '10

08 Jun '10

This adds the SSID number to snd_pci_quirk for the MacBookAir2,1 taken from codec#0 at: http://launchpadlibrarian.net/49455483/Card0.Codecs.codec.0.txt keep in mind I do not have one of these machines on hand so please if you do have this machine please test for me.. Signed-off-by: Justin P. Mattock <justinmattock(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 17d4548..9b82d28 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9487,6 +9487,7 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5), SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC885_MACMINI3), + SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC885_MBA21), {} /* terminator */ }; -- 1.6.5.2.180.gc5b3e

2 7

[alsa-devel] [PATCH]alsa:patch_realtek.c Add support for the iMac7, 1
by Justin P. Mattock 08 Jun '10

08 Jun '10

This patch adds support for the iMac7,1 Note:I do not have this machine on hand only the codec#0 file for the machine.. located at https://bugs.launchpad.net/mactel-support/+bug/360866 So please test if you have the appropriate equipment, and report any issues to the alsa devel list (alsa-devel(a)alsa-project.org). Signed-off-by: Justin P. Mattock <justinmattock(a)gmail.com> --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_realtek.c | 63 +++++++++++++++++++++++++- 2 files changed, 63 insertions(+), 1 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 1d38b0d..a9aac53 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -127,6 +127,7 @@ ALC882/883/885/888/889 macmini3 Macmini 3,1 mba21 Macbook Air 2,1 mbp3 Macbook Pro rev3 + imac71 iMac 7,1 imac24 iMac 24'' with jack detection imac91 iMac 9,1 w2jc ASUS W2JC diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fc767b6..62ef28d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -213,6 +213,7 @@ enum { ALC885_MBP3, ALC885_MB5, ALC885_MACMINI3, + ALC885_IMAC71, ALC885_IMAC24, ALC885_IMAC91, ALC883_3ST_2ch_DIG, @@ -7493,6 +7494,20 @@ static struct snd_kcontrol_new alc885_macmini3_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc882_imac71_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new alc885_imac91_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT), @@ -7994,6 +8009,25 @@ static struct hda_verb alc885_mbp3_init_verbs[] = { { } }; +/* iMac7,1 */ +static struct hda_verb alc882_imac71_init_verbs[] = { + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ + {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */ + + {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */ + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */ + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ + { } /* end */ +}; + /* iMac 9,1 */ static struct hda_verb alc885_imac91_init_verbs[] = { /* Internal Speaker Pin (0x0c) */ @@ -8099,6 +8133,14 @@ static void alc885_mba21_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x18; } +static void alc885_imac71_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x16; +} static void alc885_mbp3_setup(struct hda_codec *codec) @@ -9300,6 +9342,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [ALC885_MACMINI3] = "macmini3", [ALC885_MBA21] = "mba21", [ALC885_MBP3] = "mbp3", + [ALC885_IMAC71] = "imac71", [ALC885_IMAC24] = "imac24", [ALC885_IMAC91] = "imac91", [ALC883_3ST_2ch_DIG] = "3stack-2ch-dig", @@ -9477,7 +9520,7 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_IMAC24), SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC885_MBP3), SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889A_MB31), - SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_ASUS_A7M), + SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC885_IMAC71), SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC885_MBP3), SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC885_MBA21), SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889A_MB31), @@ -9612,6 +9655,24 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &alc882_capture_source, .init_hook = alc885_macpro_init_hook, }, + [ALC885_IMAC71] = { + .mixers = { alc882_imac71_mixer, alc882_chmode_mixer }, + .init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs, + alc882_imac71_init_verbs}, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc882_adc_nids), + .adc_nids = alc882_adc_nids, + .capsrc_nids = alc882_capsrc_nids, + .num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes), + .channel_mode = alc882_3ST_6ch_modes, + .need_dac_fix = 1, + .input_mux = &alc882_capture_source, + .unsol_event = alc_automute_amp_unsol_event, + .setup = alc885_imac71_setup, + .init_hook = alc_automute_amp, + }, [ALC885_IMAC24] = { .mixers = { alc885_imac24_mixer }, .init_verbs = { alc885_imac24_init_verbs }, -- 1.7.1.rc1.21.gf3bd6

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[alsa-devel] [RFC] Scenario / Use Case manager updates.
by Liam Girdwood 08 Jun '10

08 Jun '10

Some updates have been recently made to the ALSA audio hardware scenario / use case manager in order to provide more flexible support for smart phones and other complex audio devices. The updates are mostly based around the need to provide further simplicity for constructing and de-constructing use case transitions. It's intended these updates will be the last before upstream submission. Use cases can now be divided into three components rather than a single entity as before. This gives the advantage that we can now react better to asynchronous events like a headphone insertion or playing a ringtone for an incoming call when the user is listening to music. It also means we will have less and smaller use case configuration files. The three components of a use case are now :- 1) Verb (or action). This is mandatory and describes (in terms of ALSA control settings) the system use case action. e.g "Phone Call", "Music Playback". 2) Device. This is also mandatory and describes the audio playback or capture device. e.g. "Handset", "Bluetooth", "Headset". 3) Modifier. This is optional and is used to modify the current use case verb. i.e. The user could be listening to music when an incoming phone call requires that a ring tone is played. In this case the use case verb would be set to "Music Playback" when the music playback started and then the "Tone Playback" modifier could be enabled to play the ring tone. The use case verb would then be changed to "Voice Call" if the call was accepted. The updates have not yet made it into the alsa-lib scenario branch here yet :- http://git.alsa-project.org/?p=alsa-lib.git;a=shortlog;h=ascenario But they do exist in the use-case-verb branch here:- http://git.slimlogic.co.uk/cgi-bin/cgit.cgi/alsa-lib.git/log/?h=use-case-ve… (History to be rebased for upstream shortly) The main code changes from the scenario branch are :- o Rename API, files and directories from scenario to use case. o Support for changing use case verb (rather than use case verb + device + modifier as a large less flexible compound use case as before) o Support for enabling multiple use case devices. o Support for enabling multiple use case modifiers. o Added more code documentation to public API header and core. Still todo before upstream:- o Mutex around public API o Implement notifiers. o Support user defined transition sequences between use cases verbs, devices and modifiers. One thing that is still undecided before upstream submission is the management of the use case configuration files for each device/machine type. They could either exist in a directory of alsa-lib or alternatively we could have a separate git repository (with more frequent releases) containing use case configuration data ? Any other options ? Thanks Liam ---- /** * \file include/use-case.h * \brief use case interface for the ALSA driver * \author Liam Girdwood <lrg(a)slimlogic.co.uk> * \author Stefan Schmidt <stefan(a)slimlogic.co.uk> * \author Jaroslav Kysela <perex(a)perex.cz> * \author Justin Xu <justin(a)slimlogic.co.uk> * \date 2008-2010 */ /* * * This library is free software; you can redistribute it and/or modify * it under the terms of the GNU Lesser General Public License as * published by the Free Software Foundation; either version 2.1 of * the License, or (at your option) any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with this library; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * * Copyright (C) 2008-2010 SlimLogic Ltd * Copyright (C) 2010 Wolfson Microelectronics PLC * Copyright (C) 2010 Texas Instruments Inc. */ #ifndef __ALSA_USE_CASE_H #define __ALSA_USE_CASE_H #ifdef __cplusplus extern "C" { #endif /** * \defgroup Use Case Interface * The ALSA Use Case manager interface. * See \ref Usecase page for more details. * \{ */ /** * ALSA Use Case Interface * * The use case manager works by configuring the sound card ALSA kcontrols to * change the hardware digital and analog audio routing to match the requested * device use case. The use case manager kcontrol configurations are stored in * easy to modify text files. * * An audio use case can be defined by a verb and device parameter. The verb * describes the use case action i.e. a phone call, listening to music, recording * a conversation etc. The device describes the physical audio capture and playback * hardware i.e. headphones, phone handset, bluetooth headset, etc. * * It's intended clients will mostly only need to set the use case verb and * device for each system use case change (as the verb and device parameters * cover most audio use cases). * * However there are times when a use case has to be modified at runtime. e.g. * * o Incoming phone call when the device is playing music * o Recording sections of a phone call * o Playing tones during a call. * * In order to allow asynchronous runtime use case adaptations, we have a third * optional modifier parameter that can be used to further configure * the use case during live audio runtime. * * This interface allows clients to :- * * o Query the supported use case verbs, devices and modifiers for the machine. * o Set and Get use case verbs, devices and modifiers for the machine. * o Get the ALSA PCM playback and capture device PCMs for use case verb and * modifier. * o Get the QoS parameter for each use case verb and modifier. * o Get the ALSA master playback and capture volume/switch kcontrols * for each use case. * o Be notified when the use case verb, device, modifier changes. */ /* * Use Case Verb. * * The use case verb is the main device audio action. e.g. the "HiFi" use * case verb will configure the audio hardware for HiFi Music playback * and capture. */ #define SND_USE_CASE_VERB_INACTIVE "Inactive" #define SND_USE_CASE_VERB_HIFI "HiFi" #define SND_USE_CASE_VERB_HIFI_LOW_POWER "HiFi Low Power" #define SND_USE_CASE_VERB_VOICE "Voice" #define SND_USE_CASE_VERB_VOICE_LOW_POWER "Voice Low Power" #define SND_USE_CASE_VERB_VOICECALL "Voice Call" #define SND_USE_CASE_VERB_IP_VOICECALL "Voice Call IP" #define SND_USE_CASE_VERB_ANALOG_RADIO "FM Analog Radio" #define SND_USE_CASE_VERB_DIGITAL_RADIO "FM Digital Radio" /* add new verbs to end of list */ /* * Use Case Device. * * Physical system devices the render and capture audio. Devices can be OR'ed * together to support audio on similtanious devices. */ #define SND_USE_CASE_DEV_NONE "None" #define SND_USE_CASE_DEV_SPEAKER "Speaker" #define SND_USE_CASE_DEV_LINE "Line" #define SND_USE_CASE_DEV_HEADPHONES "Headphones" #define SND_USE_CASE_DEV_HEADSET "Headset" #define SND_USE_CASE_DEV_HANDSET "Handset" #define SND_USE_CASE_DEV_BLUETOOTH "Bluetooth" #define SND_USE_CASE_DEV_EARPIECE "Earpiece" #define SND_USE_CASE_DEV_SPDIF "SPDIF" #define SND_USE_CASE_DEV_HDMI "HDMI" /* add new devices to end of list */ /* * Use Case Modifiers. * * The use case modifier allows runtime configuration changes to deal with * asynchronous events. * * e.g. to record a voice call :- * 1. Set verb to SND_USE_CASE_VERB_VOICECALL (for voice call) * 2. Set modifier SND_USE_CASE_MOD_CAPTURE_VOICE when capture required. * 3. Call snd_use_case_get_verb_capture_pcm() to get ALSA source PCM * with captured voice pcm data. * * e.g. to play a ring tone when listenin to MP3 Music :- * 1. Set verb to SND_USE_CASE_VERB_HIFI (for MP3 playback) * 2. Set modifier to SND_USE_CASE_MOD_PLAY_TONE when incoming call happens. * 3. Call snd_use_case_get_verb_playback_pcm() to get ALSA PCM sink for * ringtone pcm data. */ #define SND_USE_CASE_MOD_CAPTURE_VOICE "Capture Voice" #define SND_USE_CASE_MOD_CAPTURE_MUSIC "Capture Music" #define SND_USE_CASE_MOD_PLAY_MUSIC "Play Music" #define SND_USE_CASE_MOD_PLAY_VOICE "Play Voice" #define SND_USE_CASE_MOD_PLAY_TONE "Play Tone" /* add new modifiers to end of list */ /** * QoS - Quality of Service * * The interface allows clients to determine the audio QoS required for each * use case verb and modifier. It's intended as an optional hint to the * audio driver in order to lower power consumption. * */ enum snd_use_case_qos { SND_USE_CASE_QOS_MUSIC = 0, SND_USE_CASE_QOS_VOICE = 1, SND_USE_CASE_QOS_TONES = 2, }; /* * Use Case Control Aliases. * * Use cases often use different internal hardware paths to route digital and * analog audio. This can mean different controls are used to change volumes * depending on the particular use case. This interface allows clients to * find out the hardware controls associated with each use case. */ enum snd_use_case_control_alias { SND_USE_CASE_ALIAS_PLAYBACK_VOLUME = 0, SND_USE_CASE_ALIAS_PLAYBACK_SWITCH, SND_USE_CASE_ALIAS_CAPTURE_VOLUME, SND_USE_CASE_ALIAS_CAPTURE_SWITCH, }; /** use case container */ typedef struct snd_use_case_mgr snd_use_case_mgr_t; /* * Use case change notifier. * * Clients can be notified when the use case verb, device or modifier changes. */ #define SND_USE_CASE_NOTIFY_VERB 0x1 #define SND_USE_CASE_NOTIFY_DEVICE 0x2 #define SND_USE_CASE_NOTIFY_MODIFIER 0x4 typedef int (*snd_use_case_notifier_t)(snd_use_case_mgr_t *, int notify_status, void *data); /** * \brief List supported use case verbs for given soundcard * \param uc_mgr Use case manager * \param verb Returned list of supported use case verbs * \return Number of use case verbs if success, otherwise a negative error code */ int snd_use_case_get_verb_list(snd_use_case_mgr_t *uc_mgr, const char **verb[]); /** * \brief List supported use case devices for given use case verb * \param uc_mgr Use case manager * \param verb Verb name. * \param device Returned list of supported use case devices * \return Number of use case devices if success, otherwise a negative error code */ int snd_use_case_get_device_list(snd_use_case_mgr_t *uc_mgr, const char *verb, const char **device[]); /** * \brief List supported use case verb modifiers for given verb * \param uc_mgr use case manager * \param verb verb id. * \param mod returned list of supported use case modifier id and names * \return number of use case modifiers if success, otherwise a negative error code */ int snd_use_case_get_mod_list(snd_use_case_mgr_t *uc_mgr, const char *verb, const char **mod[]); /** * \brief Set new use case verb for sound card * \param uc_mgr Use case manager * \param verb Verb * \return Zero if success, otherwise a negative error code */ int snd_use_case_set_verb(snd_use_case_mgr_t *uc_mgr, const char *verb); /** * \brief Enable use case device for current use case verb * \param uc_mgr Use case manager * \param device Device * \return Zero if success, otherwise a negative error code */ int snd_use_case_enable_device(snd_use_case_mgr_t *uc_mgr, const char *device); /** * \brief Disable use case device for current use case verb * \param uc_mgr Use case manager * \param device Device * \return Zero if success, otherwise a negative error code */ int snd_use_case_disable_device(snd_use_case_mgr_t *uc_mgr, const char *device); /** * \brief Enable use case modifier for current use case verb * \param uc_mgr Use case manager * \param modifier Modifier * \return Zero if success, otherwise a negative error code */ int snd_use_case_enable_modifier(snd_use_case_mgr_t *uc_mgr, const char *modifier); /** * \brief Disable use case modifier for curent use case verb * \param uc_mgr Use case manager * \param modifier Modifier * \return Zero if success, otherwise a negative error code */ int snd_use_case_disable_modifier(snd_use_case_mgr_t *uc_mgr, const char *modifier); /** * \brief Get current use case verb for sound card * \param uc_mgr Use case manager * \return Verb if success, otherwise NULL */ const char *snd_use_case_get_verb(snd_use_case_mgr_t *uc_mgr); /** * \brief Get device status for current use case verb * \param uc_mgr Use case manager * \param device_name The device we are interested in. * \return - 1 = enabled, 0 = disabled, negative = error */ int snd_use_case_get_device_status(snd_use_case_mgr_t *uc_mgr, const char *device_name); /** * \brief Get modifier status for current use case verb * \param uc_mgr Use case manager * \param device_name The device we are interested in. * \return - 1 = enabled, 0 = disabled, negative = error */ int snd_use_case_get_modifier_status(snd_use_case_mgr_t *uc_mgr, const char *modifier_name); /** * \brief Get the current use case verb QoS * \param uc_mgr Use case manager * \return QoS level */ enum snd_use_case_qos snd_use_case_get_verb_qos(snd_use_case_mgr_t *uc_mgr); /** * \brief Get use case modifier QoS * \param uc_mgr use case manager * \param modifier Modifier * \return QoS level */ enum snd_use_case_qos snd_use_case_get_mod_qos(snd_use_case_mgr_t *uc_mgr, const char *modifier); /** * \brief Get the current use case verb playback PCM sink ID. * \param uc_mgr use case manager * \return PCM number if success, otherwise negative */ int snd_use_case_get_verb_playback_pcm(snd_use_case_mgr_t *uc_mgr); /** * \brief Get the current use case verb capture PCM source ID * \param uc_mgr Use case manager * \return PCM number if success, otherwise negative */ int snd_use_case_get_verb_capture_pcm(snd_use_case_mgr_t *uc_mgr); /** * \brief Get use case modifier playback PCM sink ID * \param uc_mgr Use case manager * \param modifier Modifier * \return PCM number if success, otherwise negative */ int snd_use_case_get_mod_playback_pcm(snd_use_case_mgr_t *uc_mgr, const char *modifier); /** * \brief Get use case modifier capture PCM source ID * \param uc_mgr Use case manager * \param modifier Modifier * \return PCM number if success, otherwise negative */ int snd_use_case_get_mod_capture_pcm(snd_use_case_mgr_t *uc_mgr, const char *modifier); /** * \brief Get ALSA volume/mute control IDs depending on use case device. * \param uc_mgr Use case manager * \param type The control type we are looking for * \param device The device we are interested in. * \return ID if success, otherwise a negative error code * * Get the control id for common volume and mute controls that are aliased * in the current use case verb. */ int snd_use_case_get_device_ctl_elem_id(snd_use_case_mgr_t *uc_mgr, enum snd_use_case_control_alias type, const char *device); /** * \brief Get ALSA volume/mute control IDs depending on use case modifier. * \param uc_mgr Use case manager * \param type The control type we are looking for * \param modifier The modifier we are interested in. * \return ID if success, otherwise a negative error code * * Get the control id for common volume and mute controls that are aliased * in the current use case modifier. */ int snd_use_case_get_modifier_ctl_elem_id(snd_use_case_mgr_t *uc_mgr, enum snd_use_case_control_alias type, const char *modifier); /** * \brief Clients can register to be notified when use case changes. * \param notifier Client notify callback. * \param data Client private data. * \return 0 on success, otherwise errno */ int snd_use_case_register_notifier(snd_use_case_mgr_t *uc_mgr, snd_use_case_notifier_t notifier, void *data); /** * \brief Unregister client notifier. * \param notifier Client notify callback. * \return 0 on success, otherwise errno */ int snd_use_case_unregister_notifier(snd_use_case_mgr_t *uc_mgr, snd_use_case_notifier_t notifier); /** * \brief Open and initialise use case core for sound card * \param card_name Sound card name. * \param shared Will more than 1 client control/monitor use cases. * \return Use case handle if success, otherwise NULL */ snd_use_case_mgr_t *snd_use_case_mgr_open(const char *card_name, int shared); /** * \brief Reload and re-parse use case configuration files for sound card. * \param uc_mgr Use case manager * \return zero if success, otherwise a negative error code */ int snd_use_case_mgr_reload(snd_use_case_mgr_t *uc_mgr); /** * \brief Close use case manager * \param uc_mgr Use case manager * \return zero if success, otherwise a negative error code */ int snd_use_case_mgr_close(snd_use_case_mgr_t *uc_mgr); /** * \brief Dump current sound card use case control settings * \param uc_mgr Use case manager * \param output Output handle * \return zero if success, otherwise a negative error code */ int snd_use_case_dump(snd_use_case_mgr_t *uc_mgr, snd_output_t *output); /** * \} */ #ifdef __cplusplus } #endif #endif /* __ALSA_USE_CASE_H */

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Re: [alsa-devel] [PATCH] sound/spi: patch for the unuseful variable removal
by Mark Brown 08 Jun '10

08 Jun '10

On Fri, Jun 04, 2010 at 04:30:24PM +0800, Wan ZongShun wrote: > The '*bitclk' of structure 'snd_at73c213' seems no use > ,so I make a patch to remove the unnecessary variable. > > Signed-off-by: Wan ZongShun <mcuos.com(a)gmail.com> We should probably just remove this driver, it should be done within ASoC rather than like this.

2 2

Re: [alsa-devel] [PATCH 2/2] pxa: modify the pxa2xx-ac97 file for pxa168
by Eric Miao 08 Jun '10

08 Jun '10

2010/6/8 Wan ZongShun <mcuos.com(a)gmail.com>: > In order to allow the sharing of the pxa2xx-ac97 driver file with pxa168, > we have to do a little change to pxa2xx-ac97 file. > > All private register addresses were located in mach/include/ac97.h. This is not the right way to go, as suggested already. I hope the ac97 driver is more self-contained. i.e. having a platform device for: - where the registers are mapped from - drcmr channel numbers can be encoded within dma resource - ac97 register definitions are not exposed everywhere

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[alsa-devel] [PATCH] atmel: set "channel A event" output to debug
by Yegor Yefremov 08 Jun '10

08 Jun '10

Signed-off-by: Yegor Yefremov <yegorslists(a)googlemail.com> Index: b/sound/atmel/ac97c.c =================================================================== --- a/sound/atmel/ac97c.c 2010-06-01 15:28:23.000000000 +0200 +++ b/sound/atmel/ac97c.c 2010-06-03 15:20:43.000000000 +0200 @@ -657,7 +657,7 @@ if (sr & AC97C_SR_CAEVT) { struct snd_pcm_runtime *runtime; int offset, next_period, block_size; - dev_info(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n", + dev_dbg(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n", casr & AC97C_CSR_OVRUN ? " OVRUN" : "", casr & AC97C_CSR_RXRDY ? " RXRDY" : "", casr & AC97C_CSR_UNRUN ? " UNRUN" : "",

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[alsa-devel] [RFC] pm_qos: get rid of the allocation in pm_qos_add_request()
by James Bottomley 08 Jun '10

08 Jun '10

[alsa-devel says it's a moderated list, so feel free to drop before replying] Since every caller has to squirrel away the returned pointer anyway, they might as well supply the memory area. This fixes a bug in a few of the call sites where the returned pointer was dereferenced without checking it for NULL (which gets returned if the kzalloc failed). I'd like to hear how sound and netdev feels about this: it will add about two more pointers worth of data to struct netdev and struct snd_pcm_substream .. but I think it's worth it. If you're OK, I'll add your acks and send through the pm tree. This also looks to me like an android independent clean up (even though it renders the request_add atomically callable). I also added include guards to include/linux/pm_qos_params.h James --- drivers/net/e1000e/netdev.c | 17 ++++------ drivers/net/igbvf/netdev.c | 9 ++--- drivers/net/wireless/ipw2x00/ipw2100.c | 12 +++--- include/linux/netdevice.h | 2 +- include/linux/pm_qos_params.h | 12 +++++-- include/sound/pcm.h | 2 +- kernel/pm_qos_params.c | 55 ++++++++++++++++--------------- sound/core/pcm_native.c | 12 ++----- 8 files changed, 60 insertions(+), 61 deletions(-) diff --git a/drivers/net/e1000e/netdev.c b/drivers/net/e1000e/netdev.c index 24507f3..47ea62f 100644 --- a/drivers/net/e1000e/netdev.c +++ b/drivers/net/e1000e/netdev.c @@ -2901,10 +2901,10 @@ static void e1000_configure_rx(struct e1000_adapter *adapter) * dropped transactions. */ pm_qos_update_request( - adapter->netdev->pm_qos_req, 55); + &adapter->netdev->pm_qos_req, 55); } else { pm_qos_update_request( - adapter->netdev->pm_qos_req, + &adapter->netdev->pm_qos_req, PM_QOS_DEFAULT_VALUE); } } @@ -3196,9 +3196,9 @@ int e1000e_up(struct e1000_adapter *adapter) /* DMA latency requirement to workaround early-receive/jumbo issue */ if (adapter->flags & FLAG_HAS_ERT) - adapter->netdev->pm_qos_req = - pm_qos_add_request(PM_QOS_CPU_DMA_LATENCY, - PM_QOS_DEFAULT_VALUE); + pm_qos_add_request(&adapter->netdev->pm_qos_req, + PM_QOS_CPU_DMA_LATENCY, + PM_QOS_DEFAULT_VALUE); /* hardware has been reset, we need to reload some things */ e1000_configure(adapter); @@ -3263,11 +3263,8 @@ void e1000e_down(struct e1000_adapter *adapter) e1000_clean_tx_ring(adapter); e1000_clean_rx_ring(adapter); - if (adapter->flags & FLAG_HAS_ERT) { - pm_qos_remove_request( - adapter->netdev->pm_qos_req); - adapter->netdev->pm_qos_req = NULL; - } + if (adapter->flags & FLAG_HAS_ERT) + pm_qos_remove_request(&adapter->netdev->pm_qos_req); /* * TODO: for power management, we could drop the link and diff --git a/drivers/net/igbvf/netdev.c b/drivers/net/igbvf/netdev.c index 5e2b2a8..5da569f 100644 --- a/drivers/net/igbvf/netdev.c +++ b/drivers/net/igbvf/netdev.c @@ -48,7 +48,7 @@ #define DRV_VERSION "1.0.0-k0" char igbvf_driver_name[] = "igbvf"; const char igbvf_driver_version[] = DRV_VERSION; -struct pm_qos_request_list *igbvf_driver_pm_qos_req; +struct pm_qos_request_list igbvf_driver_pm_qos_req; static const char igbvf_driver_string[] = "Intel(R) Virtual Function Network Driver"; static const char igbvf_copyright[] = "Copyright (c) 2009 Intel Corporation."; @@ -2902,8 +2902,8 @@ static int __init igbvf_init_module(void) printk(KERN_INFO "%s\n", igbvf_copyright); ret = pci_register_driver(&igbvf_driver); - igbvf_driver_pm_qos_req = pm_qos_add_request(PM_QOS_CPU_DMA_LATENCY, - PM_QOS_DEFAULT_VALUE); + pm_qos_add_request(igbvf_driver_pm_qos_req, PM_QOS_CPU_DMA_LATENCY, + PM_QOS_DEFAULT_VALUE); return ret; } @@ -2918,8 +2918,7 @@ module_init(igbvf_init_module); static void __exit igbvf_exit_module(void) { pci_unregister_driver(&igbvf_driver); - pm_qos_remove_request(igbvf_driver_pm_qos_req); - igbvf_driver_pm_qos_req = NULL; + pm_qos_remove_request(&igbvf_driver_pm_qos_req); } module_exit(igbvf_exit_module); diff --git a/drivers/net/wireless/ipw2x00/ipw2100.c b/drivers/net/wireless/ipw2x00/ipw2100.c index 0bd4dfa..7f0d98b 100644 --- a/drivers/net/wireless/ipw2x00/ipw2100.c +++ b/drivers/net/wireless/ipw2x00/ipw2100.c @@ -174,7 +174,7 @@ that only one external action is invoked at a time. #define DRV_DESCRIPTION "Intel(R) PRO/Wireless 2100 Network Driver" #define DRV_COPYRIGHT "Copyright(c) 2003-2006 Intel Corporation" -struct pm_qos_request_list *ipw2100_pm_qos_req; +struct pm_qos_request_list ipw2100_pm_qos_req; /* Debugging stuff */ #ifdef CONFIG_IPW2100_DEBUG @@ -1741,7 +1741,7 @@ static int ipw2100_up(struct ipw2100_priv *priv, int deferred) /* the ipw2100 hardware really doesn't want power management delays * longer than 175usec */ - pm_qos_update_request(ipw2100_pm_qos_req, 175); + pm_qos_update_request(&ipw2100_pm_qos_req, 175); /* If the interrupt is enabled, turn it off... */ spin_lock_irqsave(&priv->low_lock, flags); @@ -1889,7 +1889,7 @@ static void ipw2100_down(struct ipw2100_priv *priv) ipw2100_disable_interrupts(priv); spin_unlock_irqrestore(&priv->low_lock, flags); - pm_qos_update_request(ipw2100_pm_qos_req, PM_QOS_DEFAULT_VALUE); + pm_qos_update_request(&ipw2100_pm_qos_req, PM_QOS_DEFAULT_VALUE); /* We have to signal any supplicant if we are disassociating */ if (associated) @@ -6669,8 +6669,8 @@ static int __init ipw2100_init(void) if (ret) goto out; - ipw2100_pm_qos_req = pm_qos_add_request(PM_QOS_CPU_DMA_LATENCY, - PM_QOS_DEFAULT_VALUE); + pm_qos_add_request(&ipw2100_pm_qos_req, PM_QOS_CPU_DMA_LATENCY, + PM_QOS_DEFAULT_VALUE); #ifdef CONFIG_IPW2100_DEBUG ipw2100_debug_level = debug; ret = driver_create_file(&ipw2100_pci_driver.driver, @@ -6692,7 +6692,7 @@ static void __exit ipw2100_exit(void) &driver_attr_debug_level); #endif pci_unregister_driver(&ipw2100_pci_driver); - pm_qos_remove_request(ipw2100_pm_qos_req); + pm_qos_remove_request(&ipw2100_pm_qos_req); } module_init(ipw2100_init); diff --git a/include/linux/netdevice.h b/include/linux/netdevice.h index 40291f3..393555a 100644 --- a/include/linux/netdevice.h +++ b/include/linux/netdevice.h @@ -779,7 +779,7 @@ struct net_device { */ char name[IFNAMSIZ]; - struct pm_qos_request_list *pm_qos_req; + struct pm_qos_request_list pm_qos_req; /* device name hash chain */ struct hlist_node name_hlist; diff --git a/include/linux/pm_qos_params.h b/include/linux/pm_qos_params.h index 8ba440e..d823cc0 100644 --- a/include/linux/pm_qos_params.h +++ b/include/linux/pm_qos_params.h @@ -1,8 +1,10 @@ +#ifndef _LINUX_PM_QOS_PARAMS_H +#define _LINUX_PM_QOS_PARAMS_H /* interface for the pm_qos_power infrastructure of the linux kernel. * * Mark Gross <mgross(a)linux.intel.com> */ -#include <linux/list.h> +#include <linux/plist.h> #include <linux/notifier.h> #include <linux/miscdevice.h> @@ -14,9 +16,12 @@ #define PM_QOS_NUM_CLASSES 4 #define PM_QOS_DEFAULT_VALUE -1 -struct pm_qos_request_list; +struct pm_qos_request_list { + struct plist_node list; + int pm_qos_class; +}; -struct pm_qos_request_list *pm_qos_add_request(int pm_qos_class, s32 value); +void pm_qos_add_request(struct pm_qos_request_list *l, int pm_qos_class, s32 value); void pm_qos_update_request(struct pm_qos_request_list *pm_qos_req, s32 new_value); void pm_qos_remove_request(struct pm_qos_request_list *pm_qos_req); @@ -25,3 +30,4 @@ int pm_qos_request(int pm_qos_class); int pm_qos_add_notifier(int pm_qos_class, struct notifier_block *notifier); int pm_qos_remove_notifier(int pm_qos_class, struct notifier_block *notifier); +#endif diff --git a/include/sound/pcm.h b/include/sound/pcm.h index dd76cde..6e3a297 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -366,7 +366,7 @@ struct snd_pcm_substream { int number; char name[32]; /* substream name */ int stream; /* stream (direction) */ - struct pm_qos_request_list *latency_pm_qos_req; /* pm_qos request */ + struct pm_qos_request_list latency_pm_qos_req; /* pm_qos request */ size_t buffer_bytes_max; /* limit ring buffer size */ struct snd_dma_buffer dma_buffer; unsigned int dma_buf_id; diff --git a/kernel/pm_qos_params.c b/kernel/pm_qos_params.c index 241fa79..f1d3d23 100644 --- a/kernel/pm_qos_params.c +++ b/kernel/pm_qos_params.c @@ -30,7 +30,6 @@ /*#define DEBUG*/ #include <linux/pm_qos_params.h> -#include <linux/plist.h> #include <linux/sched.h> #include <linux/spinlock.h> #include <linux/slab.h> @@ -49,11 +48,6 @@ * or pm_qos_object list and pm_qos_objects need to happen with pm_qos_lock * held, taken with _irqsave. One lock to rule them all */ -struct pm_qos_request_list { - struct plist_node list; - int pm_qos_class; -}; - enum pm_qos_type { PM_QOS_MAX, /* return the largest value */ PM_QOS_MIN, /* return the smallest value */ @@ -195,6 +189,11 @@ int pm_qos_request(int pm_qos_class) } EXPORT_SYMBOL_GPL(pm_qos_request); +static int pm_qos_request_active(struct pm_qos_request_list *req) +{ + return req->pm_qos_class != 0; +} + /** * pm_qos_add_request - inserts new qos request into the list * @pm_qos_class: identifies which list of qos request to us @@ -206,25 +205,22 @@ EXPORT_SYMBOL_GPL(pm_qos_request); * element as a handle for use in updating and removal. Call needs to save * this handle for later use. */ -struct pm_qos_request_list *pm_qos_add_request(int pm_qos_class, s32 value) +void pm_qos_add_request(struct pm_qos_request_list *dep, + int pm_qos_class, s32 value) { - struct pm_qos_request_list *dep; - - dep = kzalloc(sizeof(struct pm_qos_request_list), GFP_KERNEL); - if (dep) { - struct pm_qos_object *o = pm_qos_array[pm_qos_class]; - int new_value; - - if (value == PM_QOS_DEFAULT_VALUE) - new_value = o->default_value; - else - new_value = value; - plist_node_init(&dep->list, new_value); - dep->pm_qos_class = pm_qos_class; - update_target(o, &dep->list, 0); - } + struct pm_qos_object *o = pm_qos_array[pm_qos_class]; + int new_value; + + if (pm_qos_request_active(dep)) + return; - return dep; + if (value == PM_QOS_DEFAULT_VALUE) + new_value = o->default_value; + else + new_value = value; + plist_node_init(&dep->list, new_value); + dep->pm_qos_class = pm_qos_class; + update_target(o, &dep->list, 0); } EXPORT_SYMBOL_GPL(pm_qos_add_request); @@ -286,7 +282,7 @@ void pm_qos_remove_request(struct pm_qos_request_list *pm_qos_req) o = pm_qos_array[pm_qos_req->pm_qos_class]; update_target(o, &pm_qos_req->list, 1); - kfree(pm_qos_req); + memset(pm_qos_req, 0, sizeof(*pm_qos_req)); } EXPORT_SYMBOL_GPL(pm_qos_remove_request); @@ -334,8 +330,12 @@ static int pm_qos_power_open(struct inode *inode, struct file *filp) pm_qos_class = find_pm_qos_object_by_minor(iminor(inode)); if (pm_qos_class >= 0) { - filp->private_data = (void *) pm_qos_add_request(pm_qos_class, - PM_QOS_DEFAULT_VALUE); + struct pm_qos_request_list *req = kzalloc(GFP_KERNEL, sizeof(*req)); + if (!req) + return -ENOMEM; + + pm_qos_add_request(req, pm_qos_class, PM_QOS_DEFAULT_VALUE); + filp->private_data = req; if (filp->private_data) return 0; @@ -347,8 +347,9 @@ static int pm_qos_power_release(struct inode *inode, struct file *filp) { struct pm_qos_request_list *req; - req = (struct pm_qos_request_list *)filp->private_data; + req = filp->private_data; pm_qos_remove_request(req); + kfree(req); return 0; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 303ac04..d3b8b51 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -451,13 +451,10 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, snd_pcm_timer_resolution_change(substream); runtime->status->state = SNDRV_PCM_STATE_SETUP; - if (substream->latency_pm_qos_req) { - pm_qos_remove_request(substream->latency_pm_qos_req); - substream->latency_pm_qos_req = NULL; - } + pm_qos_remove_request(&substream->latency_pm_qos_req); if ((usecs = period_to_usecs(runtime)) >= 0) - substream->latency_pm_qos_req = pm_qos_add_request( - PM_QOS_CPU_DMA_LATENCY, usecs); + pm_qos_add_request(&substream->latency_pm_qos_req, + PM_QOS_CPU_DMA_LATENCY, usecs); return 0; _error: /* hardware might be unuseable from this time, @@ -512,8 +509,7 @@ static int snd_pcm_hw_free(struct snd_pcm_substream *substream) if (substream->ops->hw_free) result = substream->ops->hw_free(substream); runtime->status->state = SNDRV_PCM_STATE_OPEN; - pm_qos_remove_request(substream->latency_pm_qos_req); - substream->latency_pm_qos_req = NULL; + pm_qos_remove_request(&substream->latency_pm_qos_req); return result; }

2 5

[alsa-devel] [RFC PATCH v3 0/3] ep93xx i2s audio
by Ryan Mallon 08 Jun '10

08 Jun '10

The following patch series adds support for i2s audio on the ep93xx, with the Snapper CL15 used as an example implementation. This version fixes a number of things from the previous round: - Removed unused #defines - Set symmetric rates for the cpu dai - Fixed the sdiv/lrdiv calculation loop - Moved the cpu dai registration - Removed empty ep93xx_pcm_prepare callback - Fixed enable/disable when both capture and playback are running Capture still does not work. The dma is running, but the captured data is all zeros (tested used arecord and loopback). I have a 2.6.20 kernel with ep93xx i2s audio working an capture works there on the same hardware. I have spent sometime looking into this, but can't see the problem. Would appreciate if someone can lend some insight as to what I have done wrong. Ryan Mallon (3): ep93xx i2s core support ep93xx i2s audio driver Snapper cl15 audio support arch/arm/mach-ep93xx/clock.c | 69 ++++- arch/arm/mach-ep93xx/core.c | 31 ++ arch/arm/mach-ep93xx/include/mach/ep93xx-regs.h | 10 + arch/arm/mach-ep93xx/include/mach/platform.h | 1 + arch/arm/mach-ep93xx/snappercl15.c | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/ep93xx/Kconfig | 18 + sound/soc/ep93xx/Makefile | 11 + sound/soc/ep93xx/ep93xx-i2s.c | 487 +++++++++++++++++++++++ sound/soc/ep93xx/ep93xx-i2s.h | 18 + sound/soc/ep93xx/ep93xx-pcm.c | 319 +++++++++++++++ sound/soc/ep93xx/ep93xx-pcm.h | 22 + sound/soc/ep93xx/snappercl15.c | 143 +++++++ 14 files changed, 1131 insertions(+), 1 deletions(-) create mode 100644 sound/soc/ep93xx/Kconfig create mode 100644 sound/soc/ep93xx/Makefile create mode 100644 sound/soc/ep93xx/ep93xx-i2s.c create mode 100644 sound/soc/ep93xx/ep93xx-i2s.h create mode 100644 sound/soc/ep93xx/ep93xx-pcm.c create mode 100644 sound/soc/ep93xx/ep93xx-pcm.h create mode 100644 sound/soc/ep93xx/snappercl15.c

4 11

[alsa-devel] No mixers on ATI RS780 Azalia
by Jan Engelhardt 07 Jun '10

07 Jun '10

Hi, I have here a Samsung 320MXn2 monitor/mainboard (typical x86_64 environment) where I cannot get sound output to the internal speakers working. To start, there seem to be two sound cards: lspci/lspci -n: 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) 01:05.1 Audio device: ATI Technologies Inc RS780 Azalia controller 00:14.2 0403: 1002:4383 01:05.1 0403: 1002:960f /proc/asound/cards: 0 [SB ]: HDA-Intel - HDA ATI SB HDA ATI SB at 0xf0500000 irq 16 1 [HDMI]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xf0110000 irq 19 The soundcard responsible for the internal speaker is the 01:05.1/"HDMI" one. Under Windows XP, I see the following mixer elements for it: * Master Volume * Wave * SW Synth * CD Player Subsequently, there is sound (once I bump the volumes on these). However, in Linux up to including 2.6.34, I only see a single mixer element: * IEC958/S/PDIF mplayer etc. all successfully write to the device, but since there are not any further mixer controls, I can't unmute the essential channels. alsa-utils is version 1.0.21, from openSUSE 11.2 x86_64. Any help is appreciated. Thanks, Jan

5 15

[alsa-devel] ASoC updates for 2.6.36
by Mark Brown 07 Jun '10

07 Jun '10

The following changes since commit 1fab79b8a1bf0d46a7c7e6a1c06dbed1803f403c: Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2010-05-31 18:13:20 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 Eric Bénard (2): ASoC: imx-ssi.c: add new choices to platform configuration ASoC: imx: add eukrea-tlv320 Mark Brown (1): Merge branch 'for-2.6.35' into for-2.6.36 Ryan Mallon (1): ASoC: ep93xx i2s audio driver Sascha Hauer (2): ASoC: Add missing Kconfig entry for Phytec boards ASoC: MX31ads sound support should depend on MACH_MX31ADS_WM1133_EV1 Wan ZongShun (10): ASoC: nuc900: patch for SUBSTREAM_TYPE', 'PCM_TX' and 'PCM_RX' removal ASoC: nuc900: fix a wait loop bug ASoC: nuc900: fix a typo and rename the header file ASoC: nuc900: patch for modifing the ac97 delays to minimum ASoC: s6000: use resource_size for {request/release}_mem_region and ioremap ASoC: use resource_size for au1x ASoC: atmel: patch for the unnecessary variable removal ASoC: s3c: patch for the unnecessary variable 'state' removal ASoC: atmel: trivial code cleanup ASoC: patch for the useless 'break' removal in kirkwood arch/arm/plat-mxc/include/mach/ssi.h | 3 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/atmel-pcm.c | 1 - sound/soc/atmel/atmel_ssc_dai.c | 1 - sound/soc/au1x/psc-ac97.c | 13 +- sound/soc/au1x/psc-i2s.c | 13 +- sound/soc/au1x/psc.h | 1 - sound/soc/ep93xx/Kconfig | 9 + sound/soc/ep93xx/Makefile | 8 + sound/soc/ep93xx/ep93xx-i2s.c | 487 ++++++++++++++++++++ sound/soc/ep93xx/ep93xx-i2s.h | 18 + sound/soc/ep93xx/ep93xx-pcm.c | 319 +++++++++++++ sound/soc/ep93xx/ep93xx-pcm.h | 22 + sound/soc/imx/Kconfig | 20 +- sound/soc/imx/Makefile | 2 + sound/soc/imx/eukrea-tlv320.c | 135 ++++++ sound/soc/imx/imx-ssi.c | 11 +- sound/soc/kirkwood/kirkwood-i2s.c | 1 - sound/soc/nuc900/nuc900-ac97.c | 32 +- .../soc/nuc900/{nuc900-auido.h => nuc900-audio.h} | 4 - sound/soc/nuc900/nuc900-pcm.c | 18 +- sound/soc/s3c24xx/s3c-ac97.c | 1 - sound/soc/s6000/s6000-i2s.c | 38 +- 24 files changed, 1080 insertions(+), 79 deletions(-) create mode 100644 sound/soc/ep93xx/Kconfig create mode 100644 sound/soc/ep93xx/Makefile create mode 100644 sound/soc/ep93xx/ep93xx-i2s.c create mode 100644 sound/soc/ep93xx/ep93xx-i2s.h create mode 100644 sound/soc/ep93xx/ep93xx-pcm.c create mode 100644 sound/soc/ep93xx/ep93xx-pcm.h create mode 100644 sound/soc/imx/eukrea-tlv320.c rename sound/soc/nuc900/{nuc900-auido.h => nuc900-audio.h} (95%)

2 1

[alsa-devel] [GIT PULL] ASoC updates for 2.6.36
by Liam Girdwood 07 Jun '10

07 Jun '10

Hi Takashi, Some mainly OMAP McBSP specific patches to reduce power from Peter. We have Tony's Ack on all the arch/arm/omap stuff. Thanks Liam The following changes since commit 1fab79b8a1bf0d46a7c7e6a1c06dbed1803f403c: Takashi Iwai (1): Merge branch 'for-2.6.36' of git://git.kernel.org/.../broonie/sound-2.6 into topic/asoc are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.36 Peter Ujfalusi (6): OMAP: McBSP: Function to query the FIFO size OMAP3: McBSP: Change the way how the FIFO is handled OMAP3: McBSP: Use the port's buffer_size when calculating tx delay ASoC: omap-mcbsp: Save, and use wlen for threshold configuration ASoC: omap-mcbsp: Place correct constraints for streams ASoC: tlv320dac33: Add support for changing upper threshold arch/arm/mach-omap2/mcbsp.c | 10 ++-- arch/arm/plat-omap/include/plat/mcbsp.h | 2 + arch/arm/plat-omap/mcbsp.c | 51 ++++++++++----- sound/soc/codecs/tlv320dac33.c | 57 +++++++++++++--- sound/soc/omap/omap-mcbsp.c | 112 +++++++++++++++++++++++------- 5 files changed, 176 insertions(+), 56 deletions(-)

2 1

Re: [alsa-devel] [PATCH] ASoC/kirkwood: patch for the useless 'break' removal.
by Wan ZongShun 07 Jun '10

07 Jun '10

Hi Mark, How about this patch? My other patch seem to be applied, but this one still be hung. 在 2010年6月2日下午4:03，Wan ZongShun <mcuos.com(a)gmail.com> 写道： > This patch to remove the 'break;', when the 'switch' jumps to > the 'default' branch, the 'return -EINVAL' will be return with > a error number, so the 'break;' code never be run, it is unuseful > and should be removed here. > > > Signed-off-by: Wan ZongShun<mcuos.com(a)gmail.com> > --- > sound/soc/kirkwood/kirkwood-i2s.c | 1 - > 1 files changed, 0 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c > index 0adc597..0fdc7db 100644 > --- a/sound/soc/kirkwood/kirkwood-i2s.c > +++ b/sound/soc/kirkwood/kirkwood-i2s.c > @@ -296,7 +296,6 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, > > default: > return -EINVAL; > - break; > } > > return 0; > -- > 1.6.3.3 > -- *linux-arm-kernel mailing list mail addr:linux-arm-kernel@lists.infradead.org you can subscribe by: http://lists.infradead.org/mailman/listinfo/linux-arm-kernel * linux-arm-NUC900 mailing list mail addr:NUC900@googlegroups.com main web: https://groups.google.com/group/NUC900 you can subscribe it by sending me mail: mcuos.com(a)gmail.com

3 2

Re: [alsa-devel] [PATCH] Atmel/alsa: fix some code issues
by Wan ZongShun 07 Jun '10

07 Jun '10

在 2010年6月7日下午6:41，Nicolas Ferre <nicolas.ferre(a)atmel.com> 写道： > Le 28/05/2010 04:27, Wan ZongShun : >> Dear Nicolas, >> >> I am seeing the atmel codes as reference of my alsa driver programming. >> Finding a few code issues, I make a patch to fix, thanks! > > After a brief review, I come back to this > >> Signed-off-by: Wan ZongShun<mcuos.com(a)gmail.com> >> >> --- >> sound/soc/atmel/atmel-pcm.c | 2 +- >> sound/soc/atmel/atmel_ssc_dai.c | 3 +-- >> 2 files changed, 2 insertions(+), 3 deletions(-) >> >> diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c >> index f6b3cc0..c69a8fb 100644 >> --- a/sound/soc/atmel/atmel-pcm.c >> +++ b/sound/soc/atmel/atmel-pcm.c >> @@ -478,7 +478,7 @@ struct snd_soc_platform atmel_soc_platform = { >> .pcm_free = atmel_pcm_free_dma_buffers, >> .suspend = atmel_pcm_suspend, >> .resume = atmel_pcm_resume, >> -}; >> +} > > Not ok for this. It is a structure declaration: we should keep the ";". > Okay, thanks! >> EXPORT_SYMBOL_GPL(atmel_soc_platform); >> >> static int __init atmel_pcm_modinit(void) >> diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c >> index 0b59806..4c44d13 100644 >> --- a/sound/soc/atmel/atmel_ssc_dai.c >> +++ b/sound/soc/atmel/atmel_ssc_dai.c >> @@ -549,7 +549,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream >> *substream, >> printk(KERN_WARNING "atmel_ssc_dai: unsupported DAI format 0x%x\n", >> ssc_p->daifmt); >> return -EINVAL; >> - break; > > Ok. > >> } >> pr_debug("atmel_ssc_hw_params: " >> "RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", >> @@ -760,7 +759,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { >> .private_data = &ssc_info[2], >> }, >> #endif >> -}; >> +} > > Not ok for this either. ditto. > >> EXPORT_SYMBOL_GPL(atmel_ssc_dai); >> >> static int __init atmel_ssc_modinit(void) > > > -- > Nicolas Ferre > > -- *linux-arm-kernel mailing list mail addr:linux-arm-kernel@lists.infradead.org you can subscribe by: http://lists.infradead.org/mailman/listinfo/linux-arm-kernel * linux-arm-NUC900 mailing list mail addr:NUC900@googlegroups.com main web: https://groups.google.com/group/NUC900 you can subscribe it by sending me mail: mcuos.com(a)gmail.com

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Re: [alsa-devel] [PATCH v2 1/3]AsoC/nuc900: patch for SUBSTREAM_TYPE', 'PCM_TX' and 'PCM_RX' removal
by Liam Girdwood 07 Jun '10

07 Jun '10

On Wed, 2010-06-02 at 13:54 +0800, Wan ZongShun wrote: > This patch is to remove the 'SUBSTREAM_TYPE','PCM_TX' and 'PCM_RX' definition. > > There is no need to redefine SNDRV_PCM_STREAM_PLAYBACK as PCM_TX, > the SUBSTREAM_TYPE(substream) can be deleted too, the playback or record can be > judged by 'if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)' directly rather > than 'if (PCM_TX == stype)', which makes the codes easy to read. > > Signed-off-by: Wan ZongShun<mcuos.com(a)gmail.com> All Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

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Re: [alsa-devel] [PATCH] sound/spi: patch for the unuseful variable removal
by Liam Girdwood 07 Jun '10

07 Jun '10

On Fri, 2010-06-04 at 16:30 +0800, Wan ZongShun wrote: > The '*bitclk' of structure 'snd_at73c213' seems no use > ,so I make a patch to remove the unnecessary variable. > > Signed-off-by: Wan ZongShun <mcuos.com(a)gmail.com> > --- > sound/spi/at73c213.c | 1 - > 1 files changed, 0 insertions(+), 1 deletions(-) > > diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c > index 4c7b051..1bc56b2 100644 > --- a/sound/spi/at73c213.c > +++ b/sound/spi/at73c213.c > @@ -69,7 +69,6 @@ struct snd_at73c213 { > int irq; > int period; > unsigned long bitrate; > - struct clk *bitclk; > struct ssc_device *ssc; > struct spi_device *spi; > u8 spi_wbuffer[2]; Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

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[alsa-devel] [PATCH] ASoC: tlv320dac33: Add support for changing upper threshold
by Peter Ujfalusi 07 Jun '10

07 Jun '10

Upper threshold is used in mode7 of DAC33. Instead of hard wired UTHR, add control to change the upper threshold value. Changing upper threshold is not allowed when the playback is already running, since wrongly timed change in the UTHR can cause problems with the codec. With this control the length of the burst in mode7 can be changed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/tlv320dac33.c | 57 +++++++++++++++++++++++++++++++++------ 1 files changed, 48 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 65adc77..2fa946c 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -120,6 +120,8 @@ struct tlv320dac33_priv { * samples */ unsigned int mode7_us_to_lthr; /* Time to reach lthr from uthr */ + unsigned int uthr; + enum dac33_state state; }; @@ -442,6 +444,39 @@ static int dac33_set_nsample(struct snd_kcontrol *kcontrol, return ret; } +static int dac33_get_uthr(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = dac33->uthr; + + return 0; +} + +static int dac33_set_uthr(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + if (dac33->substream) + return -EBUSY; + + if (dac33->uthr == ucontrol->value.integer.value[0]) + return 0; + + if (ucontrol->value.integer.value[0] < (MODE7_LTHR + 10) || + ucontrol->value.integer.value[0] > MODE7_UTHR) + ret = -EINVAL; + else + dac33->uthr = ucontrol->value.integer.value[0]; + + return ret; +} + static int dac33_get_fifo_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -506,6 +541,8 @@ static const struct snd_kcontrol_new dac33_snd_controls[] = { static const struct snd_kcontrol_new dac33_nsample_snd_controls[] = { SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0, dac33_get_nsample, dac33_set_nsample), + SOC_SINGLE_EXT("UTHR", 0, 0, MODE7_UTHR, 0, + dac33_get_uthr, dac33_set_uthr), SOC_ENUM_EXT("FIFO Mode", dac33_fifo_mode_enum, dac33_get_fifo_mode, dac33_set_fifo_mode), }; @@ -985,7 +1022,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) * Configure the threshold levels, and leave 10 sample space * at the bottom, and also at the top of the FIFO */ - dac33_write16(codec, DAC33_UTHR_MSB, DAC33_THRREG(MODE7_UTHR)); + dac33_write16(codec, DAC33_UTHR_MSB, DAC33_THRREG(dac33->uthr)); dac33_write16(codec, DAC33_LTHR_MSB, DAC33_THRREG(MODE7_LTHR)); break; default: @@ -1052,8 +1089,8 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream) break; case DAC33_FIFO_MODE7: dac33->mode7_us_to_lthr = - SAMPLES_TO_US(substream->runtime->rate, - MODE7_UTHR - MODE7_LTHR + 1); + SAMPLES_TO_US(substream->runtime->rate, + dac33->uthr - MODE7_LTHR + 1); dac33->t_stamp1 = 0; break; default: @@ -1104,7 +1141,7 @@ static snd_pcm_sframes_t dac33_dai_delay( struct snd_soc_codec *codec = socdev->card->codec; struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); unsigned long long t0, t1, t_now; - unsigned int time_delta; + unsigned int time_delta, uthr; int samples_out, samples_in, samples; snd_pcm_sframes_t delay = 0; @@ -1182,6 +1219,7 @@ static snd_pcm_sframes_t dac33_dai_delay( case DAC33_FIFO_MODE7: spin_lock(&dac33->lock); t0 = dac33->t_stamp1; + uthr = dac33->uthr; spin_unlock(&dac33->lock); t_now = ktime_to_us(ktime_get()); @@ -1194,7 +1232,7 @@ static snd_pcm_sframes_t dac33_dai_delay( * Either the timestamps are messed or equal. Report * maximum delay */ - delay = MODE7_UTHR; + delay = uthr; goto out; } @@ -1208,8 +1246,8 @@ static snd_pcm_sframes_t dac33_dai_delay( substream->runtime->rate, time_delta); - if (likely(MODE7_UTHR > samples_out)) - delay = MODE7_UTHR - samples_out; + if (likely(uthr > samples_out)) + delay = uthr - samples_out; else delay = 0; } else { @@ -1227,8 +1265,8 @@ static snd_pcm_sframes_t dac33_dai_delay( time_delta); delay = MODE7_LTHR + samples_in - samples_out; - if (unlikely(delay > MODE7_UTHR)) - delay = MODE7_UTHR; + if (unlikely(delay > uthr)) + delay = uthr; } break; default: @@ -1484,6 +1522,7 @@ static int __devinit dac33_i2c_probe(struct i2c_client *client, dac33->irq = client->irq; dac33->nsample = NSAMPLE_MAX; dac33->nsample_max = NSAMPLE_MAX; + dac33->uthr = MODE7_UTHR; /* Disable FIFO use by default */ dac33->fifo_mode = DAC33_FIFO_BYPASS; -- 1.7.1

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[alsa-devel] 2.6.34-rc: sound hungs (ice1724)
by Alexander Beregalov 06 Jun '10

06 Jun '10

Hi Takashi Could you please help me how to debug an issue. Sometimes soung hungs, it happens few days (up to a week) after boot. I have no idea how to reproduce it or how to debug it. That's why it is very hard to bisect, I have not tried, the better way should exist. The problem appeared somewhere after 2.6.33. Card is M-Audio Revo 5.1 (1412:1724). After it hungs ALSA continues to play last sound frame forever. When I try to play something else playing frame changes. Only reboot heals it. rmmod hungs at INFO: task rmmod:30356 blocked for more than 120 seconds. "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. rmmod D db281e7c 0 30356 2345 0x00000004 db281e8c 00000086 00000002 db281e7c cc2d7b90 00000000 c14e8b80 09c3638a c10252a0 00000001 c14e2000 c14e2000 c14e8b80 c14e8b80 cc2d7a00 db281e8c 00000246 db281e8c c1043d23 00000001 f6923000 f69231b4 db281e9c db281ebc Call Trace: [<c10252a0>] ? __wake_up+0x40/0x50 [<c1043d23>] ? prepare_to_wait+0x43/0x70 [<f8ce328d>] snd_card_free+0x5d/0x90 [snd] [<c1043af0>] ? autoremove_wake_function+0x0/0x40 [<f9ebcaa6>] snd_vt1724_remove+0x13/0x25 [snd_ice1724] [<c11cddb9>] pci_device_remove+0x19/0x40 [<c1239f3e>] __device_release_driver+0x3e/0x90 [<c123a017>] driver_detach+0x87/0x90 [<c1239323>] bus_remove_driver+0x63/0xa0 [<c123a526>] driver_unregister+0x46/0x80 [<c10d616d>] ? sysfs_remove_file+0xd/0x10 [<c11cdfed>] pci_unregister_driver+0x2d/0x80 [<f9ebca91>] alsa_card_ice1724_exit+0xd/0xf [snd_ice1724] [<c1057bee>] sys_delete_module+0x15e/0x210 [<c107f91f>] ? remove_vma+0x3f/0x60 [<c1080902>] ? do_munmap+0x202/0x2d0 [<c1002b50>] sysenter_do_call+0x12/0x26 CONFIG_SOUND=y CONFIG_SND=m CONFIG_SND_TIMER=m CONFIG_SND_PCM=m CONFIG_SND_RAWMIDI=m CONFIG_SND_VMASTER=y CONFIG_SND_DMA_SGBUF=y CONFIG_SND_AC97_CODEC=m CONFIG_SND_PCI=y CONFIG_SND_ICE1724=m

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[alsa-devel] multi plugin and start_threshold = -1
by Raymond Yau 06 Jun '10

06 Jun '10

I: core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 5. I: alsa-sink.c: Starting playback. I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_START failed (-77) I: (alsa-lib)pcm_hw.c: PCM state = RUNNING any limitation to use -1 as start_thershold ? why SNDRV_PCM_IOCTL_START failed (-77) only occur with PA server when using "multi" plugin of surround40 of ca0106 ? other application (e.g. aplay , speaker-test and mplayer seem work fine https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5018

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[alsa-devel] alsa-info.sh
by Justin P. Mattock 04 Jun '10

04 Jun '10

hello.. Im looking for the codec#0 for the MacBookAir2,1 can anybody supply this?(doing a google search gives nothing but crudd). cheers, Justin P. Mattock

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[alsa-devel] ASoC: Defining "signed integer" mixer controls?
by Stuart Longland 04 Jun '10

04 Jun '10

Hi all... I'm posting this via GMane since my server at home seems to be down, hopefully it gets through. I hope to put a version of the TLV320AIC3204 driver that I've been working on up there -- I did try posting it to the list, but I suspect it's either been left in the moderation queue or silently dropped. In short, I got the sound working; just testing playback for now... Recording should work, once I get the mixer set up. My query though; the mixer on this CODEC provides gain settings, helpfully scaled in dB, as signed integers. For instance; the four line level output drivers: Headphone left/right,and Line Out left/right, can all be adjusted in gain from -6dB through to +29dB. These are set in individual registers as the lower 6 bits; so: -6 dB is represented as 111010 (0x3a) 29 dB is represented as 011101 (0x1d) At the moment, things "half work" with the following code: /* * DAC digital volumes. From -63.5 to +24 dB in 0.5 dB steps */ static DECLARE_TLV_DB_MINMAX(dac_tlv, -6350, 2400); /* ADC PGA gain volumes. From 0 to 59.5 dB in 0.5 dB steps */ static DECLARE_TLV_DB_MINMAX(adc_tlv, 0, 0); /* * Output driver stage volumes; -6dB through to +29dB */ static DECLARE_TLV_DB_MINMAX(output_stage_tlv, -600, 2900); static const struct snd_kcontrol_new aic3204_snd_controls[] = { /* Output */ SOC_DOUBLE_R_TLV("Headphone Output Volume", AIC3204_HPLGAIN, AIC3204_HPRGAIN, 0, 29, 0, output_stage_tlv), SOC_DOUBLE_R_TLV("Line Output Volume", AIC3204_LOLGAIN, AIC3204_LORGAIN, 0, 29, 0, output_stage_tlv), SOC_DOUBLE_R_TLV("PCM Playback Volume", AIC3204_LDACVOL, AIC3204_RDACVOL, 0, 127, 0, dac_tlv), ... However; it treats the bitfield as unsigned, which is wrong in this case. The bitfield is a signed integer in two's complement format. How do I convince ALSA that these integers have a sign bit? Regards, Stuart Longland

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[alsa-devel] [PATCH] alsa-lib: snd_pcm_delay and friends do not account for a write being currently in progress
by John Lindgren 04 Jun '10

04 Jun '10

Hi, In a multi-threaded application it is possible for snd_pcm_delay or an equivalent function to be called by one thread while another is sitting in snd_pcm_writei. In this case, snd_pcm_delay does not take into account that there may not be enough space for all the data passed to snd_pcm_writei to be written to the ring buffer at once, and will return incorrect values. To illustrate: Suppose a sound card has a maximum buffer size of 0.1 second. Then, suppose an application writes 0.5 second of audio. snd_pcm_writei will immediately write the first 0.1 second to the ring buffer, then will incrementally write the remaining 0.4 second as the buffer drains. During this time, snd_pcm_delay will return a constant fill level of 0.1 second. On the application side, the playback time counter will be calculated during this time as 0.5 seconds written to ALSA minus a delay of 0.1 second, giving a constant stream position of 0.4 second. (This problem showed up when a recent change in Audacious caused it to call snd_pcm_writei with larger chunks of audio than formerly -- the result was that the visualization, which is synchronized with the time counter, began to jump, where it should be a smooth 30 FPS animation.) The problem is a little tough to fix, since there must be some thread semantics involved so that the thread calling snd_pcm_delay knows at what instant snd_pcm_delay will begin taking into account the data passed to snd_pcm_writei. My solution is to create a variant of snd_pcm_writei ("snd_pcm_writei_locked"), which is the passed the additional argument of a pointer to a mutex locked in advance by the caller. snd_pcm_writei_locked will then unlock the mutex once the application can safely assume that snd_pcm_delay will take into account the new data. I realize this is a substantial change, so please tell me whether this patch is along the right track or if a different solution is called for. Signed-off-by: John Lindgren <john.lindgren(a)tds.net> --- include/pcm.h | 8 +++++ src/pcm/pcm.c | 76 ++++++++++++++++++++++++++++++++++++++++++++++++--- src/pcm/pcm_hw.c | 39 +++++++++++++++++++++++++- src/pcm/pcm_local.h | 3 ++ 4 files changed, 121 insertions(+), 5 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index f3618c3..2234279 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -33,6 +33,10 @@ extern "C" { #endif +#include <pthread.h> + +#define SND_PCM_HAVE_LOCKED_WRITE + /** * \defgroup PCM PCM Interface * See the \ref pcm page for more details. @@ -448,8 +452,12 @@ snd_pcm_sframes_t snd_pcm_rewind(snd_pcm_t *pcm, snd_pcm_uframes_t frames); snd_pcm_sframes_t snd_pcm_forwardable(snd_pcm_t *pcm); snd_pcm_sframes_t snd_pcm_forward(snd_pcm_t *pcm, snd_pcm_uframes_t frames); snd_pcm_sframes_t snd_pcm_writei(snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size); +snd_pcm_sframes_t snd_pcm_writei_locked (snd_pcm_t * pcm, const void * buffer, + snd_pcm_uframes_t size, pthread_mutex_t * mutex); snd_pcm_sframes_t snd_pcm_readi(snd_pcm_t *pcm, void *buffer, snd_pcm_uframes_t size); snd_pcm_sframes_t snd_pcm_writen(snd_pcm_t *pcm, void **bufs, snd_pcm_uframes_t size); +snd_pcm_sframes_t snd_pcm_writen_locked (snd_pcm_t * pcm, void * * bufs, + snd_pcm_uframes_t size, pthread_mutex_t * mutex); snd_pcm_sframes_t snd_pcm_readn(snd_pcm_t *pcm, void **bufs, snd_pcm_uframes_t size); int snd_pcm_wait(snd_pcm_t *pcm, int timeout); diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index f910189..db03067 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -903,8 +903,12 @@ int snd_pcm_sw_params(snd_pcm_t *pcm, snd_pcm_sw_params_t *params) */ int snd_pcm_status(snd_pcm_t *pcm, snd_pcm_status_t *status) { - assert(pcm && status); - return pcm->fast_ops->status(pcm->fast_op_arg, status); + int result; + + assert(pcm && status); + result = pcm->fast_ops->status (pcm->fast_op_arg, status); + status->delay += pcm->fast_op_arg->write_delay; + return result; } /** @@ -973,12 +977,17 @@ link_warning(snd_pcm_hwsync, "Warning: snd_pcm_hwsync() is deprecated, consider */ int snd_pcm_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) { + int result; + assert(pcm); if (CHECK_SANITY(! pcm->setup)) { SNDMSG("PCM not set up"); return -EIO; } - return pcm->fast_ops->delay(pcm->fast_op_arg, delayp); + + result = pcm->fast_ops->delay (pcm->fast_op_arg, delayp); + * delayp += pcm->fast_op_arg->write_delay; + return result; } /** @@ -1248,6 +1257,34 @@ snd_pcm_sframes_t snd_pcm_writei(snd_pcm_t *pcm, const void *buffer, snd_pcm_ufr } /** + * \brief Variant of snd_pcm_writei for multithreaded applications + * \param mutex Mutex already locked by caller + * + * In a multithreaded application, there is uncertainty inherent in calling + * snd_pcm_delay while another thread is calling and_pcm_writei, since there is + * no way to know whether the data passed to snd_pcm_writei has yet been + * accounted for in the delay calculation. This function provides a solution to + * that problem by unlocking the mutex when it is safe to call snd_pcm_delay. + * The mutex is locked again before the function returns. An application using + * this function should be careful to avoid deadlocks. + * + * Programmers should check that the macro SND_PCM_HAVE_LOCKED_WRITE is defined + * before using this function. + */ + +snd_pcm_sframes_t snd_pcm_writei_locked (snd_pcm_t * pcm, const void * buffer, + snd_pcm_uframes_t size, pthread_mutex_t * mutex) +{ + snd_pcm_sframes_t result; + + assert (pcm && pcm->fast_op_arg && ! pcm->fast_op_arg->write_mutex); + pcm->fast_op_arg->write_mutex = mutex; + result = snd_pcm_writei (pcm, buffer, size); + pcm->fast_op_arg->write_mutex = NULL; + return result; +} + +/** * \brief Write non interleaved frames to a PCM * \param pcm PCM handle * \param bufs frames containing buffers (one for each channel) @@ -1280,6 +1317,25 @@ snd_pcm_sframes_t snd_pcm_writen(snd_pcm_t *pcm, void **bufs, snd_pcm_uframes_t } /** + * \brief Variant of snd_pcm_writen for multithreaded applications + * \param Mutex already locked by caller + * + * (See snd_pcm_writei_locked for a full description.) + */ + +snd_pcm_sframes_t snd_pcm_writen_locked (snd_pcm_t * pcm, void * * bufs, + snd_pcm_uframes_t size, pthread_mutex_t * mutex) +{ + snd_pcm_sframes_t result; + + assert (pcm && pcm->fast_op_arg && ! pcm->fast_op_arg->write_mutex); + pcm->fast_op_arg->write_mutex = mutex; + result = snd_pcm_writen (pcm, bufs, size); + pcm->fast_op_arg->write_mutex = NULL; + return result; +} + +/** * \brief Read interleaved frames from a PCM * \param pcm PCM handle * \param buffer frames containing buffer @@ -2480,6 +2536,7 @@ int snd_pcm_avail_delay(snd_pcm_t *pcm, sf = pcm->fast_ops->delay(pcm->fast_op_arg, delayp); if (sf < 0) return (int)sf; + * delayp += pcm->fast_op_arg->write_delay; sf = pcm->fast_ops->avail_update(pcm->fast_op_arg); if (sf < 0) return (int)sf; @@ -6652,7 +6709,18 @@ snd_pcm_sframes_t snd_pcm_write_areas(snd_pcm_t *pcm, const snd_pcm_channel_area goto _end; } - err = snd_pcm_wait(pcm, -1); + if (pcm->write_mutex) { + pcm->write_delay = size; + assert (! pthread_mutex_unlock (pcm->write_mutex)); + } + + err = snd_pcm_wait(pcm, -1); + + if (pcm->write_mutex) { + assert (! pthread_mutex_lock (pcm->write_mutex)); + pcm->write_delay = 0; + } + if (err < 0) break; goto _again; diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 9d243d5..6a428d8 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -753,7 +753,8 @@ static int snd_pcm_hw_unlink(snd_pcm_t *pcm) return 0; } -static snd_pcm_sframes_t snd_pcm_hw_writei(snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size) +static snd_pcm_sframes_t writei_real (snd_pcm_t * pcm, const void * buffer, + snd_pcm_uframes_t size) { int err; snd_pcm_hw_t *hw = pcm->private_data; @@ -772,6 +773,42 @@ static snd_pcm_sframes_t snd_pcm_hw_writei(snd_pcm_t *pcm, const void *buffer, s return xferi.result; } +static snd_pcm_sframes_t snd_pcm_hw_writei (snd_pcm_t * pcm, const void * + buffer, snd_pcm_uframes_t size) +{ + snd_pcm_sframes_t remain = size; + snd_pcm_sframes_t result; + + while (1) { + if ((result = snd_pcm_hw_avail_update (pcm)) < 0) + return result; + if ((result = writei_real (pcm, buffer, result < remain ? result : + remain)) < 0) + return result; + + buffer = (const char *) buffer + snd_pcm_frames_to_bytes (pcm, result); + remain -= result; + + if (! remain) + return size; + + if (pcm->write_mutex) { + pcm->write_delay = remain; + assert (! pthread_mutex_unlock (pcm->write_mutex)); + } + + result = snd_pcm_wait (pcm, -1); + + if (pcm->write_mutex) { + assert (! pthread_mutex_lock (pcm->write_mutex)); + pcm->write_delay = 0; + } + + if (result < 0) + return result; + } +} + static snd_pcm_sframes_t snd_pcm_hw_writen(snd_pcm_t *pcm, void **bufs, snd_pcm_uframes_t size) { int err; diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index 9aa81e1..53eef78 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -229,6 +229,9 @@ struct _snd_pcm { snd_pcm_t *fast_op_arg; void *private_data; struct list_head async_handlers; + snd_pcm_uframes_t write_delay; /* size of data passed to snd_pcm_write* + * but not yet outputted to ring buffer */ + pthread_mutex_t * write_mutex; /* mutex passed to snd_pcm_write*_locked */ }; /* make local functions really local */

4 13

[alsa-devel] [RFC][PATCH 20/26] alsa: ASoC: Add JZ4740 codec driver
by Lars-Peter Clausen 04 Jun '10

04 Jun '10

This patch adds support for the JZ4740 internal codec. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/jz4740-codec.c | 502 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/jz4740-codec.h | 20 ++ 4 files changed, 528 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/jz4740-codec.c create mode 100644 sound/soc/codecs/jz4740-codec.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 31ac553..b8008df 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -23,6 +23,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4671 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS4270 if I2C + select SND_SOC_JZ4740 if SOC_JZ4740 select SND_SOC_MAX9877 if I2C select SND_SOC_DA7210 if I2C select SND_SOC_PCM3008 @@ -269,6 +270,9 @@ config SND_SOC_WM9712 config SND_SOC_WM9713 tristate +config SND_SOC_JZ4740_CODEC + tristate + # Amp config SND_SOC_MAX9877 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 91429ea..4c7ee31 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -56,6 +56,7 @@ snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o snd-soc-wm-hubs-objs := wm_hubs.o +snd-soc-jz4740-codec-objs := jz4740-codec.o # Amp snd-soc-max9877-objs := max9877.o @@ -121,6 +122,7 @@ obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o +obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o # Amp obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o diff --git a/sound/soc/codecs/jz4740-codec.c b/sound/soc/codecs/jz4740-codec.c new file mode 100644 index 0000000..6e4b741 --- /dev/null +++ b/sound/soc/codecs/jz4740-codec.c @@ -0,0 +1,502 @@ +/* + * Copyright (C) 2009-2010, Lars-Peter Clausen <lars(a)metafoo.de> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> + +#include <linux/delay.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc-dapm.h> +#include <sound/soc.h> + +#define JZ4740_REG_CODEC_1 0x0 +#define JZ4740_REG_CODEC_2 0x1 + +#define JZ4740_CODEC_1_LINE_ENABLE BIT(29) +#define JZ4740_CODEC_1_MIC_ENABLE BIT(28) +#define JZ4740_CODEC_1_SW1_ENABLE BIT(27) +#define JZ4740_CODEC_1_ADC_ENABLE BIT(26) +#define JZ4740_CODEC_1_SW2_ENABLE BIT(25) +#define JZ4740_CODEC_1_DAC_ENABLE BIT(24) +#define JZ4740_CODEC_1_VREF_DISABLE BIT(20) +#define JZ4740_CODEC_1_VREF_AMP_DISABLE BIT(19) +#define JZ4740_CODEC_1_VREF_PULL_DOWN BIT(18) +#define JZ4740_CODEC_1_VREF_LOW_CURRENT BIT(17) +#define JZ4740_CODEC_1_VREF_HIGH_CURRENT BIT(16) +#define JZ4740_CODEC_1_HEADPHONE_DISABLE BIT(14) +#define JZ4740_CODEC_1_HEADPHONE_AMP_CHANGE_ANY BIT(13) +#define JZ4740_CODEC_1_HEADPHONE_CHANGE BIT(12) +#define JZ4740_CODEC_1_HEADPHONE_PULL_DOWN_M BIT(11) +#define JZ4740_CODEC_1_HEADPHONE_PULL_DOWN_R BIT(10) +#define JZ4740_CODEC_1_HEADPHONE_POWER_DOWN_M BIT(9) +#define JZ4740_CODEC_1_HEADPHONE_POWER_DOWN BIT(8) +#define JZ4740_CODEC_1_SUSPEND BIT(1) +#define JZ4740_CODEC_1_RESET BIT(0) + +#define JZ4740_CODEC_1_LINE_ENABLE_OFFSET 29 +#define JZ4740_CODEC_1_MIC_ENABLE_OFFSET 28 +#define JZ4740_CODEC_1_SW1_ENABLE_OFFSET 27 +#define JZ4740_CODEC_1_ADC_ENABLE_OFFSET 26 +#define JZ4740_CODEC_1_SW2_ENABLE_OFFSET 25 +#define JZ4740_CODEC_1_DAC_ENABLE_OFFSET 24 +#define JZ4740_CODEC_1_HEADPHONE_DISABLE_OFFSET 14 +#define JZ4740_CODEC_1_HEADPHONE_POWER_DOWN_OFFSET 8 + +#define JZ4740_CODEC_2_INPUT_VOLUME_MASK 0x1f0000 +#define JZ4740_CODEC_2_SAMPLE_RATE_MASK 0x000f00 +#define JZ4740_CODEC_2_MIC_BOOST_GAIN_MASK 0x000030 +#define JZ4740_CODEC_2_HEADPHONE_VOLUME_MASK 0x000003 + +#define JZ4740_CODEC_2_INPUT_VOLUME_OFFSET 16 +#define JZ4740_CODEC_2_SAMPLE_RATE_OFFSET 8 +#define JZ4740_CODEC_2_MIC_BOOST_GAIN_OFFSET 4 +#define JZ4740_CODEC_2_HEADPHONE_VOLUME_OFFSET 0 + +struct jz4740_codec { + void __iomem *base; + struct resource *mem; + + uint32_t reg_cache[2]; + struct snd_soc_codec codec; +}; + +static inline struct jz4740_codec *codec_to_jz4740(struct snd_soc_codec *codec) +{ + return container_of(codec, struct jz4740_codec, codec); +} + +static unsigned int jz4740_codec_read(struct snd_soc_codec *codec, unsigned int reg) +{ + struct jz4740_codec *jz4740_codec = codec_to_jz4740(codec); + return readl(jz4740_codec->base + (reg << 2)); +} + +static int jz4740_codec_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + struct jz4740_codec *jz4740_codec = codec_to_jz4740(codec); + jz4740_codec->reg_cache[reg] = val; + + writel(val, jz4740_codec->base + (reg << 2)); + return 0; +} + +static const struct snd_kcontrol_new jz4740_codec_controls[] = { + SOC_SINGLE("Master Playback Volume", JZ4740_REG_CODEC_2, + JZ4740_CODEC_2_HEADPHONE_VOLUME_OFFSET, 3, 0), + SOC_SINGLE("Capture Volume", JZ4740_REG_CODEC_2, + JZ4740_CODEC_2_INPUT_VOLUME_OFFSET, 31, 0), + SOC_SINGLE("Master Playback Switch", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_HEADPHONE_DISABLE_OFFSET, 1, 1), + SOC_SINGLE("Mic Capture Volume", JZ4740_REG_CODEC_2, + JZ4740_CODEC_2_MIC_BOOST_GAIN_OFFSET, 3, 0), +}; + +static const struct snd_kcontrol_new jz4740_codec_output_controls[] = { + SOC_DAPM_SINGLE("Bypass Switch", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_SW1_ENABLE_OFFSET, 1, 0), + SOC_DAPM_SINGLE("DAC Switch", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_SW2_ENABLE_OFFSET, 1, 0), +}; + +static const struct snd_kcontrol_new jz4740_codec_input_controls[] = { + SOC_DAPM_SINGLE("Line Capture Switch", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_LINE_ENABLE_OFFSET, 1, 0), + SOC_DAPM_SINGLE("Mic Capture Switch", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_MIC_ENABLE_OFFSET, 1, 0), +}; + +static const struct snd_soc_dapm_widget jz4740_codec_dapm_widgets[] = { + SND_SOC_DAPM_ADC("ADC", "Capture", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_ADC_ENABLE_OFFSET, 0), + SND_SOC_DAPM_DAC("DAC", "Playback", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_DAC_ENABLE_OFFSET, 0), + + SND_SOC_DAPM_MIXER("Output Mixer", JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_HEADPHONE_POWER_DOWN_OFFSET, 1, + jz4740_codec_output_controls, + ARRAY_SIZE(jz4740_codec_output_controls)), + + SND_SOC_DAPM_MIXER_NAMED_CTL("Input Mixer", SND_SOC_NOPM, 0, 0, + jz4740_codec_input_controls, + ARRAY_SIZE(jz4740_codec_input_controls)), + SND_SOC_DAPM_MIXER("Line Input", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), + + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_INPUT("LIN"), + SND_SOC_DAPM_INPUT("RIN"), +}; + +static const struct snd_soc_dapm_route jz4740_codec_dapm_routes[] = { + + {"Line Input", NULL, "LIN"}, + {"Line Input", NULL, "RIN"}, + + {"Input Mixer", "Line Capture Switch", "Line Input"}, + {"Input Mixer", "Mic Capture Switch", "MIC"}, + + {"ADC", NULL, "Input Mixer"}, + + {"Output Mixer", "Bypass Switch", "Input Mixer"}, + {"Output Mixer", "DAC Switch", "DAC"}, + + {"LOUT", NULL, "Output Mixer"}, + {"ROUT", NULL, "Output Mixer"}, +}; + +static int jz4740_codec_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + uint32_t val; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S18_3LE: + break; + default: + return -EINVAL; + break; + } + + switch (params_rate(params)) { + case 8000: + val = 0; + break; + case 11025: + val = 1; + break; + case 12000: + val = 2; + break; + case 16000: + val = 3; + break; + case 22050: + val = 4; + break; + case 24000: + val = 5; + break; + case 32000: + val = 6; + break; + case 44100: + val = 7; + break; + case 48000: + val = 8; + break; + default: + return -EINVAL; + } + + val <<= JZ4740_CODEC_2_SAMPLE_RATE_OFFSET; + + snd_soc_update_bits(codec, JZ4740_REG_CODEC_2, + JZ4740_CODEC_2_SAMPLE_RATE_MASK, val); + + return 0; +} + +static int jz4740_codec_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + return 0; +} + +static struct snd_soc_dai_ops jz4740_codec_dai_ops = { + .hw_params = jz4740_codec_hw_params, + .set_fmt = jz4740_codec_set_fmt, +}; + +struct snd_soc_dai jz4740_codec_dai = { + .name = "jz-codec", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8, + }, + .ops = &jz4740_codec_dai_ops, + .symmetric_rates = 1, +}; +EXPORT_SYMBOL_GPL(jz4740_codec_dai); + +static int jz4740_codec_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + + if (codec->bias_level == SND_SOC_BIAS_OFF && level != SND_SOC_BIAS_OFF) { + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_RESET, JZ4740_CODEC_1_RESET); + udelay(2); + + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_SUSPEND | JZ4740_CODEC_1_RESET, 0); + } + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_VREF_DISABLE | + JZ4740_CODEC_1_VREF_AMP_DISABLE | + JZ4740_CODEC_1_HEADPHONE_POWER_DOWN_M | + JZ4740_CODEC_1_VREF_LOW_CURRENT | + JZ4740_CODEC_1_VREF_HIGH_CURRENT, + 0); + break; + case SND_SOC_BIAS_PREPARE: + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_VREF_LOW_CURRENT | + JZ4740_CODEC_1_VREF_HIGH_CURRENT, + JZ4740_CODEC_1_VREF_LOW_CURRENT | + JZ4740_CODEC_1_VREF_HIGH_CURRENT); + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_VREF_DISABLE | JZ4740_CODEC_1_VREF_AMP_DISABLE, + JZ4740_CODEC_1_VREF_DISABLE | JZ4740_CODEC_1_VREF_AMP_DISABLE); + break; + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_SUSPEND, JZ4740_CODEC_1_SUSPEND); + break; + } + codec->bias_level = level; + + return 0; +} + + +static struct snd_soc_codec *jz4740_codec_codec; + +static int jz4740_codec_dev_probe(struct platform_device *pdev) +{ + int ret; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = jz4740_codec_codec; + + BUG_ON(!codec); + + socdev->card->codec = codec; + + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret) { + dev_err(&pdev->dev, "Failed to create pcms: %d\n", ret); + return ret; + } + + snd_soc_add_controls(codec, jz4740_codec_controls, + ARRAY_SIZE(jz4740_codec_controls)); + + snd_soc_dapm_new_controls(codec, jz4740_codec_dapm_widgets, + ARRAY_SIZE(jz4740_codec_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, jz4740_codec_dapm_routes, + ARRAY_SIZE(jz4740_codec_dapm_routes)); + + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int jz4740_codec_dev_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_jz4740_codec = { + .probe = jz4740_codec_dev_probe, + .remove = jz4740_codec_dev_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_jz4740_codec); + +static int __devinit jz4740_codec_probe(struct platform_device *pdev) +{ + int ret; + struct jz4740_codec *jz4740_codec; + struct snd_soc_codec *codec; + + jz4740_codec = kzalloc(sizeof(*jz4740_codec), GFP_KERNEL); + + if (!jz4740_codec) + return -ENOMEM; + + jz4740_codec->mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + + if (!jz4740_codec->mem) { + dev_err(&pdev->dev, "Failed to get mmio memory resource\n"); + ret = -ENOENT; + goto err_free_codec; + } + + jz4740_codec->mem = request_mem_region(jz4740_codec->mem->start, + resource_size(jz4740_codec->mem), pdev->name); + + if (!jz4740_codec->mem) { + dev_err(&pdev->dev, "Failed to request mmio memory region\n"); + ret = -EBUSY; + goto err_free_codec; + } + + jz4740_codec->base = ioremap(jz4740_codec->mem->start, resource_size(jz4740_codec->mem)); + + if (!jz4740_codec->base) { + dev_err(&pdev->dev, "Failed to ioremap mmio memory\n"); + ret = -EBUSY; + goto err_release_mem_region; + } + + jz4740_codec_dai.dev = &pdev->dev; + + codec = &jz4740_codec->codec; + + codec->dev = &pdev->dev; + codec->name = "jz-codec"; + codec->owner = THIS_MODULE; + + codec->read = jz4740_codec_read; + codec->write = jz4740_codec_write; + codec->set_bias_level = jz4740_codec_set_bias_level; + codec->bias_level = SND_SOC_BIAS_OFF; + + codec->dai = &jz4740_codec_dai; + codec->num_dai = 1; + + codec->reg_cache = jz4740_codec->reg_cache; + codec->reg_cache_size = 2; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + jz4740_codec_codec = codec; + + snd_soc_update_bits(codec, JZ4740_REG_CODEC_1, + JZ4740_CODEC_1_SW2_ENABLE, JZ4740_CODEC_1_SW2_ENABLE); + + + platform_set_drvdata(pdev, jz4740_codec); + ret = snd_soc_register_codec(codec); + + if (ret) { + dev_err(&pdev->dev, "Failed to register codec\n"); + goto err_iounmap; + } + + ret = snd_soc_register_dai(&jz4740_codec_dai); + if (ret) { + dev_err(&pdev->dev, "Failed to register codec dai\n"); + goto err_unregister_codec; + } + + jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +err_unregister_codec: + snd_soc_unregister_codec(codec); +err_iounmap: + iounmap(jz4740_codec->base); +err_release_mem_region: + release_mem_region(jz4740_codec->mem->start, resource_size(jz4740_codec->mem)); +err_free_codec: + kfree(jz4740_codec); + + return ret; +} + +static int __devexit jz4740_codec_remove(struct platform_device *pdev) +{ + struct jz4740_codec *jz4740_codec = platform_get_drvdata(pdev); + + snd_soc_unregister_dai(&jz4740_codec_dai); + snd_soc_unregister_codec(&jz4740_codec->codec); + + iounmap(jz4740_codec->base); + release_mem_region(jz4740_codec->mem->start, resource_size(jz4740_codec->mem)); + + platform_set_drvdata(pdev, NULL); + kfree(jz4740_codec); + + return 0; +} + +static struct platform_driver jz4740_codec_driver = { + .probe = jz4740_codec_probe, + .remove = __devexit_p(jz4740_codec_remove), + .driver = { + .name = "jz4740-codec", + .owner = THIS_MODULE, + }, +}; + +static int __init jz4740_codec_init(void) +{ + return platform_driver_register(&jz4740_codec_driver); +} +module_init(jz4740_codec_init); + +static void __exit jz4740_codec_exit(void) +{ + platform_driver_unregister(&jz4740_codec_driver); +} +module_exit(jz4740_codec_exit); + +MODULE_DESCRIPTION("JZ4740 SoC internal codec driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:jz-codec"); diff --git a/sound/soc/codecs/jz4740-codec.h b/sound/soc/codecs/jz4740-codec.h new file mode 100644 index 0000000..b5a0691 --- /dev/null +++ b/sound/soc/codecs/jz4740-codec.h @@ -0,0 +1,20 @@ +/* + * Copyright (C) 2009, Lars-Peter Clausen <lars(a)metafoo.de> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ + +#ifndef __SND_SOC_CODECS_JZ4740_CODEC_H__ +#define __SND_SOC_CODECS_JZ4740_CODEC_H__ + +extern struct snd_soc_dai jz4740_codec_dai; +extern struct snd_soc_codec_device soc_codec_dev_jz4740_codec; + +#endif -- 1.5.6.5

4 8

[alsa-devel] [RFC][PATCH 21/26] alsa: ASoC: Add JZ4740 ASoC support
by Lars-Peter Clausen 03 Jun '10

03 Jun '10

This patch adds ASoC support for JZ4740 SoCs I2S module. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: alsa-devel(a)alsa-project.org --- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/jz4740/Kconfig | 13 + sound/soc/jz4740/Makefile | 9 + sound/soc/jz4740/jz4740-i2s.c | 568 +++++++++++++++++++++++++++++++++++++++++ sound/soc/jz4740/jz4740-i2s.h | 18 ++ sound/soc/jz4740/jz4740-pcm.c | 350 +++++++++++++++++++++++++ sound/soc/jz4740/jz4740-pcm.h | 22 ++ 8 files changed, 982 insertions(+), 0 deletions(-) create mode 100644 sound/soc/jz4740/Kconfig create mode 100644 sound/soc/jz4740/Makefile create mode 100644 sound/soc/jz4740/jz4740-i2s.c create mode 100644 sound/soc/jz4740/jz4740-i2s.h create mode 100644 sound/soc/jz4740/jz4740-pcm.c create mode 100644 sound/soc/jz4740/jz4740-pcm.h diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index b1749bc..5a7a724 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -36,6 +36,7 @@ source "sound/soc/s3c24xx/Kconfig" source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" source "sound/soc/txx9/Kconfig" +source "sound/soc/jz4740/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 1470141..fdbe74d 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -14,3 +14,4 @@ obj-$(CONFIG_SND_SOC) += s3c24xx/ obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ obj-$(CONFIG_SND_SOC) += txx9/ +obj-$(CONFIG_SND_SOC) += jz4740/ diff --git a/sound/soc/jz4740/Kconfig b/sound/soc/jz4740/Kconfig new file mode 100644 index 0000000..39df949 --- /dev/null +++ b/sound/soc/jz4740/Kconfig @@ -0,0 +1,13 @@ +config SND_JZ4740_SOC + tristate "SoC Audio for Ingenic JZ4740 SoC" + depends on SOC_JZ4740 && SND_SOC + help + Say Y or M if you want to add support for codecs attached to + the Jz4740 AC97, I2S or SSP interface. You will also need + to select the audio interfaces to support below. + +config SND_JZ4740_SOC_I2S + depends on SND_JZ4740_SOC + tristate "SoC Audio (I2S protocol) for Ingenic jz4740 chip" + help + Say Y if you want to use I2S protocol and I2S codec on Ingenic Jz4740 QI_LB60 board. diff --git a/sound/soc/jz4740/Makefile b/sound/soc/jz4740/Makefile new file mode 100644 index 0000000..1be8d19 --- /dev/null +++ b/sound/soc/jz4740/Makefile @@ -0,0 +1,9 @@ +# +# Jz4740 Platform Support +# +snd-soc-jz4740-objs := jz4740-pcm.o +snd-soc-jz4740-i2s-objs := jz4740-i2s.o + +obj-$(CONFIG_SND_JZ4740_SOC) += snd-soc-jz4740.o +obj-$(CONFIG_SND_JZ4740_SOC_I2S) += snd-soc-jz4740-i2s.o + diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c new file mode 100644 index 0000000..2b139fd --- /dev/null +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -0,0 +1,568 @@ +/* + * Copyright (C) 2010, Lars-Peter Clausen <lars(a)metafoo.de> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ + +#include <linux/init.h> +#include <linux/io.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> + +#include <linux/clk.h> +#include <linux/delay.h> + +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "jz4740-i2s.h" +#include "jz4740-pcm.h" + +#define JZ_REG_AIC_CONF 0x00 +#define JZ_REG_AIC_CTRL 0x04 +#define JZ_REG_AIC_I2S_FMT 0x10 +#define JZ_REG_AIC_FIFO_STATUS 0x14 +#define JZ_REG_AIC_I2S_STATUS 0x1c +#define JZ_REG_AIC_CLK_DIV 0x30 +#define JZ_REG_AIC_FIFO 0x34 + +#define JZ_AIC_CONF_FIFO_RX_THRESHOLD_MASK (0xf << 12) +#define JZ_AIC_CONF_FIFO_TX_THRESHOLD_MASK (0xf << 8) +#define JZ_AIC_CONF_OVERFLOW_PLAY_LAST BIT(6) +#define JZ_AIC_CONF_INTERNAL_CODEC BIT(5) +#define JZ_AIC_CONF_I2S BIT(4) +#define JZ_AIC_CONF_RESET BIT(3) +#define JZ_AIC_CONF_BIT_CLK_MASTER BIT(2) +#define JZ_AIC_CONF_SYNC_CLK_MASTER BIT(1) +#define JZ_AIC_CONF_ENABLE BIT(0) + +#define JZ_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET 12 +#define JZ_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET 8 + +#define JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_MASK (0x7 << 19) +#define JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK (0x7 << 16) +#define JZ_AIC_CTRL_ENABLE_RX_DMA BIT(15) +#define JZ_AIC_CTRL_ENABLE_TX_DMA BIT(14) +#define JZ_AIC_CTRL_MONO_TO_STEREO BIT(11) +#define JZ_AIC_CTRL_SWITCH_ENDIANNESS BIT(10) +#define JZ_AIC_CTRL_SIGNED_TO_UNSIGNED BIT(9) +#define JZ_AIC_CTRL_FLUSH BIT(8) +#define JZ_AIC_CTRL_ENABLE_ROR_INT BIT(6) +#define JZ_AIC_CTRL_ENABLE_TUR_INT BIT(5) +#define JZ_AIC_CTRL_ENABLE_RFS_INT BIT(4) +#define JZ_AIC_CTRL_ENABLE_TFS_INT BIT(3) +#define JZ_AIC_CTRL_ENABLE_LOOPBACK BIT(2) +#define JZ_AIC_CTRL_ENABLE_PLAYBACK BIT(1) +#define JZ_AIC_CTRL_ENABLE_CAPTURE BIT(0) + +#define JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_OFFSET 19 +#define JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET 16 + +#define JZ_AIC_I2S_FMT_DISABLE_BIT_CLK BIT(12) +#define JZ_AIC_I2S_FMT_ENABLE_SYS_CLK BIT(4) +#define JZ_AIC_I2S_FMT_MSB BIT(0) + +#define JZ_AIC_I2S_STATUS_BUSY BIT(2) + +#define JZ_AIC_CLK_DIV_MASK 0xf + +struct jz4740_i2s { + struct resource *mem; + void __iomem *base; + dma_addr_t phys_base; + + struct clk *clk_aic; + struct clk *clk_i2s; + + struct jz4740_pcm_config pcm_config_playback; + struct jz4740_pcm_config pcm_config_capture; +}; + +static inline uint32_t jz4740_i2s_read(const struct jz4740_i2s *i2s, + unsigned int reg) +{ + return readl(i2s->base + reg); +} + +static inline void jz4740_i2s_write(const struct jz4740_i2s *i2s, + unsigned int reg, uint32_t value) +{ + writel(value, i2s->base + reg); +} + +static inline struct jz4740_i2s *jz4740_dai_to_i2s(struct snd_soc_dai *dai) +{ + return dai->private_data; +} + +static int jz4740_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + uint32_t conf, ctrl; + + if (dai->active) + return 0; + + + ctrl = jz4740_i2s_read(i2s, JZ_REG_AIC_CTRL); + ctrl |= JZ_AIC_CTRL_FLUSH; + jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl); + + clk_enable(i2s->clk_i2s); + + conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); + conf |= JZ_AIC_CONF_ENABLE; + jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); + + return 0; +} + +static void jz4740_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + uint32_t conf; + + if (!dai->active) + return; + + conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); + conf &= ~JZ_AIC_CONF_ENABLE; + jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); + + clk_disable(i2s->clk_i2s); +} + + +static int jz4740_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + bool playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + + uint32_t ctrl; + uint32_t mask; + + if (playback) + mask = JZ_AIC_CTRL_ENABLE_PLAYBACK | JZ_AIC_CTRL_ENABLE_TX_DMA; + else + mask = JZ_AIC_CTRL_ENABLE_CAPTURE | JZ_AIC_CTRL_ENABLE_RX_DMA; + + ctrl = jz4740_i2s_read(i2s, JZ_REG_AIC_CTRL); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ctrl |= mask; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ctrl &= ~mask; + break; + default: + return -EINVAL; + } + + jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl); + + return 0; +} + + +static int jz4740_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + + uint32_t format = 0; + uint32_t conf; + + conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); + + conf &= ~(JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + conf |= JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER; + format |= JZ_AIC_I2S_FMT_ENABLE_SYS_CLK; + break; + case SND_SOC_DAIFMT_CBM_CFS: + conf |= JZ_AIC_CONF_SYNC_CLK_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFM: + conf |= JZ_AIC_CONF_BIT_CLK_MASTER; + break; + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_MSB: + format |= JZ_AIC_I2S_FMT_MSB; + break; + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); + jz4740_i2s_write(i2s, JZ_REG_AIC_I2S_FMT, format); + + return 0; +} + +static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + bool playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + enum jz4740_dma_width dma_width; + struct jz4740_pcm_config *pcm_config; + unsigned int sample_size; + uint32_t ctrl; + + ctrl = jz4740_i2s_read(i2s, JZ_REG_AIC_CTRL); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + sample_size = 0; + dma_width = JZ4740_DMA_WIDTH_8BIT; + break; + case SNDRV_PCM_FORMAT_S16: + sample_size = 1; + dma_width = JZ4740_DMA_WIDTH_16BIT; + break; + default: + return -EINVAL; + } + + if (playback) { + ctrl &= ~JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_MASK; + ctrl |= sample_size << JZ_AIC_CTRL_OUTPUT_SAMPLE_SIZE_OFFSET; + } else { + ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_MASK; + ctrl |= sample_size << JZ_AIC_CTRL_INPUT_SAMPLE_SIZE_OFFSET; + } + + switch (params_channels(params)) { + case 2: + break; + case 1: + if (playback) { + ctrl |= JZ_AIC_CTRL_MONO_TO_STEREO; + break; + } + default: /* Falltrough */ + return -EINVAL; + } + + jz4740_i2s_write(i2s, JZ_REG_AIC_CTRL, ctrl); + + if (playback) { + pcm_config = &i2s->pcm_config_playback; + pcm_config->dma_config.dst_width = dma_width; + } else { + pcm_config = &i2s->pcm_config_capture; + pcm_config->dma_config.src_width = dma_width; + } + + + snd_soc_dai_set_dma_data(dai, substream, pcm_config); + + return 0; +} + +static int jz4740_i2s_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + + switch (div_id) { + case JZ4740_I2S_BIT_CLK: + if (div & 1 || div > 16) + return -EINVAL; + jz4740_i2s_write(i2s, JZ_REG_AIC_CLK_DIV, div - 1); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int jz4740_i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + int ret = 0; + struct clk *parent; + + switch (clk_id) { + case JZ4740_I2S_CLKSRC_EXT: + parent = clk_get(NULL, "ext"); + clk_set_parent(i2s->clk_i2s, parent); + break; + case JZ4740_I2S_CLKSRC_PLL: + parent = clk_get(NULL, "pll half"); + clk_set_parent(i2s->clk_i2s, parent); + ret = clk_set_rate(i2s->clk_i2s, freq); + break; + default: + return -EINVAL; + } + clk_put(parent); + + return ret; +} + +static int jz4740_i2s_suspend(struct snd_soc_dai *dai) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + uint32_t conf; + + if (dai->active) { + conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); + conf &= ~JZ_AIC_CONF_ENABLE; + jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); + + clk_disable(i2s->clk_i2s); + } + + clk_disable(i2s->clk_aic); + + return 0; +} + +static int jz4740_i2s_resume(struct snd_soc_dai *dai) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + uint32_t conf; + + clk_enable(i2s->clk_aic); + + if (dai->active) { + clk_enable(i2s->clk_i2s); + + conf = jz4740_i2s_read(i2s, JZ_REG_AIC_CONF); + conf |= JZ_AIC_CONF_ENABLE; + jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); + + } + + return 0; +} + +static int jz4740_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) +{ + struct jz4740_i2s *i2s = jz4740_dai_to_i2s(dai); + uint32_t conf; + + conf = (7 << JZ_AIC_CONF_FIFO_RX_THRESHOLD_OFFSET) | + (8 << JZ_AIC_CONF_FIFO_TX_THRESHOLD_OFFSET) | + JZ_AIC_CONF_OVERFLOW_PLAY_LAST | + JZ_AIC_CONF_I2S | + JZ_AIC_CONF_INTERNAL_CODEC; + + jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, JZ_AIC_CONF_RESET); + jz4740_i2s_write(i2s, JZ_REG_AIC_CONF, conf); + + return 0; +} + + +static struct snd_soc_dai_ops jz4740_i2s_dai_ops = { + .startup = jz4740_i2s_startup, + .shutdown = jz4740_i2s_shutdown, + .trigger = jz4740_i2s_trigger, + .hw_params = jz4740_i2s_hw_params, + .set_fmt = jz4740_i2s_set_fmt, + .set_clkdiv = jz4740_i2s_set_clkdiv, + .set_sysclk = jz4740_i2s_set_sysclk, +}; + +#define JZ4740_I2S_FMTS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE) + +struct snd_soc_dai jz4740_i2s_dai = { + .name = "jz4740-i2s", + .probe = jz4740_i2s_probe, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = JZ4740_I2S_FMTS, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = JZ4740_I2S_FMTS, + }, + .symmetric_rates = 1, + .ops = &jz4740_i2s_dai_ops, + .suspend = jz4740_i2s_suspend, + .resume = jz4740_i2s_resume, +}; +EXPORT_SYMBOL_GPL(jz4740_i2s_dai); + +static void __devinit jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s) +{ + struct jz4740_dma_config *dma_config; + + /* Playback */ + dma_config = &i2s->pcm_config_playback.dma_config; + dma_config->src_width = JZ4740_DMA_WIDTH_32BIT, + dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE; + dma_config->request_type = JZ4740_DMA_TYPE_AIC_TRANSMIT; + dma_config->flags = JZ4740_DMA_SRC_AUTOINC; + dma_config->mode = JZ4740_DMA_MODE_SINGLE; + i2s->pcm_config_playback.fifo_addr = i2s->phys_base + JZ_REG_AIC_FIFO; + + /* Capture */ + dma_config = &i2s->pcm_config_capture.dma_config; + dma_config->dst_width = JZ4740_DMA_WIDTH_32BIT, + dma_config->transfer_size = JZ4740_DMA_TRANSFER_SIZE_16BYTE; + dma_config->request_type = JZ4740_DMA_TYPE_AIC_RECEIVE; + dma_config->flags = JZ4740_DMA_DST_AUTOINC; + dma_config->mode = JZ4740_DMA_MODE_SINGLE; + i2s->pcm_config_capture.fifo_addr = i2s->phys_base + JZ_REG_AIC_FIFO; +} + +static int __devinit jz4740_i2s_dev_probe(struct platform_device *pdev) +{ + struct jz4740_i2s *i2s; + int ret; + + i2s = kzalloc(sizeof(*i2s), GFP_KERNEL); + + if (!i2s) + return -ENOMEM; + + i2s->mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + + if (!i2s->mem) { + ret = -ENOENT; + goto err_free; + } + + i2s->mem = request_mem_region(i2s->mem->start, resource_size(i2s->mem), + pdev->name); + + if (!i2s->mem) { + ret = -EBUSY; + goto err_free; + } + + i2s->base = ioremap_nocache(i2s->mem->start, resource_size(i2s->mem)); + + if (!i2s->base) { + ret = -EBUSY; + goto err_release_mem_region; + } + + i2s->phys_base = i2s->mem->start; + + i2s->clk_aic = clk_get(&pdev->dev, "aic"); + if (IS_ERR(i2s->clk_aic)) { + ret = PTR_ERR(i2s->clk_aic); + goto err_iounmap; + } + + i2s->clk_i2s = clk_get(&pdev->dev, "i2s"); + if (IS_ERR(i2s->clk_i2s)) { + ret = PTR_ERR(i2s->clk_i2s); + goto err_iounmap; + } + + clk_enable(i2s->clk_aic); + + jz4740_i2c_init_pcm_config(i2s); + + jz4740_i2s_dai.private_data = i2s; + ret = snd_soc_register_dai(&jz4740_i2s_dai); + + platform_set_drvdata(pdev, i2s); + + return 0; + +err_iounmap: + iounmap(i2s->base); +err_release_mem_region: + release_mem_region(i2s->mem->start, resource_size(i2s->mem)); +err_free: + kfree(i2s); + + return ret; +} + +static int __devexit jz4740_i2s_dev_remove(struct platform_device *pdev) +{ + struct jz4740_i2s *i2s = platform_get_drvdata(pdev); + + snd_soc_unregister_dai(&jz4740_i2s_dai); + + clk_disable(i2s->clk_aic); + clk_put(i2s->clk_i2s); + clk_put(i2s->clk_aic); + + iounmap(i2s->base); + release_mem_region(i2s->mem->start, resource_size(i2s->mem)); + + platform_set_drvdata(pdev, NULL); + kfree(i2s); + + return 0; +} + +static struct platform_driver jz4740_i2s_driver = { + .probe = jz4740_i2s_dev_probe, + .remove = __devexit_p(jz4740_i2s_dev_remove), + .driver = { + .name = "jz4740-i2s", + .owner = THIS_MODULE, + }, +}; + +static int __init jz4740_i2s_init(void) +{ + return platform_driver_register(&jz4740_i2s_driver); +} +module_init(jz4740_i2s_init); + +static void __exit jz4740_i2s_exit(void) +{ + platform_driver_unregister(&jz4740_i2s_driver); +} +module_exit(jz4740_i2s_exit); + +MODULE_AUTHOR("Lars-Peter Clausen, <lars(a)metafoo.de>"); +MODULE_DESCRIPTION("Ingenic JZ4740 SoC I2S driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:jz4740-i2s"); diff --git a/sound/soc/jz4740/jz4740-i2s.h b/sound/soc/jz4740/jz4740-i2s.h new file mode 100644 index 0000000..da22ed8 --- /dev/null +++ b/sound/soc/jz4740/jz4740-i2s.h @@ -0,0 +1,18 @@ +/* + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _JZ4740_I2S_H +#define _JZ4740_I2S_H + +/* I2S clock source */ +#define JZ4740_I2S_CLKSRC_EXT 0 +#define JZ4740_I2S_CLKSRC_PLL 1 + +#define JZ4740_I2S_BIT_CLK 0 + +extern struct snd_soc_dai jz4740_i2s_dai; + +#endif diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c new file mode 100644 index 0000000..fd1c203 --- /dev/null +++ b/sound/soc/jz4740/jz4740-pcm.c @@ -0,0 +1,350 @@ +/* + * Copyright (C) 2010, Lars-Peter Clausen <lars(a)metafoo.de> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ + +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/slab.h> + +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/mach-jz4740/dma.h> +#include "jz4740-pcm.h" + +struct jz4740_runtime_data { + unsigned int dma_period; + dma_addr_t dma_start; + dma_addr_t dma_pos; + dma_addr_t dma_end; + + struct jz4740_dma_chan *dma; + + dma_addr_t fifo_addr; +}; + +/* identify hardware playback capabilities */ +static const struct snd_pcm_hardware jz4740_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8, + + .rates = SNDRV_PCM_RATE_8000_48000, + .channels_min = 1, + .channels_max = 2, + .period_bytes_min = 16, + .period_bytes_max = 2 * PAGE_SIZE, + .periods_min = 2, + .periods_max = 128, + .buffer_bytes_max = 128 * 2 * PAGE_SIZE, + .fifo_size = 32, +}; + +static void jz4740_pcm_start_transfer(struct jz4740_runtime_data *prtd, int stream) +{ + unsigned int count; + + if (prtd->dma_pos + prtd->dma_period > prtd->dma_end) + count = prtd->dma_end - prtd->dma_pos; + else + count = prtd->dma_period; + + jz4740_dma_disable(prtd->dma); + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + jz4740_dma_set_src_addr(prtd->dma, prtd->dma_pos); + jz4740_dma_set_dst_addr(prtd->dma, prtd->fifo_addr); + } else { + jz4740_dma_set_src_addr(prtd->dma, prtd->fifo_addr); + jz4740_dma_set_dst_addr(prtd->dma, prtd->dma_pos); + } + + jz4740_dma_set_transfer_count(prtd->dma, count); + + jz4740_dma_enable(prtd->dma); + + prtd->dma_pos += prtd->dma_period; + if (prtd->dma_pos >= prtd->dma_end) + prtd->dma_pos = prtd->dma_start; +} + +static void jz4740_pcm_dma_transfer_done(struct jz4740_dma_chan *dma, int err, + void *dev_id) +{ + struct snd_pcm_substream *substream = dev_id; + struct snd_pcm_runtime *runtime = substream->runtime; + struct jz4740_runtime_data *prtd = runtime->private_data; + + snd_pcm_period_elapsed(substream); + + jz4740_pcm_start_transfer(prtd, substream->stream); +} + +static int jz4740_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct jz4740_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct jz4740_pcm_config *config; + + config = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + if (!prtd->dma) { + const char *dma_channel_name; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_channel_name = "PCM Playback"; + else + dma_channel_name = "PCM Capture"; + + prtd->dma = jz4740_dma_request(substream, dma_channel_name); + } + + if (!prtd->dma) + return -EBUSY; + + jz4740_dma_configure(prtd->dma, &config->dma_config); + prtd->fifo_addr = config->fifo_addr; + + jz4740_dma_set_complete_cb(prtd->dma, jz4740_pcm_dma_transfer_done); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->dma_period = params_period_bytes(params); + prtd->dma_start = runtime->dma_addr; + prtd->dma_pos = prtd->dma_start; + prtd->dma_end = prtd->dma_start + runtime->dma_bytes; + + return 0; +} + +static int jz4740_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct jz4740_runtime_data *prtd = substream->runtime->private_data; + + snd_pcm_set_runtime_buffer(substream, NULL); + if (prtd->dma) { + jz4740_dma_free(prtd->dma); + prtd->dma = NULL; + } + + return 0; +} + +static int jz4740_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct jz4740_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + if (!prtd->dma) + return 0; + + prtd->dma_pos = prtd->dma_start; + + return ret; +} + +static int jz4740_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct jz4740_runtime_data *prtd = runtime->private_data; + + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + jz4740_pcm_start_transfer(prtd, substream->stream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + jz4740_dma_disable(prtd->dma); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static snd_pcm_uframes_t jz4740_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct jz4740_runtime_data *prtd = runtime->private_data; + unsigned long count, pos; + snd_pcm_uframes_t offset; + struct jz4740_dma_chan *dma = prtd->dma; + + count = jz4740_dma_get_residue(dma); + if (prtd->dma_pos == prtd->dma_start) + pos = prtd->dma_end - prtd->dma_start - count; + else + pos = prtd->dma_pos - prtd->dma_start - count; + + offset = bytes_to_frames(runtime, pos); + if (offset >= runtime->buffer_size) + offset = 0; + + return offset; +} + +static int jz4740_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct jz4740_runtime_data *prtd; + + snd_soc_set_runtime_hwparams(substream, &jz4740_pcm_hardware); + prtd = kzalloc(sizeof(struct jz4740_runtime_data), GFP_KERNEL); + + if (prtd == NULL) + return -ENOMEM; + + runtime->private_data = prtd; + return 0; +} + +static int jz4740_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct jz4740_runtime_data *prtd = runtime->private_data; + + kfree(prtd); + + return 0; +} + +static int jz4740_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +static struct snd_pcm_ops jz4740_pcm_ops = { + .open = jz4740_pcm_open, + .close = jz4740_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = jz4740_pcm_hw_params, + .hw_free = jz4740_pcm_hw_free, + .prepare = jz4740_pcm_prepare, + .trigger = jz4740_pcm_trigger, + .pointer = jz4740_pcm_pointer, + .mmap = jz4740_pcm_mmap, +}; + +static int jz4740_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = jz4740_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + + buf->area = dma_alloc_noncoherent(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + + return 0; +} + +static void jz4740_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_noncoherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static u64 jz4740_pcm_dmamask = DMA_BIT_MASK(32); + +int jz4740_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &jz4740_pcm_dmamask; + + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (dai->playback.channels_min) { + ret = jz4740_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto err; + } + + if (dai->capture.channels_min) { + ret = jz4740_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto err; + } + +err: + return ret; +} + +struct snd_soc_platform jz4740_soc_platform = { + .name = "jz4740-pcm", + .pcm_ops = &jz4740_pcm_ops, + .pcm_new = jz4740_pcm_new, + .pcm_free = jz4740_pcm_free, +}; +EXPORT_SYMBOL_GPL(jz4740_soc_platform); + +static int __init jz4740_soc_platform_init(void) +{ + return snd_soc_register_platform(&jz4740_soc_platform); +} +module_init(jz4740_soc_platform_init); + +static void __exit jz4740_soc_platform_exit(void) +{ + snd_soc_unregister_platform(&jz4740_soc_platform); +} +module_exit(jz4740_soc_platform_exit); + +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_DESCRIPTION("Ingenic SoC JZ4740 PCM driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/jz4740/jz4740-pcm.h b/sound/soc/jz4740/jz4740-pcm.h new file mode 100644 index 0000000..e3f221e --- /dev/null +++ b/sound/soc/jz4740/jz4740-pcm.h @@ -0,0 +1,22 @@ +/* + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _JZ4740_PCM_H +#define _JZ4740_PCM_H + +#include <linux/dma-mapping.h> +#include <asm/mach-jz4740/dma.h> + +/* platform data */ +extern struct snd_soc_platform jz4740_soc_platform; + +struct jz4740_pcm_config { + struct jz4740_dma_config dma_config; + phys_addr_t fifo_addr; +}; + +#endif -- 1.5.6.5

5 10

[alsa-devel] [RFC][PATCH 26/26] alsa: ASoC: JZ4740: Add qi_lb60 board driver
by Lars-Peter Clausen 03 Jun '10

03 Jun '10

This patch adds ASoC support for the qi_lb60 board. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: alsa-devel(a)alsa-project.org --- sound/soc/jz4740/Kconfig | 8 ++ sound/soc/jz4740/Makefile | 4 + sound/soc/jz4740/qi_lb60.c | 175 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 187 insertions(+), 0 deletions(-) create mode 100644 sound/soc/jz4740/qi_lb60.c diff --git a/sound/soc/jz4740/Kconfig b/sound/soc/jz4740/Kconfig index 39df949..aaa116f 100644 --- a/sound/soc/jz4740/Kconfig +++ b/sound/soc/jz4740/Kconfig @@ -11,3 +11,11 @@ config SND_JZ4740_SOC_I2S tristate "SoC Audio (I2S protocol) for Ingenic jz4740 chip" help Say Y if you want to use I2S protocol and I2S codec on Ingenic Jz4740 QI_LB60 board. + +config SND_JZ4740_SOC_QI_LB60 + tristate "SoC Audio support for Qi Hardware Ben Nanonote" + depends on SND_JZ4740_SOC && JZ4740_QI_LB60 + select SND_JZ4740_SOC_I2S + select SND_SOC_JZ4740_CODEC + help + Say Y if you want to add support for SoC audio of internal codec on Ingenic Jz4740 QI_LB60 board. diff --git a/sound/soc/jz4740/Makefile b/sound/soc/jz4740/Makefile index 1be8d19..be873c1 100644 --- a/sound/soc/jz4740/Makefile +++ b/sound/soc/jz4740/Makefile @@ -7,3 +7,7 @@ snd-soc-jz4740-i2s-objs := jz4740-i2s.o obj-$(CONFIG_SND_JZ4740_SOC) += snd-soc-jz4740.o obj-$(CONFIG_SND_JZ4740_SOC_I2S) += snd-soc-jz4740-i2s.o +# Jz4740 Machine Support +snd-soc-qi-lb60-objs := qi_lb60.o + +obj-$(CONFIG_SND_JZ4740_SOC_QI_LB60) += snd-soc-qi-lb60.o diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c new file mode 100644 index 0000000..c3f1f91 --- /dev/null +++ b/sound/soc/jz4740/qi_lb60.c @@ -0,0 +1,175 @@ +/* + * Copyright (C) 2009, Lars-Peter Clausen <lars(a)metafoo.de> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/gpio.h> + +#include "../codecs/jz4740-codec.h" +#include "jz4740-pcm.h" +#include "jz4740-i2s.h" + + +#define QI_LB60_SND_GPIO JZ_GPIO_PORTB(29) +#define QI_LB60_AMP_GPIO JZ_GPIO_PORTD(4) + +static int qi_lb60_spk_event(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *ctrl, int event) +{ + int on = 0; + if (event & SND_SOC_DAPM_POST_PMU) + on = 1; + else if (event & SND_SOC_DAPM_PRE_PMD) + on = 0; + + gpio_set_value(QI_LB60_SND_GPIO, on); + gpio_set_value(QI_LB60_AMP_GPIO, on); + + return 0; +} + +static const struct snd_soc_dapm_widget qi_lb60_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", qi_lb60_spk_event), + SND_SOC_DAPM_MIC("Mic", NULL), +}; + +static const struct snd_soc_dapm_route qi_lb60_routes[] = { + {"Mic", NULL, "MIC"}, + {"Speaker", NULL, "LOUT"}, + {"Speaker", NULL, "ROUT"}, +}; + +#define QI_LB60_DAIFMT (SND_SOC_DAIFMT_I2S | \ + SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBM_CFM) + +static int qi_lb60_codec_init(struct snd_soc_codec *codec) +{ + int ret; + struct snd_soc_dai *cpu_dai = codec->socdev->card->dai_link->cpu_dai; + struct snd_soc_dai *codec_dai = codec->socdev->card->dai_link->codec_dai; + + snd_soc_dapm_nc_pin(codec, "LIN"); + snd_soc_dapm_nc_pin(codec, "RIN"); + + ret = snd_soc_dai_set_fmt(codec_dai, QI_LB60_DAIFMT); + if (ret < 0) { + dev_err(codec->dev, "Failed to set codec dai format: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_fmt(cpu_dai, QI_LB60_DAIFMT); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cpu dai format: %d\n", ret); + return ret; + } + + snd_soc_dapm_new_controls(codec, qi_lb60_widgets, ARRAY_SIZE(qi_lb60_widgets)); + + snd_soc_dapm_add_routes(codec, qi_lb60_routes, ARRAY_SIZE(qi_lb60_routes)); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link qi_lb60_dai = { + .name = "jz4740-codec", + .stream_name = "jz4740-codec", + .cpu_dai = &jz4740_i2s_dai, + .codec_dai = &jz4740_codec_dai, + .init = qi_lb60_codec_init, +}; + +static struct snd_soc_card qi_lb60 = { + .name = "QI LB60", + .dai_link = &qi_lb60_dai, + .num_links = 1, + .platform = &jz4740_soc_platform, +}; + +static struct snd_soc_device qi_lb60_snd_devdata = { + .card = &qi_lb60, + .codec_dev = &soc_codec_dev_jz4740_codec, +}; + +static struct platform_device *qi_lb60_snd_device; + +static int __init qi_lb60_init(void) +{ + int ret; + + qi_lb60_snd_device = platform_device_alloc("soc-audio", -1); + + if (!qi_lb60_snd_device) + return -ENOMEM; + + + ret = gpio_request(QI_LB60_SND_GPIO, "SND"); + if (ret) { + pr_err("qi_lb60 snd: Failed to request SND GPIO(%d): %d\n", + QI_LB60_SND_GPIO, ret); + goto err_device_put; + } + + ret = gpio_request(QI_LB60_AMP_GPIO, "AMP"); + if (ret) { + pr_err("qi_lb60 snd: Failed to request AMP GPIO(%d): %d\n", + QI_LB60_AMP_GPIO, ret); + goto err_gpio_free_snd; + } + + gpio_direction_output(JZ_GPIO_PORTB(29), 0); + gpio_direction_output(JZ_GPIO_PORTD(4), 0); + + platform_set_drvdata(qi_lb60_snd_device, &qi_lb60_snd_devdata); + qi_lb60_snd_devdata.dev = &qi_lb60_snd_device->dev; + ret = platform_device_add(qi_lb60_snd_device); + if (ret) { + pr_err("qi_lb60 snd: Failed to add snd soc device: %d\n", ret); + goto err_unset_pdata; + } + + return 0; + +err_unset_pdata: + platform_set_drvdata(qi_lb60_snd_device, NULL); +/*err_gpio_free_amp:*/ + gpio_free(QI_LB60_AMP_GPIO); +err_gpio_free_snd: + gpio_free(QI_LB60_SND_GPIO); +err_device_put: + platform_device_put(qi_lb60_snd_device); + + return ret; +} +module_init(qi_lb60_init); + +static void __exit qi_lb60_exit(void) +{ + gpio_free(QI_LB60_AMP_GPIO); + gpio_free(QI_LB60_SND_GPIO); + platform_device_unregister(qi_lb60_snd_device); +} +module_exit(qi_lb60_exit); + +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_DESCRIPTION("ALSA SoC QI LB60 Audio support"); +MODULE_LICENSE("GPL v2"); -- 1.5.6.5

2 1

Re: [alsa-devel] [PATCH] AsoC/s3c: patch for the unnecessary variable 'state' removal
by Mark Brown 03 Jun '10

03 Jun '10

On Thu, Jun 03, 2010 at 12:54:59AM +0800, Wan ZongShun wrote: > The variable 'state' of structure 's3c_ac97_info' seems no use here, > so this patch is to remove the unnecessary variable. > > Signed-off-by: Wan ZongShun <mcuos.com(a)gmail.com> Applied, thanks.

1 0

Re: [alsa-devel] [PATCH] AsoC/atmel: patch for the unnecessary variable removal
by Mark Brown 03 Jun '10

03 Jun '10

On Thu, Jun 03, 2010 at 12:28:46AM +0800, Wan ZongShun wrote: > The variable 'periods' of structure 'atmel_runtime_data' > seems no use in whole atmel alsa driver,so I make a patch > to remove the unnecessary variable. > > Signed-off-by: Wan ZongShun <mcuos.com(a)gmail.com> Applied, thanks. ASoC, BTW - not AsoC.

1 0

[alsa-devel] [PATCH v2 1/4] imx-ssi.c: add new choices to platform configuration
by Eric Bénard 03 Jun '10

03 Jun '10

* introduce 3 new flags to allow a more detailed configuration of the SSI link : IMX_SSI_NET : enable Network Mode IMX_SSI_SYN : enable Synchronous Mode IMX_SSI_USE_I2S_SLAVE : enable I2S Slave Mode * new platform can use these settings without breaking actual platforms. Signed-off-by: Eric Bénard <eric(a)eukrea.com> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> --- arch/arm/plat-mxc/include/mach/ssi.h | 3 +++ sound/soc/imx/imx-ssi.c | 11 +++++++++-- 2 files changed, 12 insertions(+), 2 deletions(-) diff --git a/arch/arm/plat-mxc/include/mach/ssi.h b/arch/arm/plat-mxc/include/mach/ssi.h index c34ded5..63f3c28 100644 --- a/arch/arm/plat-mxc/include/mach/ssi.h +++ b/arch/arm/plat-mxc/include/mach/ssi.h @@ -10,6 +10,9 @@ struct imx_ssi_platform_data { unsigned int flags; #define IMX_SSI_DMA (1 << 0) #define IMX_SSI_USE_AC97 (1 << 1) +#define IMX_SSI_NET (1 << 2) +#define IMX_SSI_SYN (1 << 3) +#define IMX_SSI_USE_I2S_SLAVE (1 << 4) void (*ac97_reset) (struct snd_ac97 *ac97); void (*ac97_warm_reset)(struct snd_ac97 *ac97); }; diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 80b4fee..50f5162 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -83,8 +83,6 @@ static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, /* * SSI DAI format configuration. * Should only be called when port is inactive (i.e. SSIEN = 0). - * Note: We don't use the I2S modes but instead manually configure the - * SSI for I2S because the I2S mode is only a register preset. */ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -99,6 +97,10 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) /* data on rising edge of bclk, frame low 1clk before data */ strcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0; scr |= SSI_SCR_NET; + if (ssi->flags & IMX_SSI_USE_I2S_SLAVE) { + scr &= ~SSI_I2S_MODE_MASK; + scr |= SSI_SCR_I2S_MODE_SLAVE; + } break; case SND_SOC_DAIFMT_LEFT_J: /* data on rising edge of bclk, frame high with data */ @@ -143,6 +145,11 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) strcr |= SSI_STCR_TFEN0; + if (ssi->flags & IMX_SSI_NET) + scr |= SSI_SCR_NET; + if (ssi->flags & IMX_SSI_SYN) + scr |= SSI_SCR_SYN; + writel(strcr, ssi->base + SSI_STCR); writel(strcr, ssi->base + SSI_SRCR); writel(scr, ssi->base + SSI_SCR); -- 1.6.3.3

4 13

[alsa-devel] [PATCH v5 0/5] OMAP/ASoC: McBSP: FIFO handling related fixes
by Peter Ujfalusi 03 Jun '10

03 Jun '10

Hello, Tony: Could you take a look at the arch/arm patches in this series? Thanks! Changes since v3/4: - Fixed commit subjects and messages for OMAP3 related patches - Added Acked-by from Mark, and Jarkko Intro message from the original series: This series aims to correct how the McBSP FIFO is viewed, and handled. Introduction of the problem: OMAP McBSP FIFO is word structured: McBSP2 has 1024 + 256 = 1280 word long buffer, McBSP1,3,4,5 has 128 word long buffer This means, that the size of the FIFO depends on the McBSP word size configuration. For example on McBSP3: 16bit samples: size is 128 * 2 = 256 bytes 32bit samples: size is 128 * 4 = 512 bytes It is simpler to place constraint for buffer and period based on channels. McBSP3 as example again (16 or 32 bit samples): 1 channel (mono): size is 128 frames (128 words) 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words) 4 channels: size is 128 / 4 = 32 frames (4 * 32 words) Since now the McBSP codec supports not only 16bit samples (32biut has been recently added), the FIFO size handling is no longer correct, since it has been hard wired for 16bit word length. The series changes how the users of McBSP are configuring the FIFO: It used to be 0 based (0 meant 1 word threshold). After this series users can configure the threshold in 1 base mode (1 means 1 word threshold). The platform code now provides the full size of the FIFO in words, instead of the already limited value used in the past. In ASoC omap-mcbsp code hw_rule based constraint refinement is going to be used instead of the hardwired static constraint, which was correct only in case of 16bit word length. The hw_rule is refining the minimum buffer size based on the channel number going to be used by the coming stream. In case of threshold mode additional hw_rule refines the maximum allowed period size. The series are generated agains Takashi's sound-2.6: topic/asoc branch. CCing also Eduardo, and Eero since they have worked on the original FIFO/threshold implementation. All commetns and testers are welcome! Peter --- Peter Ujfalusi (5): OMAP: McBSP: Function to query the FIFO size OMAP3: McBSP: Change the way how the FIFO is handled OMAP3: McBSP: Use the port's buffer_size when calculating tx delay ASoC: omap-mcbsp: Save, and use wlen for threshold configuration ASoC: omap-mcbsp: Place correct constraints for streams arch/arm/mach-omap2/mcbsp.c | 10 ++-- arch/arm/plat-omap/include/plat/mcbsp.h | 2 + arch/arm/plat-omap/mcbsp.c | 51 ++++++++++----- sound/soc/omap/omap-mcbsp.c | 112 +++++++++++++++++++++++------- 4 files changed, 128 insertions(+), 47 deletions(-)

3 7

Re: [alsa-devel] [PATCH] AsoC/s3c: patch for the unnecessary variable 'state' removal
by Liam Girdwood 03 Jun '10

03 Jun '10

On Thu, 2010-06-03 at 00:54 +0800, Wan ZongShun wrote: > The variable 'state' of structure 's3c_ac97_info' seems no use here, > so this patch is to remove the unnecessary variable. > > Signed-off-by: Wan ZongShun <mcuos.com(a)gmail.com> > --- > sound/soc/s3c24xx/s3c-ac97.c | 1 - > 1 files changed, 0 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/s3c24xx/s3c-ac97.c b/sound/soc/s3c24xx/s3c-ac97.c > index ecf4fd0..31f6d45 100644 > --- a/sound/soc/s3c24xx/s3c-ac97.c > +++ b/sound/soc/s3c24xx/s3c-ac97.c > @@ -31,7 +31,6 @@ > #define AC_CMD_DATA(x) (x & 0xffff) > > struct s3c_ac97_info { > - unsigned state; > struct clk *ac97_clk; > void __iomem *regs; > struct mutex lock; Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

1 0

Re: [alsa-devel] [PATCH] AsoC/atmel: patch for the unnecessary variable removal
by Liam Girdwood 03 Jun '10

03 Jun '10

On Thu, 2010-06-03 at 00:28 +0800, Wan ZongShun wrote: > The variable 'periods' of structure 'atmel_runtime_data' > seems no use in whole atmel alsa driver,so I make a patch > to remove the unnecessary variable. > > Signed-off-by: Wan ZongShun <mcuos.com(a)gmail.com> > --- > sound/soc/atmel/atmel-pcm.c | 1 - > 1 files changed, 0 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c > index f6b3cc0..dc5249f 100644 > --- a/sound/soc/atmel/atmel-pcm.c > +++ b/sound/soc/atmel/atmel-pcm.c > @@ -77,7 +77,6 @@ struct atmel_runtime_data { > size_t period_size; > > dma_addr_t period_ptr; /* physical address of next period */ > - int periods; /* period index of period_ptr */ > > /* PDC register save */ > u32 pdc_xpr_save; Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

1 0

Re: [alsa-devel] [PATCH V4] ASoC/au1x: use resource_size for au1x
by Wan ZongShun 03 Jun '10

03 Jun '10

Hi mark, How about this patch? Applied? 在 2010年6月1日下午3:16，Wan ZongShun <mcuos.com(a)gmail.com> 写道： > Dear all, > > Use the resource_size function instead of manually calculating the > resource size.This patch can reduce the chance of introducing off-by-one > errors. > > Thanks! > > Signed-off-by: Wan ZongShun<mcuos.com(a)gmail.com> > --- > sound/soc/au1x/psc-ac97.c | 13 +++++-------- > sound/soc/au1x/psc-i2s.c | 13 +++++-------- > sound/soc/au1x/psc.h | 1 - > 3 files changed, 10 insertions(+), 17 deletions(-) > > diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c > index a61ccd2..d14a5a9 100644 > --- a/sound/soc/au1x/psc-ac97.c > +++ b/sound/soc/au1x/psc-ac97.c > @@ -375,12 +375,10 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) > } > > ret = -EBUSY; > - wd->ioarea = request_mem_region(r->start, r->end - r->start + 1, > - "au1xpsc_ac97"); > - if (!wd->ioarea) > + if (!request_mem_region(r->start, resource_size(r), pdev->name)) > goto out0; > > - wd->mmio = ioremap(r->start, 0xffff); > + wd->mmio = ioremap(r->start, resource_size(r)); > if (!wd->mmio) > goto out1; > > @@ -410,8 +408,7 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) > > snd_soc_unregister_dai(&au1xpsc_ac97_dai); > out1: > - release_resource(wd->ioarea); > - kfree(wd->ioarea); > + release_mem_region(r->start, resource_size(r)); > out0: > kfree(wd); > return ret; > @@ -420,6 +417,7 @@ out0: > static int __devexit au1xpsc_ac97_drvremove(struct platform_device *pdev) > { > struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); > + struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0); > > if (wd->dmapd) > au1xpsc_pcm_destroy(wd->dmapd); > @@ -433,8 +431,7 @@ static int __devexit au1xpsc_ac97_drvremove(struct platform_device *pdev) > au_sync(); > > iounmap(wd->mmio); > - release_resource(wd->ioarea); > - kfree(wd->ioarea); > + release_mem_region(r->start, resource_size(r)); > kfree(wd); > > au1xpsc_ac97_workdata = NULL; /* MDEV */ > diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c > index 495be6e..737b238 100644 > --- a/sound/soc/au1x/psc-i2s.c > +++ b/sound/soc/au1x/psc-i2s.c > @@ -321,12 +321,10 @@ static int __init au1xpsc_i2s_drvprobe(struct platform_device *pdev) > } > > ret = -EBUSY; > - wd->ioarea = request_mem_region(r->start, r->end - r->start + 1, > - "au1xpsc_i2s"); > - if (!wd->ioarea) > + if (!request_mem_region(r->start, resource_size(r), pdev->name)) > goto out0; > > - wd->mmio = ioremap(r->start, 0xffff); > + wd->mmio = ioremap(r->start, resource_size(r)); > if (!wd->mmio) > goto out1; > > @@ -362,8 +360,7 @@ static int __init au1xpsc_i2s_drvprobe(struct platform_device *pdev) > > snd_soc_unregister_dai(&au1xpsc_i2s_dai); > out1: > - release_resource(wd->ioarea); > - kfree(wd->ioarea); > + release_mem_region(r->start, resource_size(r)); > out0: > kfree(wd); > return ret; > @@ -372,6 +369,7 @@ out0: > static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev) > { > struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); > + struct resource *r = platform_get_resource(pdev, IORESOURCE_MEM, 0); > > if (wd->dmapd) > au1xpsc_pcm_destroy(wd->dmapd); > @@ -384,8 +382,7 @@ static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev) > au_sync(); > > iounmap(wd->mmio); > - release_resource(wd->ioarea); > - kfree(wd->ioarea); > + release_mem_region(r->start, resource_size(r)); > kfree(wd); > > au1xpsc_i2s_workdata = NULL; /* MDEV */ > diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h > index 32d3807..093775d 100644 > --- a/sound/soc/au1x/psc.h > +++ b/sound/soc/au1x/psc.h > @@ -32,7 +32,6 @@ struct au1xpsc_audio_data { > unsigned long rate; > > unsigned long pm[2]; > - struct resource *ioarea; > struct mutex lock; > struct platform_device *dmapd; > }; > -- > 1.6.3.3 > -- *linux-arm-kernel mailing list mail addr:linux-arm-kernel@lists.infradead.org you can subscribe by: http://lists.infradead.org/mailman/listinfo/linux-arm-kernel * linux-arm-NUC900 mailing list mail addr:NUC900@googlegroups.com main web: https://groups.google.com/group/NUC900 you can subscribe it by sending me mail: mcuos.com(a)gmail.com

3 3

[alsa-devel] ASoC: Hooking a TI CODEC to a i.MX27 MCU
by Stuart Longland 03 Jun '10

03 Jun '10

Hi all, I'm trying to write a new machine driver and CODEC driver for a device we're building, and as yet, haven't figured out the magic that gets ALSA to enumerate all the devices on the board. I hope to be able to release any code here back to the community, but I probably need to clear it with a few people first (and it'd be nice to contribute _working_ code). Scenario: MCU is a Freescale i.MX27 processor on a Ka-Ro TX27 module CODEC is a Texas Instruments TLV320AIC3204, control via I²C, data via I²S. Chip ID for this CODEC is locked at 0x18. Kernel is 2.6.34, userland is Gentoo/ARM via root-over-NFS. My goal: Just to get audio being transmitted along the I²S bus. (Don't care about mixers, etc.) So far my only means of debugging everything is to sprinkle the code liberally with printk's everywhere... crude, but it gives me some idea what's going on. The CODEC driver is at this moment, an empty shell based on the TLV320AIC3x driver already in the tree. I've tried basing a machine driver on the rx51 driver -- substituting the TLV320AIC3x driver for my own shell driver, and swapping the original cpu_dai configuration for the i.MX driver. I suspect this machine driver is giving me grief. My machine code setup is as follows: /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link jem3_dai[] = { { .name = "TLV320AIC3204", .stream_name = "AIC3204", .cpu_dai = &imx_ssi_pcm_dai[0], .codec_dai = &aic3204_dai, .init = jem3_aic3204_init, .ops = &jem3_ops, }, }; /* Audio private data */ static struct aic3204_setup_data jem3_aic3204_setup = { .gpio_func[0] = AIC3204_GPIO1_FUNC_DISABLED, .gpio_func[1] = AIC3204_GPIO2_FUNC_DIGITAL_MIC_INPUT, }; /* Audio card */ static struct snd_soc_card jem3_sound_card = { .name = "JEM3", .dai_link = jem3_dai, .num_links = ARRAY_SIZE(jem3_dai), .platform = &imx_soc_platform, }; /* Audio subsystem */ static struct snd_soc_device jem3_snd_devdata = { .card = &jem3_sound_card, .codec_dev = &soc_codec_dev_aic3204, .codec_data = &jem3_aic3204_setup, }; static struct platform_device *jem3_snd_device; static int __init jem3_soc_init(void) { int err; printk( KERN_INFO "%s: hello...\n", __FUNCTION__ ); jem3_snd_device = platform_device_alloc("soc-audio", -1); printk( KERN_INFO "%s: jem3_snd_device = %p\n", __FUNCTION__, jem3_snd_device ); if (!jem3_snd_device) { err = -ENOMEM; goto err1; } printk( KERN_INFO "%s: calling platform_set_drvdata( %p, %p )\n", __FUNCTION__, jem3_snd_device, &jem3_snd_devdata ); platform_set_drvdata(jem3_snd_device, &jem3_snd_devdata); jem3_snd_devdata.dev = &jem3_snd_device->dev; printk( KERN_INFO "%s: calling platform_device_add(%p)\n", __FUNCTION__, jem3_snd_device ); err = platform_device_add(jem3_snd_device); printk( KERN_INFO "%s: platform_device_add(%p) = %d\n", __FUNCTION__, jem3_snd_device, err ); if (err) goto err2; printk( KERN_INFO "%s = 0 (success)\n", __FUNCTION__ ); return 0; err2: platform_device_put(jem3_snd_device); err1: printk( KERN_INFO "%s = %d\n", __FUNCTION__, err ); return err; } Now this compiles... but when I go to load it; one of two things happens... either practically nothing (at this stage; no modules are loaded prior to calling modprobe): 192 / # modprobe snd-soc-jem3 aic3204_i2c_init: adding driver at bf068f9c aic3204_i2c_init: i2c_add_driver(bf068f9c) = 0 jem3_soc_init: hello... jem3_soc_init: jem3_snd_device = c3c600a0 jem3_soc_init: calling platform_set_drvdata( c3c600a0, bf07a780 ) jem3_soc_init: calling platform_device_add(c3c600a0) jem3_soc_init: platform_device_add(c3c600a0) = 0 jem3_soc_init = 0 (success) 192 / # mount /proc 192 / # cat /proc/asound/cards --- no soundcards --- ... Or it goes kaboom... particularly if I rmmod the snd-soc-jem3 module, but leave snd-soc-tlv320aic3204 in place... 192 / # rmmod snd-soc-jem3 jem3_soc_exit: unregistering... jem3_soc_exit: goodbye 192 / # modprobe snd-soc-jem3 jem3_soc_init: hello... jem3_soc_init: jem3_snd_device = c3c600a0 jem3_soc_init: calling platform_set_drvdata( c3c600a0, bf080780 ) jem3_soc_init: calling platform_device_add(c3c600a0) Unable to handle kernel NULL pointer dereference at virtual address 00000008 pgd = c3e44000 [00000008] *pgd=a3e62031, *pte=00000000, *ppte=00000000 Internal error: Oops: 17 [#1] PREEMPT last sysfs file: Modules linked in: snd_soc_jem3(+) snd_soc_imx snd_soc_tlv320aic3204 snd_soc_core snd_pcm snd_timer snd soundcore snd_page_alloc ac97_bus [last unloaded: snd_soc_jem3] CPU: 0 Not tainted (2.6.34-jacques-jem3 #16) PC is at snd_soc_instantiate_cards+0x2c/0x7b4 [snd_soc_core] LR is at 0x0 pc : [<bf0513d8>] lr : [<00000000>] psr: 20000013 sp : c3e33db0 ip : bf05906c fp : bf059074 r10: 00000001 r9 : bf058fd0 r8 : 00000000 r7 : bf07a790 r6 : c3c600a8 r5 : bf072960 r4 : bf059074 r3 : 00000000 r2 : 00000000 r1 : bf072960 r0 : bf059074 Flags: nzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user Control: 0005317f Table: a3e44000 DAC: 00000015 Process modprobe (pid: 278, stack limit = 0xc3e32270) Stack: (0xc3e33db0 to 0xc3e34000) 3da0: c3e33dec 00000001 c3c600a8 00000000 3dc0: bf0687c8 00000000 c034f1c8 00000000 c3e32000 c0167c18 c02eee78 c3e33e34 3de0: c3e33e6c c3c0e7a8 00000000 c0275b24 c3c15040 c003a008 c3c15040 000000d0 3e00: c02e2fbc c3e32000 c3eae9c8 c3e85428 c3e33e34 00000008 c034f1dc c3eafac0 3e20: c3eafac0 c3e85428 c3eae9c8 c00f2674 00000000 0000072d c3e33e60 c3eae9c8 3e40: c3e85428 c00f2724 c0355190 00000000 c3eae9c8 bf080790 bf080798 c3c600a8 3e60: 00000000 c3c600b0 00000000 c033ec88 00000000 bf052ed0 c0355190 c3c600a8 3e80: c3e33eb0 bf059020 c0355190 c01ac47c bf059020 c01ab394 c3c600a8 00000000 3ea0: c3e33eb0 c3c600a8 c01ab548 c01aa628 c3c044e8 c3e76ad4 c02de99c c3c600a8 3ec0: c3c600a8 c3c600dc 00000000 c01ab5f4 c3c600a8 c3c600a8 00000000 c01aa5a8 3ee0: 00000000 c01a8c20 bf080870 c3c600b0 c3c600b0 00000000 00000000 c0168c18 3f00: c3c600a8 c3c600a8 c3c600a0 00000000 bf083000 c0021b88 00000000 c034578c 3f20: 000192dc c01acaa0 c3c600a8 bf0809ac fffffff4 bf080780 bf083000 bf083090 3f40: 000020f1 bf080864 c3e32000 c0021374 00000000 00000000 00000000 000020f1 3f60: bf080864 00021388 000020f1 bf080864 00021388 00000000 c0021b88 c3e32000 3f80: 00000000 c006e48c 00000001 00000000 00019738 0000cf60 000190b0 00019330 3fa0: 00000080 c00219e0 0000cf60 000190b0 00021388 000020f1 00019370 00019370 3fc0: 0000cf60 000190b0 00019330 00000080 00000000 bec39994 00000000 000192dc 3fe0: 000192d0 bec39524 0000bc98 4010ab44 60000010 00021388 00ffff00 00ffff00 [<bf0513d8>] (snd_soc_instantiate_cards+0x2c/0x7b4 [snd_soc_core]) from [<bf052ed0>] (soc_probe+0x74/0xb0 [snd_soc_core]) [<bf052ed0>] (soc_probe+0x74/0xb0 [snd_soc_core]) from [<c01ac47c>] (platform_drv_probe+0x1c/0x24) [<c01ac47c>] (platform_drv_probe+0x1c/0x24) from [<c01ab394>] (driver_probe_device+0x88/0x180) [<c01ab394>] (driver_probe_device+0x88/0x180) from [<c01aa628>] (bus_for_each_drv+0x60/0x8c) [<c01aa628>] (bus_for_each_drv+0x60/0x8c) from [<c01ab5f4>] (device_attach+0x5c/0x74) [<c01ab5f4>] (device_attach+0x5c/0x74) from [<c01aa5a8>] (bus_probe_device+0x30/0x50) [<c01aa5a8>] (bus_probe_device+0x30/0x50) from [<c01a8c20>] (device_add+0x1f4/0x4c0) [<c01a8c20>] (device_add+0x1f4/0x4c0) from [<c01acaa0>] (platform_device_add+0xf0/0x194) [<c01acaa0>] (platform_device_add+0xf0/0x194) from [<bf083090>] (jem3_soc_init+0x90/0x110 [snd_soc_jem3]) [<bf083090>] (jem3_soc_init+0x90/0x110 [snd_soc_jem3]) from [<c0021374>] (do_one_initcall+0x2c/0x1a8) [<c0021374>] (do_one_initcall+0x2c/0x1a8) from [<c006e48c>] (sys_init_module+0xc4/0x1f8) [<c006e48c>] (sys_init_module+0xc4/0x1f8) from [<c00219e0>] (ret_fast_syscall+0x0/0x2c) Code: e1530002 0a000021 e597203c e5973010 (e5922008) ---[ end trace 2b2b9768e2c2da90 ]--- The address mentioned there makes me think there's an uninitialised pointer to a struct somewhere... but I've never been able to figure out which one. If I reboot, and try to make it oops like before by loading the CODEC driver then machine driver, everything is serine as one would expect: 192 / # modprobe snd-soc-tlv320aic3204 aic3204_i2c_init: adding driver at bf068f9c aic3204_i2c_init: i2c_add_driver(bf068f9c) = 0 192 / # modprobe snd-soc-jem3 jem3_soc_init: hello... jem3_soc_init: jem3_snd_device = c3dbb2a0 jem3_soc_init: calling platform_set_drvdata( c3dbb2a0, bf07a780 ) jem3_soc_init: calling platform_device_add(c3dbb2a0) jem3_soc_init: platform_device_add(c3dbb2a0) = 0 jem3_soc_init = 0 (success) ...but very quiet... 192 / # mount /proc 192 / # cat /proc/asound/cards --- no soundcards --- Queries: - I notice in old drivers, the I²C chip address of the CODEC could be passed in via the same means that is used here for GPIO configuration. How is this done now? Or how do I tell the kernel to only look at address 0x18? - Despite duplicating what I can see being done in other drivers, I still don't see a sound device created. What am I missing to make an audio device appear? - How does one determine what line foo_bar+0x12/0x34 refers to? As you can tell, I'm a newcomer to kernel hacking, so my appologies if these have been answered elsewhere... I've spent many days looking and haven't stumbled upon the answers as yet, hence why I ask here. Thanks in advance. Regards, -- Stuart Longland (aka Redhatter, VK4MSL) .'''. Gentoo Linux/MIPS Cobalt and Docs Developer '.'` : . . . . . . . . . . . . . . . . . . . . . . .'.' http://dev.gentoo.org/~redhatter :.' I haven't lost my mind... ...it's backed up on a tape somewhere.

5 19

Re: [alsa-devel] UAC2 sound-unstable kernel and SDR-widget
by Daniel Mack 03 Jun '10

03 Jun '10

Hi, On Thu, Jun 03, 2010 at 06:41:24AM +0800, Alex Lee wrote: > On 03-Jun-2010, at 1:17 AM, Ken McGuire <kenm(a)paonia.com> wrote: > > >Hi Alex, > >Well said. You might also send a copy to: > >Daniel Mack <daniel(a)caiaq.de> > >since I think he is the source of the current patches. Just out of interest - where did you send that email to? I can't see any such message on the ALSA or LKML mailing lists. > >>We are a group of amateur radio enthusiasts who are developing > >>an open > >>source "sound card" for Software Defined Radio (SDR) applications. > >> > >>The project is called SDR-widget, and you can find more details in: > >> > >>http://code.google.com/p/sdr-widget/ > >> > >>We also have a google group for the sdr-widget developers, and I have > >>just sent you an invitation to join the group. > >> > >>The sdr-widget is a USB sound card based on the AK5394A ADC and a > >>companion AK DAC, controlled by an AT32UC3A3 uProcessor. We already > >>have a prototype working. > >> > >>We have a version of the firmware running with UAC1, at 96khz 24bit > >>mono, and 48khz 24 bit stereo capture. The widget works under Linux > >>and Windows with this firmware. It enumerates as a composite high > >>speed USB device with 4 interfaces: CDC, HID, custom DG8SAQ, and > >>Audio. Wow, interesting project. Thanks for letting us know. > >>We are now working on a version of the firmware with UAC2. Cool. Luckily, as the specs for this standard are freely available, you shouldn't have much problems doing that. From my experience, it's alsmost completely straight forward. > >>We have recently managed to demonstrate 96khz 24bit stereo capture > >>using Ubuntu 10.04 LTS, with kernel 2.6.34-4 and alsa 1.0.23. We are > >>shooting for 192khz 24 bit stereo full duplex eventually. > >> > >>We are now trying to get the sdr-widget running under kernel > >>2.6.35-rc1 with your latest usb-audio updates. That's very good to hear. My current problem is that I can only test the driver with prototypes we made ourselves, and from reading the specs. So more tester's feedback is certainly appreciated. > >>We feel that this is an opportunity for us to work closely, as both > >>the sound-unstable kernel development and the sdr-widget development > >>are actively being pursued, and both are open source projects. This > >>will allow us to make sure both our projects become fully compliant > >>with the UAC2 specs. Agreed. > >>For example, one area we are looking at currently is the different > >>interpretations of the UAC2 specs for the CLOCK SELECT descriptor: > >> > >> > >> > >>This is the audio-v2.h version: > >> > >> > >>/* 4.7.2.2 Clock Source Descriptor */ > >> > >> > >>struct uac_clock_selector_descriptor { > >> __u8 bLength; > >> __u8 bDescriptorType; > >> __u8 bDescriptorSubtype; > >> __u8 bClockID; > >> __u8 bNrInPins; > >> __u8 baCSourceID[]; > >> /* bmControls, bAssocTerminal and iClockSource omitted */ > >> > >> > >>} __attribute__((packed)); > >> > >> > >>baCSourceID is a array of U8. > >> > >> > >>Whereas widget firmware uses this: > >> > >> > >>//! Clock Selector descriptor pp 4.7.2.2 WARNING this descriptor > >>complies with > >>//! the alsa 1.0.23 usb_audio_v2_compat.h which is different from pp. > >>4.7.2.2 > >>typedef > >>#if (defined __ICCAVR32__) > >>#pragma pack(1) > >>#endif > >>struct > >>#if (defined __GNUC__) > >>__attribute__((__packed__)) > >>#endif > >>{ > >> U8 bLength; /* Size of this descriptor in bytes */ > >> U8 bDescriptorType; /* CS_INTERFACE descriptor type */ > >> U8 bDescritorSubtype; /* CLOCK_SELECTOR subtype */ > >> U8 bClockID; /* Clock Selector ID */ > >> U8 bNrInPins; /* Number of Input Pins */ > >> U8 baCSourceID1; /* variable length */ > >> U8 baCSourceID2; > >> U8 bmControls; /* Clock selector control > >>bitmap !!! comes after > >>baCSource in the spec!! */ Yeah, I know. I fixed this up already, and Takashi queued the patch in his tree. Sorry about that, but ironically, you quoted the corrected version above already. Our two structs are actually in sync now. And more than that, the wrong version of the struct was in fact never used. The code which started to parse clock selectors was added along with the fix for the struct. > >>So we are now discussing whether we should modify the sdr-widget > >>firmware, or to patch the kernel :-) Oh, we of course need to keep to the specs and implement it properly on both sides. I can't promise this whole UAC2 stack to be bug-free or complete at this point. As I said, it is only tested by prototypes that work well under Mac OS X, too. So that's at least some kind of evidence :) Just let me know whether you encounter any kind of bug in the driver stack. Patches are very welcome :) Thanks, Daniel

1 0

Re: [alsa-devel] Crash 2.6.32.x, Echo 3G, Alsa, SCSI
by Clemens Ladisch 03 Jun '10

03 Jun '10

Frederick V Heitkamp wrote: > Repeatable Hard Crash. > What else do I need to provide? > > [ 5484.995249] WriteControlReg: not written, no change > [ 5485.069621] divide error: 0000 [#1] pcm_hw_params ok > [ 5485.070159] Prepare rate=44100 format=2 channels=2 > [ 5485.070161] set_audio_format[14] = 5 > [ 5485.070166] Prepare rate=44100 format=2 channels=2 > [ 5485.070167] set_audio_format[14] = 5 > [ 5485.070003] PREEMPT SMP > ... > [ 5485.070003] EIP is at pcm_pointer+0x37/0x70 [snd_echo3g] > ... > [ 5485.965788] [<c10041c0>] ? do_divide_error+0x0/0x90 > [ 5485.980619] [<f87aa037>] ? pcm_pointer+0x37/0x70 [snd_echo3g] > [ 5485.998047] [<c104764e>] ? run_timer_softirq+0x17e/0x2e0 > [ 5486.014175] [<f87ac9bf>] ? snd_echo_interrupt+0x11f/0x240 [snd_echo3g] > [ 5486.033940] [<c107a5d5>] ? handle_IRQ_event+0x45/0x190 bytes_to_frames() divides by runtime->frame_bits which is not set until after the hw_params callback has succeeded, but the corresponding chip->substream[] entry is set in that callback, by init_engine(). It should probably have been set in the prepare callback. Frederick, please try the patch below. Takashi, is there any reason why the runtime-> fields are initialized after the hw_params callback? Regards, Clemens --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -403,12 +403,6 @@ if (err < 0) goto _error; - if (substream->ops->hw_params != NULL) { - err = substream->ops->hw_params(substream, params); - if (err < 0) - goto _error; - } - runtime->access = params_access(params); runtime->format = params_format(params); runtime->subformat = params_subformat(params); @@ -446,6 +440,13 @@ runtime->boundary *= 2; snd_pcm_timer_resolution_change(substream); + + if (substream->ops->hw_params != NULL) { + err = substream->ops->hw_params(substream, params); + if (err < 0) + goto _error; + } + runtime->status->state = SNDRV_PCM_STATE_SETUP; pm_qos_remove_requirement(PM_QOS_CPU_DMA_LATENCY,

6 10

[alsa-devel] Trouble compiling alsa-driver-1.0.23 on Fedora 13
by Uros Bizjak 02 Jun '10

02 Jun '10

Hello! Trying to compile alsa-driver-1.0.23 on Fedora 13, I got following failure: gmake -C /lib/modules/2.6.33.5-112.fc13.x86_64/source SUBDIRS=/home/uros/alsa-driver-1.0.23 CPP="gcc -E" CC="gcc" modules gmake[1]: Entering directory `/usr/src/kernels/2.6.33.5-112.fc13.x86_64' Makefile:535: /usr/src/kernels/2.6.33.5-112.fc13.x86_64/arch//Makefile: No such file or directory gmake[1]: *** No rule to make target `/usr/src/kernels/2.6.33.5-112.fc13.x86_64/arch//Makefile'. Stop. gmake[1]: Leaving directory `/usr/src/kernels/2.6.33.5-112.fc13.x86_64' gmake: *** [compile] Error 2 I have installed required kernel-devel package. Am I missing something else to successfully compile alsa-kernel drivers? Thanks, Uros.

4 5

[PATCH] Ensure codec patch files are checked for the correct codec ID Signed-off-by: David Henningsson <diwic@ubuntu.com>
by David Henningsson 02 Jun '10

02 Jun '10

--- sound/pci/hda/hda_hwdep.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index a1fc837..0e2cb27 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -649,7 +649,9 @@ static void parse_codec_mode(char *buf, struct hda_bus *bus, *codecp = NULL; if (sscanf(buf, "%i %i %i", &vendorid, &subid, &caddr) == 3) { list_for_each_entry(codec, &bus->codec_list, list) { - if (codec->addr == caddr) { + if (codec->vendor_id == vendorid && + codec->subsystem_id == subid && + codec->addr == caddr) { *codecp = codec; break; } -- 1.7.0.4 --------------020601030309080805090208--

1 0

[PATCH] Ensure codec patch files are checked for the correct codec ID Signed-off-by: David Henningsson <diwic@ubuntu.com>
by David Henningsson 02 Jun '10

02 Jun '10

--- sound/pci/hda/hda_hwdep.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index a1fc837..0e2cb27 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -649,7 +649,9 @@ static void parse_codec_mode(char *buf, struct hda_bus *bus, *codecp = NULL; if (sscanf(buf, "%i %i %i", &vendorid, &subid, &caddr) == 3) { list_for_each_entry(codec, &bus->codec_list, list) { - if (codec->addr == caddr) { + if (codec->vendor_id == vendorid && + codec->subsystem_id == subid && + codec->addr == caddr) { *codecp = codec; break; } -- 1.7.0.4 --------------000400040903050100030008--

1 0

Re: [alsa-devel] [PATCH] AsoC/s6000: use resource_size for {request/release}_mem_region and ioremap
by Mark Brown 02 Jun '10

02 Jun '10

On Wed, Jun 02, 2010 at 06:49:53PM +0800, Wan ZongShun wrote: > The size calculation is end - start + 1. But,sometimes, the '1' can > be forgotten carelessly, witch will have potential risk, so use resource_size > for {request/release}_mem_region and ioremap here should be good habit. > > Signed-off-by: Wan ZongShun <mcuos.com(a)gmail.com> CCing in Daniel who wrote the S6000 support. Daniel, do you want to add an entry to MAINTAINERS for this stuff or are you happy without having stuff run past you? > --- > sound/soc/s6000/s6000-i2s.c | 38 +++++++++++++++++--------------------- > 1 files changed, 17 insertions(+), 21 deletions(-) > > diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c > index 5b9ac17..59e3fa7 100644 > --- a/sound/soc/s6000/s6000-i2s.c > +++ b/sound/soc/s6000/s6000-i2s.c > @@ -451,16 +451,15 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev) > goto err_release_none; > } > > - region = request_mem_region(scbmem->start, > - scbmem->end - scbmem->start + 1, > - pdev->name); > + region = request_mem_region(scbmem->start, resource_size(scbmem), > + pdev->name); > if (!region) { > dev_err(&pdev->dev, "I2S SCB region already claimed\n"); > ret = -EBUSY; > goto err_release_none; > } > > - mmio = ioremap(scbmem->start, scbmem->end - scbmem->start + 1); > + mmio = ioremap(scbmem->start, resource_size(scbmem)); > if (!mmio) { > dev_err(&pdev->dev, "can't ioremap SCB region\n"); > ret = -ENOMEM; > @@ -474,9 +473,8 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev) > goto err_release_map; > } > > - region = request_mem_region(sifmem->start, > - sifmem->end - sifmem->start + 1, > - pdev->name); > + region = request_mem_region(sifmem->start, resource_size(sifmem), > + pdev->name); > if (!region) { > dev_err(&pdev->dev, "I2S SIF region already claimed\n"); > ret = -EBUSY; > @@ -490,8 +488,8 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev) > goto err_release_sif; > } > > - region = request_mem_region(dma1->start, dma1->end - dma1->start + 1, > - pdev->name); > + region = request_mem_region(dma1->start, resource_size(dma1), > + pdev->name); > if (!region) { > dev_err(&pdev->dev, "I2S DMA region already claimed\n"); > ret = -EBUSY; > @@ -500,9 +498,8 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev) > > dma2 = platform_get_resource(pdev, IORESOURCE_DMA, 1); > if (dma2) { > - region = request_mem_region(dma2->start, > - dma2->end - dma2->start + 1, > - pdev->name); > + region = request_mem_region(dma2->start, resource_size(dma2), > + pdev->name); > if (!region) { > dev_err(&pdev->dev, > "I2S DMA region already claimed\n"); > @@ -561,15 +558,15 @@ err_release_dev: > kfree(dev); > err_release_dma2: > if (dma2) > - release_mem_region(dma2->start, dma2->end - dma2->start + 1); > + release_mem_region(dma2->start, resource_size(dma2)); > err_release_dma1: > - release_mem_region(dma1->start, dma1->end - dma1->start + 1); > + release_mem_region(dma1->start, resource_size(dma1)); > err_release_sif: > - release_mem_region(sifmem->start, (sifmem->end - sifmem->start) + 1); > + release_mem_region(sifmem->start, resource_size(sifmem)); > err_release_map: > iounmap(mmio); > err_release_scb: > - release_mem_region(scbmem->start, (scbmem->end - scbmem->start) + 1); > + release_mem_region(scbmem->start, resource_size(scbmem)); > err_release_none: > return ret; > } > @@ -590,19 +587,18 @@ static void __devexit s6000_i2s_remove(struct platform_device *pdev) > kfree(dev); > > region = platform_get_resource(pdev, IORESOURCE_DMA, 0); > - release_mem_region(region->start, region->end - region->start + 1); > + release_mem_region(region->start, resource_size(region)); > > region = platform_get_resource(pdev, IORESOURCE_DMA, 1); > if (region) > - release_mem_region(region->start, > - region->end - region->start + 1); > + release_mem_region(region->start, resource_size(region)); > > region = platform_get_resource(pdev, IORESOURCE_MEM, 0); > - release_mem_region(region->start, (region->end - region->start) + 1); > + release_mem_region(region->start, resource_size(region)); > > iounmap(mmio); > region = platform_get_resource(pdev, IORESOURCE_IO, 0); > - release_mem_region(region->start, (region->end - region->start) + 1); > + release_mem_region(region->start, resource_size(region)); > } > > static struct platform_driver s6000_i2s_driver = { > -- > 1.6.3.3

4 9

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 02 Jun '10

02 Jun '10

The following changes since commit 1fab79b8a1bf0d46a7c7e6a1c06dbed1803f403c: Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2010-05-31 18:13:20 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Simple Kconfig fixes, should only affect the relevant platforms. Sascha Hauer (2): ASoC: Add missing Kconfig entry for Phytec boards ASoC: MX31ads sound support should depend on MACH_MX31ADS_WM1133_EV1 sound/soc/imx/Kconfig | 11 ++++++++++- 1 files changed, 10 insertions(+), 1 deletions(-)

2 1

Re: [alsa-devel] [PATCH v2 2/4]AsoC/nuc900: patch for modifing the ac97 delays to minimum
by Liam Girdwood 02 Jun '10

02 Jun '10

On Wed, 2010-06-02 at 13:57 +0800, Wan ZongShun wrote: > This patch is to modify the ac97 delays to minimum, all these 1000 micro > seconds delays seem over spec for the AC97 interface. > > I deleted some unnecessary delays here and changed the AC97 cold and warm reset > delays from 1000ms to 100ms. > I think you mean "from 1000µs to 100µs" here as ms is milliseconds. Liam -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

2 2

Re: [alsa-devel] [PATCH v2 1/3]AsoC/nuc900: patch for SUBSTREAM_TYPE', 'PCM_TX' and 'PCM_RX' removal
by Wan ZongShun 02 Jun '10

02 Jun '10

I am sorry for wrong subject. It should be [PATCH v2 1/4]. > This patch is to remove the 'SUBSTREAM_TYPE','PCM_TX' and 'PCM_RX' definition. > > There is no need to redefine SNDRV_PCM_STREAM_PLAYBACK as PCM_TX, > the SUBSTREAM_TYPE(substream) can be deleted too, the playback or record can be > judged by 'if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)' directly rather > than 'if (PCM_TX == stype)', which makes the codes easy to read. > > Signed-off-by: Wan ZongShun<mcuos.com(a)gmail.com> > > --- > sound/soc/nuc900/nuc900-ac97.c | 6 +++--- > sound/soc/nuc900/nuc900-auido.h | 4 ---- > sound/soc/nuc900/nuc900-pcm.c | 18 ++++++++++-------- > 3 files changed, 13 insertions(+), 15 deletions(-)

1 0

[alsa-devel] [PATCH] regression 2.6.35-rc1+ hda realtek missing beep 10ec0883
by David Fries 02 Jun '10

02 Jun '10

From: David Fries <david(a)fries.net> The PC-Spekaer Beep control worked in 2.6.34, but not 2.6.35-rc1, this is the patch that modified that behavior. "ALSA: hda - Add parse customize define function for Realtek codecs" regression in commit da00c24493bf6ae3772dfe7343dca033ebc75955. Debug output (after this patch), HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 SKU: Nid=0x0 sku_cfg=0x0000140d SKU: port_connectivity=0x0 SKU: enable_pcbeep=0x0 SKU: check_sum=0x00000000 SKU: customization=0x00000000 SKU: external_amp=0x1 SKU: platform_type=0x1 SKU: swap=0x0 SKU: override=0x1 hda_codec: ALC883: BIOS auto-probing. autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) hp_outs=0 (0x0/0x0/0x0/0x0/0x0) mono: mono_out=0x0 dig-out=0x1e/0x0 inputs: mic=0x18, fmic=0x0, line=0x1a, fline=0x0, cd=0x1c, aux=0x0 realtek: Enabling init ASM_ID=0x140d CODEC_ID=10ec0883 input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:14.2/input/input16 Signed-off-by: David Fries <david(a)fries.net> Cc: Kailang Yang <kailang(a)realtek.com> Cc: Takashi Iwai <tiwai(a)suse.de> --- My patch restores both the Beep volume and mute. I'll let someone else determine if this is the correct solution or not. sound/pci/hda/patch_realtek.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 17d4548..c31c8b9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10611,7 +10611,7 @@ static int patch_alc882(struct hda_codec *codec) set_capture_mixer(codec); - if (spec->cdefine.enable_pcbeep) + if (spec->cdefine.enable_pcbeep || codec->vendor_id == 0x10ec0883) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); if (board_config == ALC882_AUTO) -- 1.7.1

1 0

Re: [alsa-devel] [RFC PATCH 2/5] OMAP3: McBSP: Change the way how the FIFO is handled
by Peter Ujfalusi 02 Jun '10

02 Jun '10

On Monday 31 May 2010 20:41:12 ext Nishanth Menon wrote: > On 05/31/2010 11:16 AM, Peter Ujfalusi wrote: > > Use the actual FIFO size in words as buffer_size on OMAP2. > > Change the threshold configuration to use 1 based numbering, when > > specifying the allowed threshold maximum or the McBSP threshold value. > > Set the default maximum threshold to (buffer_size - 0x10) intialy. > > > >> From users of McBSP, now it is expected to use this method. > > > > Asking for threshold 1 means that the value written to threshold > > registers are going to be 0, which means 1 word threshold. > > just a 2cent minor comment: maybe omap_mcbsp_platform_data needs > structure documentation.. it might be difficult for folks to figure that > out from commit ID itself.. I can add comments in the mach-omap2/mcbsp.c like this: > > > Signed-off-by: Peter Ujfalusi<peter.ujfalusi(a)nokia.com> > > --- > > > > arch/arm/mach-omap2/mcbsp.c | 10 +++++----- > > arch/arm/plat-omap/mcbsp.c | 10 ++++++---- > > 2 files changed, 11 insertions(+), 9 deletions(-) > > > > diff --git a/arch/arm/mach-omap2/mcbsp.c b/arch/arm/mach-omap2/mcbsp.c > > index 016fe60..9139958 100644 > > --- a/arch/arm/mach-omap2/mcbsp.c > > +++ b/arch/arm/mach-omap2/mcbsp.c > > @@ -132,7 +132,7 @@ static struct omap_mcbsp_platform_data > > omap34xx_mcbsp_pdata[] = { > > > > .rx_irq = INT_24XX_MCBSP1_IRQ_RX, > > .tx_irq = INT_24XX_MCBSP1_IRQ_TX, > > .ops =&omap2_mcbsp_ops, > > > > - .buffer_size = 0x6F, > > + .buffer_size = 0x80, + .buffer_size = 0x80, /* The FIFO has 128 locations */ > > > > }, > > { > > > > .phys_base = OMAP34XX_MCBSP2_BASE, > > > > @@ -142,7 +142,7 @@ static struct omap_mcbsp_platform_data > > omap34xx_mcbsp_pdata[] = { > > > > .rx_irq = INT_24XX_MCBSP2_IRQ_RX, > > .tx_irq = INT_24XX_MCBSP2_IRQ_TX, > > .ops =&omap2_mcbsp_ops, > > > > - .buffer_size = 0x3FF, > > + .buffer_size = 0x500, + .buffer_size = 0x500, /* The FIFO has 1024 + 256 locations */ > > > > }, > > { > > > > .phys_base = OMAP34XX_MCBSP3_BASE, > > > > @@ -152,7 +152,7 @@ static struct omap_mcbsp_platform_data > > omap34xx_mcbsp_pdata[] = { > > > > .rx_irq = INT_24XX_MCBSP3_IRQ_RX, > > .tx_irq = INT_24XX_MCBSP3_IRQ_TX, > > .ops =&omap2_mcbsp_ops, > > > > - .buffer_size = 0x6F, > > + .buffer_size = 0x80, + .buffer_size = 0x80, /* The FIFO has 128 locations */ > > > > }, > > { > > > > .phys_base = OMAP34XX_MCBSP4_BASE, > > > > @@ -161,7 +161,7 @@ static struct omap_mcbsp_platform_data > > omap34xx_mcbsp_pdata[] = { > > > > .rx_irq = INT_24XX_MCBSP4_IRQ_RX, > > .tx_irq = INT_24XX_MCBSP4_IRQ_TX, > > .ops =&omap2_mcbsp_ops, > > > > - .buffer_size = 0x6F, > > + .buffer_size = 0x80, > > > > }, > > { > > > > .phys_base = OMAP34XX_MCBSP5_BASE, > > > > @@ -170,7 +170,7 @@ static struct omap_mcbsp_platform_data > > omap34xx_mcbsp_pdata[] = { > > > > .rx_irq = INT_24XX_MCBSP5_IRQ_RX, > > .tx_irq = INT_24XX_MCBSP5_IRQ_TX, > > .ops =&omap2_mcbsp_ops, > > > > - .buffer_size = 0x6F, > > + .buffer_size = 0x80, > > > > }, > > > > }; > > #define OMAP34XX_MCBSP_PDATA_SZ ARRAY_SIZE(omap34xx_mcbsp_pdata) > > > > diff --git a/arch/arm/plat-omap/mcbsp.c b/arch/arm/plat-omap/mcbsp.c > > index 51d8abf..6462968 100644 > > --- a/arch/arm/plat-omap/mcbsp.c > > +++ b/arch/arm/plat-omap/mcbsp.c > > @@ -497,7 +497,8 @@ void omap_mcbsp_set_tx_threshold(unsigned int id, u16 > > threshold) > > > > } > > mcbsp = id_to_mcbsp_ptr(id); > > > > - MCBSP_WRITE(mcbsp, THRSH2, threshold); > > + if (threshold&& threshold<= mcbsp->max_tx_thres) > > + MCBSP_WRITE(mcbsp, THRSH2, threshold - 1); > > > > } > > EXPORT_SYMBOL(omap_mcbsp_set_tx_threshold); > > > > @@ -519,7 +520,8 @@ void omap_mcbsp_set_rx_threshold(unsigned int id, u16 > > threshold) > > > > } > > mcbsp = id_to_mcbsp_ptr(id); > > > > - MCBSP_WRITE(mcbsp, THRSH1, threshold); > > + if (threshold&& threshold<= mcbsp->max_rx_thres) > > + MCBSP_WRITE(mcbsp, THRSH1, threshold - 1); > > > > } > > EXPORT_SYMBOL(omap_mcbsp_set_rx_threshold); > > > > @@ -1696,8 +1698,8 @@ static inline void __devinit > > omap34xx_device_init(struct omap_mcbsp *mcbsp) > > > > { > > > > mcbsp->dma_op_mode = MCBSP_DMA_MODE_ELEMENT; > > if (cpu_is_omap34xx()) { > > > > - mcbsp->max_tx_thres = max_thres(mcbsp); > > - mcbsp->max_rx_thres = max_thres(mcbsp); > > + mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10; > > + mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10; > > > > /* > > > > * REVISIT: Set dmap_op_mode to THRESHOLD as default > > * for mcbsp2 instances. Is this sufficient? -- Péter

2 1

[alsa-devel] [PATCH v4 5/5] ASoC: omap-mcbsp: Place correct constraints for streams
by Peter Ujfalusi 01 Jun '10

01 Jun '10

OMAP McBSP FIFO is word structured: McBSP2 has 1024 + 256 = 1280 word long buffer, McBSP1,3,4,5 has 128 word long buffer This means, that the size of the FIFO depends on the McBSP word size configuration. For example on McBSP3: 16bit samples: size is 128 * 2 = 256 bytes 32bit samples: size is 128 * 4 = 512 bytes It is simpler to place constraint for buffer and period based on channels. McBSP3 as example again (16 or 32 bit samples): 1 channel (mono): size is 128 frames (128 words) 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words) 4 channels: size is 128 / 4 = 32 frames (4 * 32 words) Use the second method to place hw_rule on buffer size, and in threshold mode to period size. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- Hello, The only change since v3 is the removal of blank line from the end of omap_mcbsp_hwrule_min_buffersize function. This patch should replace the one in the v3 series. Peter sound/soc/omap/omap-mcbsp.c | 98 +++++++++++++++++++++++++++++++++--------- 1 files changed, 77 insertions(+), 21 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index b06d8f1..aebd3af 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -173,6 +173,50 @@ static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream) omap_mcbsp_set_rx_threshold(mcbsp_data->bus_id, words); } +static int omap_mcbsp_hwrule_min_buffersize(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *buffer_size = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct omap_mcbsp_data *mcbsp_data = rule->private; + struct snd_interval frames; + int size; + + snd_interval_any(&frames); + size = omap_mcbsp_get_fifo_size(mcbsp_data->bus_id); + + frames.min = size / channels->min; + frames.integer = 1; + return snd_interval_refine(buffer_size, &frames); +} + +static int omap_mcbsp_hwrule_max_periodsize(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *period_size = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_pcm_substream *substream = rule->private; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); + struct snd_interval frames; + int size; + + snd_interval_any(&frames); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + size = omap_mcbsp_get_max_tx_threshold(mcbsp_data->bus_id); + else + size = omap_mcbsp_get_max_rx_threshold(mcbsp_data->bus_id); + + frames.max = size / channels->min; + frames.integer = 1; + return snd_interval_refine(period_size, &frames); +} + static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -185,33 +229,45 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, if (!cpu_dai->active) err = omap_mcbsp_request(bus_id); + /* + * OMAP3 McBSP FIFO is word structured. + * McBSP2 has 1024 + 256 = 1280 word long buffer, + * McBSP1,3,4,5 has 128 word long buffer + * This means that the size of the FIFO depends on the sample format. + * For example on McBSP3: + * 16bit samples: size is 128 * 2 = 256 bytes + * 32bit samples: size is 128 * 4 = 512 bytes + * It is simpler to place constraint for buffer and period based on + * channels. + * McBSP3 as example again (16 or 32 bit samples): + * 1 channel (mono): size is 128 frames (128 words) + * 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words) + * 4 channels: size is 128 / 4 = 32 frames (4 * 32 words) + */ if (cpu_is_omap343x()) { int dma_op_mode = omap_mcbsp_get_dma_op_mode(bus_id); - int max_period; /* - * McBSP2 in OMAP3 has 1024 * 32-bit internal audio buffer. - * Set constraint for minimum buffer size to the same than FIFO - * size in order to avoid underruns in playback startup because - * HW is keeping the DMA request active until FIFO is filled. - */ - if (bus_id == 1) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_BUFFER_BYTES, - 4096, UINT_MAX); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - max_period = omap_mcbsp_get_max_tx_threshold(bus_id); - else - max_period = omap_mcbsp_get_max_rx_threshold(bus_id); - - max_period++; - max_period <<= 1; + * The first rule is for the buffer size, we should not allow + * smaller buffer than the FIFO size to avoid underruns + */ + snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + omap_mcbsp_hwrule_min_buffersize, + mcbsp_data, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1); + /* + * In case of threshold mode, the rule will ensure, that the + * period size is not bigger than the maximum allowed threshold + * value. + */ if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_PERIOD_BYTES, - 32, max_period); + snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + omap_mcbsp_hwrule_max_periodsize, + substream, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); } return err; -- 1.7.1

1 0

[alsa-devel] [RFC PATCH 0/5] OMAP/ASoC: McBSP: FIFO handling related fixes
by Peter Ujfalusi 01 Jun '10

01 Jun '10

Hello, This series aims to correct how the McBSP FIFO is viewed, and handled. Introduction of the problem: OMAP McBSP FIFO is word structured: McBSP2 has 1024 + 256 = 1280 word long buffer, McBSP1,3,4,5 has 128 word long buffer This means, that the size of the FIFO depends on the McBSP word size configuration. For example on McBSP3: 16bit samples: size is 128 * 2 = 256 bytes 32bit samples: size is 128 * 4 = 512 bytes It is simpler to place constraint for buffer and period based on channels. McBSP3 as example again (16 or 32 bit samples): 1 channel (mono): size is 128 frames (128 words) 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words) 4 channels: size is 128 / 4 = 32 frames (4 * 32 words) Since now the McBSP codec supports not only 16bit samples (32biut has been recently added), the FIFO size handling is no longer correct, since it has been hard wired for 16bit word length. The series changes how the users of McBSP are configuring the FIFO: It used to be 0 based (0 meant 1 word threshold). After this series users can configure the threshold in 1 base mode (1 means 1 word threshold). The platform code now provides the _full_ size of the FIFO in words, instead of the already limited value used in the past. In ASoC omap-mcbsp code hw_rule based constraint refinement is going to be used instead of the hardwired static constraint, which was correct only in case of 16bit word length. The hw_rule is refining the minimum buffer size based on the channel number going to be used by the coming stream. In case of threshold mode additional hw_rule refines the maximum allowed period size. The series are generated agains Takashi's sound-2.6: topic/asoc branch. CCing also Eduardo, and Eero since they have worked on the original FIFO/threshold implementation. All commetns and testers are welcome! Peter --- Peter Ujfalusi (5): OMAP: McBSP: Function to query the FIFO size OMAP3: McBSP: Change the way how the FIFO is handled OMAP3: McBSP: Use the port's buffer_size when calculating tx delay ASoC: omap-mcbsp: Save, and use wlen for threshold configuration ASoC: omap-mcbsp: Place correct constraints for streams arch/arm/mach-omap2/mcbsp.c | 10 ++-- arch/arm/plat-omap/include/plat/mcbsp.h | 2 + arch/arm/plat-omap/mcbsp.c | 31 ++++++--- sound/soc/omap/omap-mcbsp.c | 112 +++++++++++++++++++++++------- 4 files changed, 114 insertions(+), 41 deletions(-)

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Re: [alsa-devel] [PATCH] ASoC/kirkwood: the 'break' is useless
by Mark Brown 01 Jun '10

01 Jun '10

On Tue, Jun 01, 2010 at 12:56:11PM +0800, Wan ZongShun wrote: > Dear Mark, > > The 'break' is useless here, I think it should be removed. > > Thanks! > > Signed-off-by: Wan ZongShun<mcuos.com(a)gmail.com> Just to emphasise my previous comments about patch formatting it's particularly important to get stuff like that right with cleanup patches like this - with simple updates the effort required to apply the patch (cleaning up the commit message and so on) can be greater than the effort required to make the original change.

1 0

Re: [alsa-devel] [PATCH 1/3] ALSA/NUC900: Fix some codes according to Liam for nuc900 alsa driver.
by Mark Brown 01 Jun '10

01 Jun '10

On Tue, Jun 01, 2010 at 10:48:19AM +0800, Wan ZongShun wrote: > Dear Liam & Mark, > > This patch is to fix some codes according to > Liam for nuc900 alsa driver. So, these changes look like they're fine but since you've not provided a changelog it's hard to tell for sure. It also looks like you've got two different changes in here, some of the code is adjusting the delays used between register writes in reset while other bits change the code over to using the standard method for checking for substream type. As I said to you yesterday please do read and follow the instructions in Documentation/SubmittingPatches - in particular, you need to include a proper changelog in your mail and you shouldn't have any "Dear X" or similar. > Signed-off-by: Wan ZongShun<mcuos.com(a)gmail.com> You're missing a space between your name and the e-mail address too.

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Re: [alsa-devel] [PATCH 3/3] ASoC/NUC900: Fix a typo for nuc900 alsa driver.
by Liam Girdwood 01 Jun '10

01 Jun '10

On Tue, 2010-06-01 at 10:52 +0800, Wan ZongShun wrote: > Dear all, > > This patch is to fix a typo for nuc900 alsa driver. > The changes look fine here, just the commit message and subject is missing information about renaming the file. Thanks Liam -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk

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Re: [alsa-devel] [PATCH 2/3] ALSA/NUC900: Fix a while loop bug for nuc900 alsa driver
by Liam Girdwood 01 Jun '10

01 Jun '10

On Tue, 2010-06-01 at 10:50 +0800, Wan ZongShun wrote: > Dear all, > > This patch is to fix a while loop bug for nuc900 alsa driver. Which bug ? It's best to say the bug you are fixing when submitting a patch like this. Thanks Liam > > Signed-off-by: Wan ZongShun<mcuos.com(a)gmail.com> > --- > sound/soc/nuc900/nuc900-ac97.c | 10 ++++------ > 1 files changed, 4 insertions(+), 6 deletions(-) > > diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c > index 5b864f9..db7fc13 100644 > --- a/sound/soc/nuc900/nuc900-ac97.c > +++ b/sound/soc/nuc900/nuc900-ac97.c > @@ -66,9 +66,8 @@ static unsigned short nuc900_ac97_read(struct snd_ac97 *ac97, > udelay(100); > > /* polling the AC_R_FINISH */ > - val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON); > - val &= AC_R_FINISH; > - while (!val && timeout--) > + while (!(AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON) & AC_R_FINISH) > + && timeout--) > mdelay(1); > > if (!timeout) { > @@ -121,9 +120,8 @@ static void nuc900_ac97_write(struct snd_ac97 *ac97, unsigned > short reg, > udelay(100); > > /* polling the AC_W_FINISH */ > - tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON); > - tmp &= AC_W_FINISH; > - while (tmp && timeout--) > + while ((AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON) & AC_W_FINISH) > + && timeout--) > mdelay(1); > > if (!timeout)

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Re: [alsa-devel] [PATCH 1/3] ALSA/NUC900: Fix some codes according to Liam for nuc900 alsa driver.
by Liam Girdwood 01 Jun '10

01 Jun '10

On Tue, 2010-06-01 at 10:48 +0800, Wan ZongShun wrote: > Dear Liam & Mark, > > This patch is to fix some codes according to > Liam for nuc900 alsa driver. > > Signed-off-by: Wan ZongShun<mcuos.com(a)gmail.com> The changes here looks fine. Can you resend and add a summary of the changes to the commit message. It's also good practice to split into a new patch for every issue you are fixing, i.e. in this case I would have one patch for the AC97 delays and another patch for the PCM_TX removal. Thanks Liam > > --- > sound/soc/nuc900/nuc900-ac97.c | 20 +++++--------------- > sound/soc/nuc900/nuc900-auido.h | 4 ---- > sound/soc/nuc900/nuc900-pcm.c | 18 ++++++++++-------- > 3 files changed, 15 insertions(+), 27 deletions(-) > > diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c > index f7b44e0..5b864f9 100644 > --- a/sound/soc/nuc900/nuc900-ac97.c > +++ b/sound/soc/nuc900/nuc900-ac97.c > @@ -149,7 +149,7 @@ static void nuc900_ac97_warm_reset(struct snd_ac97 *ac97) > val |= AC_W_RES; > AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val); > > - udelay(1000); > + udelay(100); > > val = nuc900_checkready(); > if (!!val) > @@ -170,40 +170,30 @@ static void nuc900_ac97_cold_reset(struct snd_ac97 *ac97) > val |= ACTL_RESET_BIT; > AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); > > - udelay(1000); > - > val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); > val &= (~ACTL_RESET_BIT); > AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); > > - udelay(1000); > - > /* reset AC-link interface */ > > val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); > val |= AC_RESET; > AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); > > - udelay(1000); > - > val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); > val &= ~AC_RESET; > AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); > > - udelay(1000); > - > /* cold reset AC 97 */ > val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON); > val |= AC_C_RES; > AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val); > > - udelay(1000); > - > val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON); > val &= (~AC_C_RES); > AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val); > > - udelay(1000); > + udelay(100); > > mutex_unlock(&ac97_mutex); > > @@ -222,7 +212,7 @@ static int nuc900_ac97_trigger(struct snd_pcm_substream > *substream, > int cmd, struct snd_soc_dai *dai) > { > struct nuc900_audio *nuc900_audio = nuc900_ac97_data; > - int ret, stype = SUBSTREAM_TYPE(substream); > + int ret; > unsigned long val, tmp; > > ret = 0; > @@ -231,7 +221,7 @@ static int nuc900_ac97_trigger(struct snd_pcm_substream > *substream, > case SNDRV_PCM_TRIGGER_START: > case SNDRV_PCM_TRIGGER_RESUME: > val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); > - if (PCM_TX == stype) { > + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { > tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0); > tmp |= (SLOT3_VALID | SLOT4_VALID | VALID_FRAME); > AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp); > @@ -254,7 +244,7 @@ static int nuc900_ac97_trigger(struct snd_pcm_substream > *substream, > case SNDRV_PCM_TRIGGER_STOP: > case SNDRV_PCM_TRIGGER_SUSPEND: > val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); > - if (PCM_TX == stype) { > + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { > tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0); > tmp &= ~(SLOT3_VALID | SLOT4_VALID); > AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp); > diff --git a/sound/soc/nuc900/nuc900-auido.h b/sound/soc/nuc900/nuc900-auido.h > index 95ac4ef..3038f51 100644 > --- a/sound/soc/nuc900/nuc900-auido.h > +++ b/sound/soc/nuc900/nuc900-auido.h > @@ -96,10 +96,6 @@ > #define RESET_PRSR 0x00 > #define AUDIO_WRITE(addr, val) __raw_writel(val, addr) > #define AUDIO_READ(addr) __raw_readl(addr) > -#define PCM_TX 0 > -#define PCM_RX 1 > -#define SUBSTREAM_TYPE(substream) \ > - ((substream)->stream == SNDRV_PCM_STREAM_PLAYBACK ? PCM_TX : PCM_RX) > > struct nuc900_audio { > void __iomem *mmio; > diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c > index 32a503c..445a180 100644 > --- a/sound/soc/nuc900/nuc900-pcm.c > +++ b/sound/soc/nuc900/nuc900-pcm.c > @@ -47,7 +47,7 @@ static int nuc900_dma_hw_params(struct snd_pcm_substream > *substream, > { > struct snd_pcm_runtime *runtime = substream->runtime; > struct nuc900_audio *nuc900_audio = runtime->private_data; > - unsigned long flags, stype = SUBSTREAM_TYPE(substream); > + unsigned long flags; > int ret = 0; > > spin_lock_irqsave(&nuc900_audio->lock, flags); > @@ -57,8 +57,9 @@ static int nuc900_dma_hw_params(struct snd_pcm_substream > *substream, > return ret; > > nuc900_audio->substream = substream; > - nuc900_audio->dma_addr[stype] = runtime->dma_addr; > - nuc900_audio->buffersize[stype] = params_buffer_bytes(params); > + nuc900_audio->dma_addr[substream->stream] = runtime->dma_addr; > + nuc900_audio->buffersize[substream->stream] = > + params_buffer_bytes(params); > > spin_unlock_irqrestore(&nuc900_audio->lock, flags); > > @@ -72,7 +73,7 @@ static void nuc900_update_dma_register(struct > snd_pcm_substream *substream, > struct nuc900_audio *nuc900_audio = runtime->private_data; > void __iomem *mmio_addr, *mmio_len; > > - if (SUBSTREAM_TYPE(substream) == PCM_TX) { > + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { > mmio_addr = nuc900_audio->mmio + ACTL_PDSTB; > mmio_len = nuc900_audio->mmio + ACTL_PDST_LENGTH; > } else { > @@ -167,18 +168,19 @@ static int nuc900_dma_prepare(struct snd_pcm_substream > *substream) > { > struct snd_pcm_runtime *runtime = substream->runtime; > struct nuc900_audio *nuc900_audio = runtime->private_data; > - unsigned long flags, val, stype = SUBSTREAM_TYPE(substream);; > + unsigned long flags, val; > > spin_lock_irqsave(&nuc900_audio->lock, flags); > > nuc900_update_dma_register(substream, > - nuc900_audio->dma_addr[stype], nuc900_audio->buffersize[stype]); > + nuc900_audio->dma_addr[substream->stream], > + nuc900_audio->buffersize[substream->stream]); > > val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); > > switch (runtime->channels) { > case 1: > - if (PCM_TX == stype) { > + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { > val &= ~(PLAY_LEFT_CHNNEL | PLAY_RIGHT_CHNNEL); > val |= PLAY_RIGHT_CHNNEL; > } else { > @@ -188,7 +190,7 @@ static int nuc900_dma_prepare(struct snd_pcm_substream > *substream) > AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); > break; > case 2: > - if (PCM_TX == stype) > + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) > val |= (PLAY_LEFT_CHNNEL | PLAY_RIGHT_CHNNEL); > else > val |= (RECORD_LEFT_CHNNEL | RECORD_RIGHT_CHNNEL);

1 0

[alsa-devel] snd_hda_intel : broken microphone
by mutenmaster＠tiscali.it 01 Jun '10

01 Jun '10

hi there, I'd like to report a possible bug. After this kernel linux commit: f5271101faf1655d862849f42518c2a88ef394fb my front microphone stopped working, and as of the most recent version, the front microphone still doesn't work. Doing $ arecord -f cd prova.wav produces a mere 44 bytes long file. some versions after the commit pointed above create a file that is bigger than 44 bytes, but plays nothing at all. I just checked the rear microphone,and the result is the same. I attached my codec0 and lspci -vvv output, please let me know if you need anything else. best, vasilis Risparmia con Tutto Incluso Light: telefono + adsl 8 mega a soli 14,95 € al mese. Gratis la Sim Tiscali Mobile con 25 euro di traffico! L'offerta è valida solo se attivi entro il 03/06/10 http://abbonati.tiscali.it/telefono-adsl/prodotti/tc/tuttoincluso_light/?WT…

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Re: [alsa-devel] Problem with snd_hda_intel from 05-27-2010 snapshot
by Jaroslav Kysela 01 Jun '10

01 Jun '10

On Mon, 31 May 2010, VDR User wrote: > On Mon, May 31, 2010 at 1:43 PM, Jaroslav Kysela <perex(a)perex.cz> wrote: >>> I've got a bit of a strange result. With the above patch (and the >>> wallclk logging patch from your previous email), sound does not work. >>> However, it _does_ start working only if I `echo 29 > >>> /proc/asound/card1/pcm3p/xrun_debug`. I tested these results with a >>> reboot in between each attempt just to make sure I'm working with a >>> clean boot. I can't imagine what's causing this weird condition. Any >>> ideas? >> >> Try to disable jiffies check (value 25 for xrun_debug). Please, post your >> log for value 29 for inspection. Thanks. > > With xrun_debug 25 I get sound for about 1 second and then it > disappears. 29 of course works fine as mentioned previously. Thanks. Here is next correction: diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 170610e..b6ba210 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1910,11 +1910,11 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) if (WARN_ONCE(!azx_dev->period_bytes, "hda-intel: zero azx_dev->period_bytes")) return -1; /* this shouldn't happen! */ - if (wallclk <= azx_dev->period_wallclk && + if (wallclk < (azx_dev->period_wallclk * 3) / 2 && pos % azx_dev->period_bytes > azx_dev->period_bytes / 2) /* NG - it's below the first next period boundary */ return bdl_pos_adj[chip->dev_index] ? 0 : -1; - azx_dev->start_wallclk = wallclk; + azx_dev->start_wallclk += wallclk; return 1; /* OK, it's fine */ } Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

2 1

[alsa-devel] [RFC PATCH 0/5] OMAP/ASoC: McBSP: FIFO handling related fixes
by Peter Ujfalusi 01 Jun '10

01 Jun '10

Hello, This series aims to correct how the McBSP FIFO is viewed, and handled. Introduction of the problem: OMAP McBSP FIFO is word structured: McBSP2 has 1024 + 256 = 1280 word long buffer, McBSP1,3,4,5 has 128 word long buffer This means, that the size of the FIFO depends on the McBSP word size configuration. For example on McBSP3: 16bit samples: size is 128 * 2 = 256 bytes 32bit samples: size is 128 * 4 = 512 bytes It is simpler to place constraint for buffer and period based on channels. McBSP3 as example again (16 or 32 bit samples): 1 channel (mono): size is 128 frames (128 words) 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words) 4 channels: size is 128 / 4 = 32 frames (4 * 32 words) Since now the McBSP codec supports not only 16bit samples (32biut has been recently added), the FIFO size handling is no longer correct, since it has been hard wired for 16bit word length. The series changes how the users of McBSP are configuring the FIFO: It used to be 0 based (0 meant 1 word threshold). After this series users can configure the threshold in 1 base mode (1 means 1 word threshold). The platform code now provides the _full_ size of the FIFO in words, instead of the already limited value used in the past. In ASoC omap-mcbsp code hw_rule based constraint refinement is going to be used instead of the hardwired static constraint, which was correct only in case of 16bit word length. The hw_rule is refining the minimum buffer size based on the channel number going to be used by the coming stream. In case of threshold mode additional hw_rule refines the maximum allowed period size. The series are generated agains Takashi's sound-2.6: topic/asoc branch. CCing also Eduardo, and Eero since they have worked on the original FIFO/threshold implementation. All commetns and testers are welcome! Peter --- Peter Ujfalusi (5): OMAP: McBSP: Function to query the FIFO size OMAP3: McBSP: Change the way how the FIFO is handled OMAP3: McBSP: Use the port's buffer_size when calculating tx delay ASoC: omap-mcbsp: Save, and use wlen for threshold configuration ASoC: omap-mcbsp: Place correct constraints for streams arch/arm/mach-omap2/mcbsp.c | 10 ++-- arch/arm/plat-omap/include/plat/mcbsp.h | 2 + arch/arm/plat-omap/mcbsp.c | 31 ++++++--- sound/soc/omap/omap-mcbsp.c | 112 +++++++++++++++++++++++------- 4 files changed, 114 insertions(+), 41 deletions(-)

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Re: [alsa-devel] [PATCH v3] AsoC/au1x: use resource_size
by Manuel Lauss 01 Jun '10

01 Jun '10

2010/6/1 Wan ZongShun <mcuos.com(a)gmail.com>: > Dear Mark & Manuel, > > I re-modify the patch to use resource_size. > > My mail client has been set as line wrap when more 80 chars, > so it causes the wordwrap issues for previous patch. > > Now, I fixed it. > > Thanks! > > > Signed-off-by: Wan ZongShun<mcuos.com(a)gmail.com> The patch is fine with me, all that's missing now is a proper Changelog entry. Manuel

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[alsa-devel] [PATCH] ALSA: hda: Use LPIB for ASUS M2V
by Daniel Chen 01 Jun '10

01 Jun '10

Hi, Attached is a patch against sound-2.6 master that resolves the original reporter's symptom of the PulseAudio daemon crashing during playback. Please apply. This patch is applicable to all 2.6.3x stable series. Best, -Dan

2 1

Re: [alsa-devel] Problem with snd_hda_intel from 05-27-2010 snapshot
by Jaroslav Kysela 31 May '10

31 May '10

On Sun, 30 May 2010, VDR User wrote: > On Sun, May 30, 2010 at 8:37 AM, Jaroslav Kysela <perex(a)perex.cz> wrote: >> Thanks. The wallclock register seems working OK. Please, use value 29 for >> the xrun_debug file to do more accurate inspection what's going wrong (as I >> suggested in my first reply): > > Attached is my full syslog with xrun_debug 29. Thanks. Could you try this patch? diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 170610e..ac3dbab 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1910,7 +1910,7 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) if (WARN_ONCE(!azx_dev->period_bytes, "hda-intel: zero azx_dev->period_bytes")) return -1; /* this shouldn't happen! */ - if (wallclk <= azx_dev->period_wallclk && + if (wallclk < (azx_dev->period_wallclk * 3) / 2 && pos % azx_dev->period_bytes > azx_dev->period_bytes / 2) /* NG - it's below the first next period boundary */ return bdl_pos_adj[chip->dev_index] ? 0 : -1; ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

2 2

[alsa-devel] [PATCH 1/9] ALSA: usb-audio: UAC2: clean up parsing of bmaControls
by Daniel Mack 31 May '10

31 May '10

Introduce two new static inline functions for a more readable parsing of UAC2 bmaControls. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- include/linux/usb/audio-v2.h | 10 ++++++++++ sound/usb/mixer.c | 8 ++++---- 2 files changed, 14 insertions(+), 4 deletions(-) diff --git a/include/linux/usb/audio-v2.h b/include/linux/usb/audio-v2.h index 92f1d99..8f91be3 100644 --- a/include/linux/usb/audio-v2.h +++ b/include/linux/usb/audio-v2.h @@ -18,6 +18,16 @@ /* v1.0 and v2.0 of this standard have many things in common. For the rest * of the definitions, please refer to audio.h */ +static inline bool uac2_control_is_readable(u32 bmControls, u8 control) +{ + return (bmControls >> (control * 2)) & 0x1; +} + +static inline bool uac2_control_is_writeable(u32 bmControls, u8 control) +{ + return (bmControls >> (control * 2)) & 0x2; +} + /* 4.7.2.1 Clock Source Descriptor */ struct uac_clock_source_descriptor { diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 03ce971..43d6417 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1188,9 +1188,9 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void for (j = 0; j < channels; j++) { unsigned int mask = snd_usb_combine_bytes(bmaControls + csize * (j+1), csize); - if (mask & (1 << (i * 2))) { + if (uac2_control_is_readable(mask, i)) { ch_bits |= (1 << j); - if (~mask & (1 << ((i * 2) + 1))) + if (!uac2_control_is_writeable(mask, i)) ch_read_only |= (1 << j); } } @@ -1198,9 +1198,9 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void /* FIXME: the whole unit is read-only if any of the channels is marked read-only */ if (ch_bits & 1) /* the first channel must be set (for ease of programming) */ build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid, !!ch_read_only); - if (master_bits & (1 << i * 2)) + if (uac2_control_is_readable(master_bits, i)) build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, - ~master_bits & (1 << ((i * 2) + 1))); + !uac2_control_is_writeable(master_bits, i)); } } -- 1.7.1

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[alsa-devel] [PATCH] ASoC: spdif: Add codec driver to use spdif stand-alone
by Seungwhan Youn 31 May '10

31 May '10

This patch adds spdif dummy codec driver for using spdif-dit as a stand-alone. Until this, spdif-dit can be used only with other codecs like tlv320aci3x in davinci platform. Signed-off-by: Seungwhan Youn <sw.youn(a)samsung.com> --- sound/soc/codecs/spdif_transciever.c | 93 +++++++++++++++++++++++++++++++++- 1 files changed, 92 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c index a631911..f0945ab 100644 --- a/sound/soc/codecs/spdif_transciever.c +++ b/sound/soc/codecs/spdif_transciever.c @@ -18,6 +18,7 @@ #include <linux/moduleparam.h> #include <sound/soc.h> #include <sound/pcm.h> +#include <sound/initval.h> #include "spdif_transciever.h" @@ -26,6 +27,48 @@ MODULE_LICENSE("GPL"); #define STUB_RATES SNDRV_PCM_RATE_8000_96000 #define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE +static struct snd_soc_codec *spdif_dit_codec; + +static int spdif_dit_codec_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + if (spdif_dit_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = spdif_dit_codec; + codec = spdif_dit_codec; + + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto err_create_pcms; + } + + return 0; + +err_create_pcms: + return ret; +} + +static int spdif_dit_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_spdif_dit = { + .probe = spdif_dit_codec_probe, + .remove = spdif_dit_codec_remove, +}; EXPORT_SYMBOL_GPL(soc_codec_dev_spdif_dit); + struct snd_soc_dai dit_stub_dai = { .name = "DIT", .playback = { @@ -40,13 +83,61 @@ EXPORT_SYMBOL_GPL(dit_stub_dai); static int spdif_dit_probe(struct platform_device *pdev) { + struct snd_soc_codec *codec; + int ret; + + if (spdif_dit_codec) { + dev_err(&pdev->dev, "Another Codec is registered\n"); + ret = -EINVAL; + goto err_reg_codec; + } + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + codec->dev = &pdev->dev; + + mutex_init(&codec->mutex); + + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->name = "spdif-dit"; + codec->owner = THIS_MODULE; + codec->dai = &dit_stub_dai; + codec->num_dai = 1; + + spdif_dit_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err_reg_codec; + } + dit_stub_dai.dev = &pdev->dev; - return snd_soc_register_dai(&dit_stub_dai); + ret = snd_soc_register_dai(&dit_stub_dai); + if (ret < 0) { + dev_err(codec->dev, "Failed to register dai: %d\n", ret); + goto err_reg_dai; + } + + return 0; + +err_reg_dai: + snd_soc_unregister_codec(codec); +err_reg_codec: + kfree(spdif_dit_codec); + return ret; } static int spdif_dit_remove(struct platform_device *pdev) { snd_soc_unregister_dai(&dit_stub_dai); + snd_soc_unregister_codec(spdif_dit_codec); + kfree(spdif_dit_codec); + spdif_dit_codec = NULL; return 0; } -- 1.6.2.5

4 4

[alsa-devel] i.MX SSI Kconfig fixes
by Sascha Hauer 31 May '10

31 May '10

Hi, Two small fixes for i.MX SSI. Sascha The following changes since commit 67a3e12b05e055c0415c556a315a3d3eb637e29e: Linux 2.6.35-rc1 (2010-05-30 13:21:02 -0700) are available in the git repository at: git://git.pengutronix.de/git/imx/linux-2.6.git imx-ssi-fixes Sascha Hauer (2): i.MX ssi: MX31ads sound support should depend on MACH_MX31ADS i.MX ssi: Add missing Kconfig entry for Phytec boards sound/soc/imx/Kconfig | 11 ++++++++++- 1 files changed, 10 insertions(+), 1 deletions(-)

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[alsa-devel] ASoC updates for 2.6.36
by Mark Brown 31 May '10

31 May '10

The following changes since commit 67a3e12b05e055c0415c556a315a3d3eb637e29e: Linux 2.6.35-rc1 (2010-05-30 13:21:02 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.36 This includes a merge up of 2.6.35-rc1, plus the commits below. Barry Song (1): ASoC: ad193x: add set_sysclk entry to support different clock input Jassi Brar (2): ASOC: S5PC100: Enable AC97 support ASOC: S5PV210: Enable AC97 support Mark Brown (7): Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-2.6.36 ASoC: Remove version display from WM8990 ASoC: Fix dB scales for WM835x ASoC: Fix dB scales for WM8400 ASoC: Fix dB scales for WM8990 Merge commit 'v2.6.35-rc1' into for-2.6.36 ASoC: Fix S/PDIF build Seungwhan Youn (1): ASoC: spdif: Add codec driver to use spdif stand-alone Wan ZongShun (1): ASoC: NUC900/audio: add nuc900 audio driver support apatard(a)mandriva.com (7): ASoC: Add SOC_DOUBLE_R_SX_TLV control ASoC: Add SND_SOC_DAPM_PRE_POST_PMD event ASoC: Add driver for cs42l51 orion/kirkwood: add audio functions. openrd-client: initialise audio ASoC: kirkwood: Add i2s support ASoC: kirkwood: Add audio support to openrd client platforms arch/arm/mach-kirkwood/common.c | 38 ++ arch/arm/mach-kirkwood/common.h | 2 + arch/arm/mach-kirkwood/include/mach/kirkwood.h | 3 + arch/arm/mach-kirkwood/openrd-setup.c | 13 + arch/arm/plat-orion/include/plat/audio.h | 11 + include/sound/soc-dapm.h | 2 + include/sound/soc.h | 21 + sound/soc/Kconfig | 2 + sound/soc/Makefile | 2 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad193x.c | 41 ++- sound/soc/codecs/ad193x.h | 5 + sound/soc/codecs/cs42l51.c | 763 ++++++++++++++++++++++++ sound/soc/codecs/cs42l51.h | 163 +++++ sound/soc/codecs/spdif_transciever.c | 94 +++- sound/soc/codecs/spdif_transciever.h | 1 + sound/soc/codecs/wm8990.c | 4 - sound/soc/kirkwood/Kconfig | 20 + sound/soc/kirkwood/Makefile | 9 + sound/soc/kirkwood/kirkwood-dma.c | 383 ++++++++++++ sound/soc/kirkwood/kirkwood-dma.h | 17 + sound/soc/kirkwood/kirkwood-i2s.c | 485 +++++++++++++++ sound/soc/kirkwood/kirkwood-i2s.h | 17 + sound/soc/kirkwood/kirkwood-openrd.c | 126 ++++ sound/soc/kirkwood/kirkwood.h | 126 ++++ sound/soc/nuc900/Kconfig | 27 + sound/soc/nuc900/Makefile | 11 + sound/soc/nuc900/nuc900-ac97.c | 442 ++++++++++++++ sound/soc/nuc900/nuc900-audio.c | 81 +++ sound/soc/nuc900/nuc900-auido.h | 121 ++++ sound/soc/nuc900/nuc900-pcm.c | 352 +++++++++++ sound/soc/s3c24xx/Kconfig | 4 +- sound/soc/s3c24xx/smdk_wm9713.c | 3 + sound/soc/soc-core.c | 95 +++ 35 files changed, 3482 insertions(+), 8 deletions(-) create mode 100644 arch/arm/plat-orion/include/plat/audio.h create mode 100644 sound/soc/codecs/cs42l51.c create mode 100644 sound/soc/codecs/cs42l51.h create mode 100644 sound/soc/kirkwood/Kconfig create mode 100644 sound/soc/kirkwood/Makefile create mode 100644 sound/soc/kirkwood/kirkwood-dma.c create mode 100644 sound/soc/kirkwood/kirkwood-dma.h create mode 100644 sound/soc/kirkwood/kirkwood-i2s.c create mode 100644 sound/soc/kirkwood/kirkwood-i2s.h create mode 100644 sound/soc/kirkwood/kirkwood-openrd.c create mode 100644 sound/soc/kirkwood/kirkwood.h create mode 100644 sound/soc/nuc900/Kconfig create mode 100644 sound/soc/nuc900/Makefile create mode 100644 sound/soc/nuc900/nuc900-ac97.c create mode 100644 sound/soc/nuc900/nuc900-audio.c create mode 100644 sound/soc/nuc900/nuc900-auido.h create mode 100644 sound/soc/nuc900/nuc900-pcm.c

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[alsa-devel] [GIT PULL] ASoC updates for 2.6.36
by Liam Girdwood 31 May '10

31 May '10

Hi Takashi, Please pull the following ASoC updates for 2.6.36. Thanks Liam --- The following changes since commit ad8332c1302bcb4f80d593fd3eb477be9d7f5604: Jarkko Nikula (1): ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.36 Jorge Eduardo Candelaria (1): ASoC: TWL6040: Fix playback with 19.2 Mhz MCLK Peter Ujfalusi (8): ASoC: TWL4030: Revisit codec defaults ASoC: TWL4030: Remove wrapper for power down ASoC: TWL4030: Make offset cancellation path configurable ASoC: TWL4030: Optimize the power up sequence ASoC: TWL4030: Helper to check chip default registers ASoC: TWL4030: Correct the ARXR2_APGA_CTL chip default ASoC: TWL4030: Use BIAS_OFF instead of BIAS_STANDBY, when not in use ASoC: TWL4030: Add functionalty to reset the registers sound/soc/codecs/twl4030.c | 291 +++++++++++++++++++++++++------------------- sound/soc/codecs/twl4030.h | 3 + sound/soc/codecs/twl6040.c | 2 +- 3 files changed, 170 insertions(+), 126 deletions(-)

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[alsa-devel] alsa-driver-1.0.23 fails to build on ppc
by Elimar Riesebieter 31 May '10

31 May '10

Hi, preparing alsa-driver-1.0.23 for Debian upload I found while building modules all: LD [M] /usr/src/modass/usr_src/modules/alsa-driver/aoa/core/snd-aoa.o CC [M] /usr/src/modass/usr_src/modules/alsa-driver/aoa/fabrics/layout.o In file included from /usr/src/modass/usr_src/modules/alsa-driver/aoa/fabrics/layout.c:2: /usr/src/modass/usr_src/modules/alsa-driver/aoa/fabrics/../../alsa-kernel/aoa/fabrics/layout.c: In function 'check_codec': /usr/src/modass/usr_src/modules/alsa-driver/aoa/fabrics/../../alsa-kernel/aoa/fabrics/layout.c:772: error: 'struct device_node' has no member named 'phandle' make[8]: *** [/usr/src/modass/usr_src/modules/alsa-driver/aoa/fabrics/layout.o] Error 1 make[7]: *** [/usr/src/modass/usr_src/modules/alsa-driver/aoa/fabrics] Error 2 make[6]: *** [/usr/src/modass/usr_src/modules/alsa-driver/aoa] Error 2 make[5]: *** [_module_/usr/src/modass/usr_src/modules/alsa-driver] Error 2 make[4]: *** [sub-make] Error 2 make[3]: *** [all] Error 2 I found no fix in git. I would be glad if someone could give me a hint. Thanks Elimar -- Planung: Ersatz des Zufalls durch den Irrtum. -unknown-

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Re: [alsa-devel] [PATCH] alsa/Au1xxx-PSC: use resource_size
by Manuel Lauss 31 May '10

31 May '10

2010/5/28 Wan ZongShun <mcuos.com(a)gmail.com>: > Dear Manuel & Mark, > > I make a little modification to use the 'resource_size'. It's okay with me. You could also modifiy psc-ac97.c in the same way. Manuel Lauss

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[alsa-devel] [patch 0/6] kirkwood openrd client audio support - v4
by apatard＠mandriva.com 31 May '10

31 May '10

Hi, This patchset is an other try to get merged openrd audio support mainline. The main change is going back to using kirkwood instead of orion, as the only orion having i2s is not merged mainline (if I got it right). I've also fixed the lastests comments on the cs42l51 codec patch. Arnaud Changes : - changed back from orion to kirkwood - removed cs42l51 adc input register initialisation - moved the DAPM event macro to soc-dapm - fixed cs42l51_set_dai_fmt() master/slave check - removed cs42l51 startup message - use snd_soc_update_bits() in cs42l51_pdn_event()

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[alsa-devel] arecord and DMA misaligned error with device
by Ruchi Sirauthiya 31 May '10

31 May '10

Hi all, I am working on an OMAP board with Linux kernel 2.6.22.18-omap3 and ALSA. I got the following stack trace when running 'arecord' continuously for a long time (~30 minutes). What could be the reason for "page allocation failure" and "DMA misaligned error"? How can I solve this issue? Thanks and Regards, Ruchi Sirauthiya ---- LOG START ---- page allocation failure. order:5, mode:0xd1 [ 1249.046875] [<c0050380>] (dump_stack+0x0/0x14) from [<c00ae260>] (__alloc_pages+0x288/0x29c) [ 1249.054687] [<c00adfd8>] (__alloc_pages+0x0/0x29c) from [<c0051cac>] (__dma_alloc+0x17c/0x3d4) [ 1249.062500] [<c0051b30>] (__dma_alloc+0x0/0x3d4) from [<c0051f58>] (dma_alloc_coherent+0x24/0x30) [ 1249.070312] [<c0051f34>] (dma_alloc_coherent+0x0/0x30) from [<c02717b4>] (snd_omap_alsa_hw_params+0x54/0x7c) [ 1249.085937] [<c0271760>] (snd_omap_alsa_hw_params+0x0/0x7c) from [<c02683dc>] (snd_pcm_hw_params+0x90/0x2e0) [ 1249.093750] r5:c19d6000 r4:c15da000 [ 1249.093750] [<c026834c>] (snd_pcm_hw_params+0x0/0x2e0) from [<c02686b0>] (snd_pcm_hw_params_user+0x84/0xe4) [ 1249.109375] r9:cb8f4000 r8:c004c028 r7:c077e360 r6:c19d6000 r5:c077e360 [ 1249.109375] r4:001bdb48 [ 1249.117187] [<c026862c>] (snd_pcm_hw_params_user+0x0/0xe4) from [<c02689c0>] (snd_pcm_common_ioctl1+0x1cc/0xf04) [ 1249.125000] r7:c077e360 r6:001bdb48 r5:001bdb48 r4:c25c4111 [ 1249.132812] [<c02687f4>] (snd_pcm_common_ioctl1+0x0/0xf04) from [<c0269b44>] (snd_pcm_capture_ioctl1+0x44c/0x470) [ 1249.140625] [<c02696f8>] (snd_pcm_capture_ioctl1+0x0/0x470) from [<c0269b9c>] (snd_pcm_capture_ioctl+0x34/0x38) [ 1249.148437] [<c0269b68>] (snd_pcm_capture_ioctl+0x0/0x38) from [<c00d0adc>] (do_ioctl+0x34/0x78) [ 1249.164062] [<c00d0aa8>] (do_ioctl+0x0/0x78) from [<c00d0d80>] (vfs_ioctl+0x260/0x278) [ 1249.171875] r5:001bdb48 r4:ccb9a4e0 [ 1249.171875] [<c00d0b20>] (vfs_ioctl+0x0/0x278) from [<c00d0dd8>] (sys_ioctl+0x40/0x64) [ 1249.179687] r7:ccb9a4e0 r6:c25c4111 r5:001bdb48 r4:00000039 [ 1249.187500] [<c00d0d98>] (sys_ioctl+0x0/0x64) from [<c004be80>] (ret_fast_syscall+0x0/0x2c) [ 1249.195312] r7:00000036 r6:001ec5c8 r5:413b3bd4 r4:001bdb48 [ 1249.203125] Mem-info: [ 1249.203125] DMA per-cpu: [ 1249.203125] CPU 0: Hot: hi: 90, btch: 15 usd: 66 Cold: hi: 30, btch: 7 usd: 24 [ 1249.210937] Active:10183 inactive:35469 dirty:4 writeback:0 unstable:0 [ 1249.210937] free:739 slab:5167 mapped:2540 pagetables:125 bounce:0 [ 1249.226562] DMA free:2956kB min:1888kB low:2360kB high:2832kB active:40732kB inactive:141876kB present:223520kB pages_scanned:0 all_unreclaimable? no [ 1249.242187] lowmem_reserve[]: 0 0 [ 1249.242187] DMA: 13*4kB 33*8kB 27*16kB 3*32kB 31*64kB 1*128kB 0*256kB 0*512kB 0*1024kB 0*2048kB 0*4096kB 0*8192kB = 2956kB [ 1249.257812] Free swap: 0kB [ 1249.265625] 56320 pages of RAM [ 1249.273437] 1016 free pages [ 1249.273437] 3616 reserved pages [ 1249.273437] 5167 slab pages [ 1249.281250] 41756 pages shared [ 1249.281250] 0 pages swap cached [ 1249.343750] DMA misaligned error with device 1 [ 1249.343750] DMA misaligned error with device 1 [ 1249.351562] DMA misaligned error with device 1 [ 1249.359375] DMA misaligned error with device 1 [ 1249.359375] DMA misaligned error with device 1 [ 1249.367187] DMA misaligned error with device 1 [ 1249.367187] DMA misaligned error with device 1 [ 1249.375000] DMA misaligned error with device 1 [ 1249.375000] DMA misaligned error with device 1 [ 1249.382812] DMA misaligned error with device 1 [ 1249.390625] DMA misaligned error with device 1 [ 1249.390625] DMA misaligned error with device 1 [ 1249.398437] DMA misaligned error with device 1 [ 1249.398437] DMA misaligned error with device 1 [ 1249.406250] DMA misaligned error with device 1 [ 1249.406250] DMA misaligned error with device 1 ... ... log truncated ... ... ---- LOG END ----

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