April 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: fsl_ssi: Remove custom fsl_ssi_debugfs_create/remove()
by Fabio Estevam 08 Apr '15

08 Apr '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> Since commit 81c7cfd1b22a0 ("ASoC: Move debugfs registration to the component level") ASoC core code deals with registering debugfs, so we should remove the custom fsl_ssi_debugfs_create/remove() to avoid the following warning: fsl-ssi-dai 2028000.ssi: ASoC: Failed to create component debugfs directory Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 11 +---------- sound/soc/fsl/fsl_ssi.h | 13 ------------- sound/soc/fsl/fsl_ssi_dbg.c | 22 ---------------------- 3 files changed, 1 insertion(+), 45 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 6b0c8f7..b8cdb8f 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1411,10 +1411,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) } } - ret = fsl_ssi_debugfs_create(&ssi_private->dbg_stats, &pdev->dev); - if (ret) - goto error_irq; - /* * If codec-handle property is missing from SSI node, we assume * that the machine driver uses new binding which does not require @@ -1441,7 +1437,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (IS_ERR(ssi_private->pdev)) { ret = PTR_ERR(ssi_private->pdev); dev_err(&pdev->dev, "failed to register platform: %d\n", ret); - goto error_sound_card; + goto error_irq; } done: @@ -1451,9 +1447,6 @@ done: return 0; -error_sound_card: - fsl_ssi_debugfs_remove(&ssi_private->dbg_stats); - error_irq: snd_soc_unregister_component(&pdev->dev); @@ -1468,8 +1461,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) { struct fsl_ssi_private *ssi_private = dev_get_drvdata(&pdev->dev); - fsl_ssi_debugfs_remove(&ssi_private->dbg_stats); - if (ssi_private->pdev) platform_device_unregister(ssi_private->pdev); snd_soc_unregister_component(&pdev->dev); diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 5065105..4767df1 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -241,10 +241,6 @@ struct fsl_ssi_dbg { void fsl_ssi_dbg_isr(struct fsl_ssi_dbg *ssi_dbg, u32 sisr); -int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev); - -void fsl_ssi_debugfs_remove(struct fsl_ssi_dbg *ssi_dbg); - #else struct fsl_ssi_dbg { @@ -254,15 +250,6 @@ static inline void fsl_ssi_dbg_isr(struct fsl_ssi_dbg *stats, u32 sisr) { } -static inline int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, - struct device *dev) -{ - return 0; -} - -static inline void fsl_ssi_debugfs_remove(struct fsl_ssi_dbg *ssi_dbg) -{ -} #endif /* ! IS_ENABLED(CONFIG_DEBUG_FS) */ #endif diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c index 5469ffb..118064c 100644 --- a/sound/soc/fsl/fsl_ssi_dbg.c +++ b/sound/soc/fsl/fsl_ssi_dbg.c @@ -139,25 +139,3 @@ static const struct file_operations fsl_ssi_stats_ops = { .llseek = seq_lseek, .release = single_release, }; - -int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev) -{ - ssi_dbg->dbg_dir = debugfs_create_dir(dev_name(dev), NULL); - if (!ssi_dbg->dbg_dir) - return -ENOMEM; - - ssi_dbg->dbg_stats = debugfs_create_file("stats", S_IRUGO, - ssi_dbg->dbg_dir, ssi_dbg, &fsl_ssi_stats_ops); - if (!ssi_dbg->dbg_stats) { - debugfs_remove(ssi_dbg->dbg_dir); - return -ENOMEM; - } - - return 0; -} - -void fsl_ssi_debugfs_remove(struct fsl_ssi_dbg *ssi_dbg) -{ - debugfs_remove(ssi_dbg->dbg_stats); - debugfs_remove(ssi_dbg->dbg_dir); -} -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Intel: remove unused functions
by Sudip Mukherjee 08 Apr '15

08 Apr '15

these functions were never called by anyone. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/soc/intel/atom/sst/sst_pvt.c | 24 ------------------------ 1 file changed, 24 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index 2d74249..adb32fe 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -111,30 +111,6 @@ int sst_wait_interruptible(struct intel_sst_drv *sst_drv_ctx, } -static unsigned long long read_shim_data(struct intel_sst_drv *sst, int addr) -{ - unsigned long long val = 0; - - switch (sst->dev_id) { - case SST_MRFLD_PCI_ID: - case SST_BYT_ACPI_ID: - val = sst_shim_read64(sst->shim, addr); - break; - } - return val; -} - -static void write_shim_data(struct intel_sst_drv *sst, int addr, - unsigned long long data) -{ - switch (sst->dev_id) { - case SST_MRFLD_PCI_ID: - case SST_BYT_ACPI_ID: - sst_shim_write64(sst->shim, addr, (u64) data); - break; - } -} - /* * sst_wait_timeout - wait on event for timeout * -- 1.8.1.2

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[alsa-devel] [PATCH] ALSA: bebob: fix to processing in big-endian machine for sending cue
by Takashi Sakamoto 08 Apr '15

08 Apr '15

Some M-Audio devices require to receive bootup command just after powering on, while codes in BeBoB driver doesn't work properly in big-endian machine because the command should be aligned by little-endian. This commit fixes this bug. This fix should go to stable kernel. Cc: Takayuki Shiroma <t.shiroma.oki(a)gmail.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_maudio.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c index a422aaa..9ee25a6 100644 --- a/sound/firewire/bebob/bebob_maudio.c +++ b/sound/firewire/bebob/bebob_maudio.c @@ -96,10 +96,10 @@ int snd_bebob_maudio_load_firmware(struct fw_unit *unit) struct fw_device *device = fw_parent_device(unit); int err, rcode; u64 date; - __be32 cues[3] = { - MAUDIO_BOOTLOADER_CUE1, - MAUDIO_BOOTLOADER_CUE2, - MAUDIO_BOOTLOADER_CUE3 + __le32 cues[3] = { + cpu_to_le32(MAUDIO_BOOTLOADER_CUE1), + cpu_to_le32(MAUDIO_BOOTLOADER_CUE2), + cpu_to_le32(MAUDIO_BOOTLOADER_CUE3) }; /* check date of software used to build */ -- 2.1.0

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[alsa-devel] [PATCH 1/2] ALSA: hda - Create AFG sysfs node at last
by Takashi Iwai 08 Apr '15

08 Apr '15

... so that user-space can know that the whole nodes have been created. Unfortunately, this can't be implemented easily in race-free way, so it's a kind of compromise. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_sysfs.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c index b358d5157802..18aea43d230d 100644 --- a/sound/hda/hdac_sysfs.c +++ b/sound/hda/hdac_sysfs.c @@ -313,12 +313,12 @@ static void widget_tree_free(struct hdac_device *codec) if (!tree) return; + free_widget_node(tree->afg, &widget_afg_group); if (tree->nodes) { for (p = tree->nodes; *p; p++) free_widget_node(*p, &widget_node_group); kfree(tree->nodes); } - free_widget_node(tree->afg, &widget_afg_group); if (tree->root) kobject_put(tree->root); kfree(tree); @@ -362,13 +362,6 @@ static int widget_tree_create(struct hdac_device *codec) if (!tree->root) return -ENOMEM; - if (codec->afg) { - err = add_widget_node(tree->root, codec->afg, - &widget_afg_group, &tree->afg); - if (err < 0) - return err; - } - tree->nodes = kcalloc(codec->num_nodes + 1, sizeof(*tree->nodes), GFP_KERNEL); if (!tree->nodes) @@ -381,6 +374,13 @@ static int widget_tree_create(struct hdac_device *codec) return err; } + if (codec->afg) { + err = add_widget_node(tree->root, codec->afg, + &widget_afg_group, &tree->afg); + if (err < 0) + return err; + } + kobject_uevent(tree->root, KOBJ_CHANGE); return 0; } -- 2.3.4

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[alsa-devel] [PATCH v2] ASoC: max98090: add shutdown callback for max98090
by Caesar Wang 08 Apr '15

08 Apr '15

To fix pop noise when shutdown,the pop noise during shutdown is the pmic cutoff power of codec without any notice. Signed-off-by: jay.xu <xjq(a)rock-chips.com> Signed-off-by: zhengxing <zhengxing(a)rock-chips.com> Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> Serien-cc: linux-kernel(a)vger.kernel.org Serien-cc: devicetree(a)vger.kernel.org Serien-cc: dianders(a)chromium.org --- Changes in v2: - remove the dev_info(..) - fix the comment style - add the max98090_i2c_shutdown() in remove() fuction sound/soc/codecs/max98090.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index b112b1c..e59013a 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2607,10 +2607,26 @@ err_enable: static int max98090_i2c_remove(struct i2c_client *client) { + max98090_i2c_shutdown(client); snd_soc_unregister_codec(&client->dev); return 0; } +static void max98090_i2c_shutdown(struct i2c_client *i2c) +{ + struct max98090_priv *max98090 = dev_get_drvdata(&i2c->dev); + + /* + * Enable volume smoothing, disable zero cross. This will cause + * a quick 40ms ramp to mute on shutdown. + */ + regmap_write(max98090->regmap, + M98090_REG_LEVEL_CONTROL, M98090_VSENN_MASK); + regmap_write(max98090->regmap, + M98090_REG_DEVICE_SHUTDOWN, 0x00); + msleep(40); +} + #ifdef CONFIG_PM static int max98090_runtime_resume(struct device *dev) { @@ -2697,6 +2713,7 @@ static struct i2c_driver max98090_i2c_driver = { }, .probe = max98090_i2c_probe, .remove = max98090_i2c_remove, + .shutdown = max98090_i2c_shutdown, .id_table = max98090_i2c_id, }; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Intel: Fix a buffer overflow issue
by Jie Yang 08 Apr '15

08 Apr '15

0day robot reported a buffer overflow issue: ... sound/soc/intel/haswell/sst-haswell-pcm.c:1107 hsw_pcm_probe() error: buffer\ overflow 'hsw_dais' 4 <= 4 sound/soc/intel/haswell/sst-haswell-pcm.c:1109 hsw_pcm_probe() error: buffer\ overflow 'hsw_dais' 4 <= 4 ... Fix it by initializing the index(i) to correct value. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/haswell/sst-haswell-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 157b3a6..23ae040 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -1103,7 +1103,7 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) return 0; err: - for (;i >= 0; i--) { + for (--i; i >= 0; i--) { if (hsw_dais[i].playback.channels_min) snd_dma_free_pages(&priv_data->dmab[i][0]); if (hsw_dais[i].capture.channels_min) -- 1.9.1

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[alsa-devel] [PATCH 0/1] ASoC: max98090: add shutdown callback for max98090
by Caesar Wang 08 Apr '15

08 Apr '15

Tested on veyron devices,play music then shutdown device,here carefully to speaker or headphone. Caesar Wang (1): ASoC: max98090: add shutdown callback for max98090 sound/soc/codecs/max98090.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) -- 1.9.1

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Re: [alsa-devel] Add delay in alc283_init
by Takashi Iwai 08 Apr '15

08 Apr '15

At Wed, 8 Apr 2015 08:56:48 +0000, Kailang wrote: > > Hi Takashi, > > This patch will make more stable for getting pin sense during resume state. > This is tested for Chrome Box machine. > Many Thanks. Thanks, applied now with Cc to stable. Takashi > > BR, > Kailang > [1.2 <text/html; us-ascii (quoted-printable)>] > > [2 0000-add-delay-before-get-pin-sense.patch <application/octet-stream (base64)>] >

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[alsa-devel] 1st draft: ALSA Json Gateway and HTML5 UI
by Fulup Ar Foll 07 Apr '15

07 Apr '15

Hi, I would need some help to finish a json version of amixer. Porting cget is done, but for cset I have an issue with "snd_hctl_find_elem (handle, id)". I'm not sure what's the right method to pass it. I've tried every options I could imagine. But if it works on Mint/32bit, it miserably fail on OpenSuse/64. Furthermore amixer.c as this strange comment /* FIXME: Remove it when hctl find works ok !!! */ and they do not use snd_hctl_find_elem. I posted a small C code to highlight the issue at [1]https://gist.github.com/fulup-bzh/891e5627be325150337f My question is simple, did I miss something ? should I move to lower level API ? Note: While the painting for my json-gateway is not dry, I nevertheless pushed a initial running version with cget at [2]https://github.com/fulup-bzh/AlsaJsonGateway I also have a small HTML5/UI at [3]http://breizhme.net/alsajson/mixers/ajg#/scarlett?card=2 While UI code is very dependant on each soundcard [in my case Scarlett/Focurite] the Json-Gateway is 100% independent of the board. If anything one have an idea, please let me know. Fulup References 1. https://gist.github.com/fulup-bzh/891e5627be325150337f 2. https://github.com/fulup-bzh/AlsaJsonGateway 3. http://breizhme.net/alsajson/mixers/ajg#/scarlett?card=2

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[alsa-devel] [PATCH][ASoC]Add ability to remove rate constraints from generic ASoC AC'97 CODEC driver
by Maciej S. Szmigiero 07 Apr '15

07 Apr '15

Add ability to remove rate constraints from generic ASoC AC'97 CODEC driver via passed platform data, make it selectable in config. This way this driver can be used for platforms which don't need specialized AC'97 CODEC drivers while at the same avoiding code duplication from implementing equivalent functionality in a controller driver. There is no change in behavior when no platform data is passed. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> diff --git a/include/sound/soc-ac97.h b/include/sound/soc-ac97.h new file mode 100644 index 0000000..ceb4e2f --- /dev/null +++ b/include/sound/soc-ac97.h @@ -0,0 +1,17 @@ +/* + * Platform data for generic ASoC AC97 CODEC driver. + * + * 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 __LINUX_SND__SOC_AC97_H +#define __LINUX_SND__SOC_AC97_H + +struct snd_soc_ac97_codec_platform_data { + bool playback_rate_constrained; + bool capture_rate_constrained; +}; + +#endif /* __LINUX_SND__SOC_AC97_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0bddd92..92d6d39 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,7 +16,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 select SND_SOC_AB8500_CODEC if ABX500_CORE - select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS + select SND_SOC_AC97_CODEC select SND_SOC_AD1836 if SPI_MASTER select SND_SOC_AD193X_SPI if SPI_MASTER select SND_SOC_AD193X_I2C if I2C @@ -210,8 +210,9 @@ config SND_SOC_AB8500_CODEC tristate config SND_SOC_AC97_CODEC - tristate + tristate "Build generic ASoC AC97 CODEC driver" select SND_AC97_CODEC + select SND_SOC_AC97_BUS config SND_SOC_AD1836 tristate diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index d0ac723..c3fb845 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -20,9 +20,16 @@ #include <sound/core.h> #include <sound/pcm.h> #include <sound/ac97_codec.h> +#include <sound/soc-ac97.h> #include <sound/initval.h> #include <sound/soc.h> +struct ac97_private { + struct snd_ac97 *ac97; + bool playback_rate_constrained; + bool capture_rate_constrained; +}; + static const struct snd_soc_dapm_widget ac97_widgets[] = { SND_SOC_DAPM_INPUT("RX"), SND_SOC_DAPM_OUTPUT("TX"), @@ -33,22 +40,53 @@ static const struct snd_soc_dapm_route ac97_routes[] = { { "TX", NULL, "AC97 Playback" }, }; +static const unsigned int default_rates[] = { + 8000, 11025, 22050, 44100, 48000 +}; + +static const struct snd_pcm_hw_constraint_list default_rate_constraints = { + .count = ARRAY_SIZE(default_rates), + .list = default_rates, +}; + +static int ac97_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ac97_private *ac97 = snd_soc_codec_get_drvdata(codec); + bool constrained; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + constrained = ac97->playback_rate_constrained; + else + constrained = ac97->capture_rate_constrained; + + if (!constrained) + return 0; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &default_rate_constraints); + else + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &default_rate_constraints); + + return 0; +} + static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + struct ac97_private *ac97 = snd_soc_codec_get_drvdata(codec); int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; - return snd_ac97_set_rate(ac97, reg, substream->runtime->rate); + return snd_ac97_set_rate(ac97->ac97, reg, substream->runtime->rate); } -#define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000) - static const struct snd_soc_dai_ops ac97_dai_ops = { + .startup = ac97_startup, .prepare = ac97_prepare, }; @@ -58,20 +96,21 @@ static struct snd_soc_dai_driver ac97_dai = { .stream_name = "AC97 Playback", .channels_min = 1, .channels_max = 2, - .rates = STD_AC97_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = SND_SOC_STD_AC97_FMTS,}, .capture = { .stream_name = "AC97 Capture", .channels_min = 1, .channels_max = 2, - .rates = STD_AC97_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = SND_SOC_STD_AC97_FMTS,}, .ops = &ac97_dai_ops, }; static int ac97_soc_probe(struct snd_soc_codec *codec) { - struct snd_ac97 *ac97; + struct ac97_private *ac97 = snd_soc_codec_get_drvdata(codec); + struct snd_ac97_bus *ac97_bus; struct snd_ac97_template ac97_template; int ret; @@ -83,31 +122,28 @@ static int ac97_soc_probe(struct snd_soc_codec *codec) return ret; memset(&ac97_template, 0, sizeof(struct snd_ac97_template)); - ret = snd_ac97_mixer(ac97_bus, &ac97_template, &ac97); + ret = snd_ac97_mixer(ac97_bus, &ac97_template, &ac97->ac97); if (ret < 0) return ret; - snd_soc_codec_set_drvdata(codec, ac97); - return 0; } #ifdef CONFIG_PM static int ac97_soc_suspend(struct snd_soc_codec *codec) { - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + struct ac97_private *ac97 = snd_soc_codec_get_drvdata(codec); - snd_ac97_suspend(ac97); + snd_ac97_suspend(ac97->ac97); return 0; } static int ac97_soc_resume(struct snd_soc_codec *codec) { + struct ac97_private *ac97 = snd_soc_codec_get_drvdata(codec); - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); - - snd_ac97_resume(ac97); + snd_ac97_resume(ac97->ac97); return 0; } @@ -129,6 +165,27 @@ static struct snd_soc_codec_driver soc_codec_dev_ac97 = { static int ac97_probe(struct platform_device *pdev) { + struct ac97_private *ac97 = + devm_kzalloc(&pdev->dev, sizeof(struct ac97_private), + GFP_KERNEL); + struct snd_soc_ac97_codec_platform_data *pdata = + pdev->dev.platform_data; + + if (!ac97) + return -ENOMEM; + + if (pdata) { + ac97->playback_rate_constrained = + pdata->playback_rate_constrained; + ac97->capture_rate_constrained = + pdata->capture_rate_constrained; + } else { + ac97->playback_rate_constrained = 1; + ac97->capture_rate_constrained = 1; + } + + platform_set_drvdata(pdev, ac97); + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_ac97, &ac97_dai, 1); }

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