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

[alsa-devel] [PATCH v2] ASoC: fsl_ssi: Do not create debugfs directory
by Fabio Estevam 09 Apr '15

09 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 not be creating the debugfs directory locally in order 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> --- Changes since v1: - Only remove dir creation/remove as suggested by Mark Brown sound/soc/fsl/fsl_ssi.h | 1 - sound/soc/fsl/fsl_ssi_dbg.c | 13 +++---------- 2 files changed, 3 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 5065105..68aef90 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -211,7 +211,6 @@ struct device; #if IS_ENABLED(CONFIG_DEBUG_FS) struct fsl_ssi_dbg { - struct dentry *dbg_dir; struct dentry *dbg_stats; struct { diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c index 5469ffb..46cea84 100644 --- a/sound/soc/fsl/fsl_ssi_dbg.c +++ b/sound/soc/fsl/fsl_ssi_dbg.c @@ -142,22 +142,15 @@ static const struct file_operations fsl_ssi_stats_ops = { 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) + ssi_dbg->dbg_stats = debugfs_create_file("stats", S_IRUGO, NULL, + ssi_dbg, &fsl_ssi_stats_ops); + if (!ssi_dbg->dbg_stats) 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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Re: [alsa-devel] [PATCH] ALSA: hda - make power_save threshold per-codec
by Takashi Iwai 09 Apr '15

09 Apr '15

At Wed, 8 Apr 2015 23:24:52 -0700, Matthew Garrett wrote: > > On Wed, Apr 8, 2015 at 10:20 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > At Wed, 8 Apr 2015 18:53:48 -0700, > > Matthew Garrett wrote: > > > > > > Modern hardware will often have multiple HDA devices, and the desired > > > power saving configuration may vary depending on the codecs attached to > > > each of them. Push the power_save value down to the individual codec > > > structures, keeping the module parameter as a global control mechanism > > > for compatibility purposes. > > > > This patch doesn't apply any longer at all to the recent code, since > > the power saving stuff has been already translated to the standard > > runtime PM. Take a look at linux-next tree. > > Ah! Wonderful! > > > Also we can't break the current behavior controlling the power save > > via option. Many programs rely on this mechanism, thus changing this > > would break its behavior. > > Yeah, I kept that behaviour - anything that touches > /sys/module/snd_hda_intel/parameters/power_save just changes the > settings for all codecs. > > > In the latest code, the common control via power_save module option is > > still there for compatibility, but you can also adjust the value for > > each codec in runtime attribute of each codec sysfs in addition. > > That's ideal - I'll pull down next and rewrite my udev rules. Thanks! Let me know if you find anything broken with the new code. It's pretty new, so any intensive testing is appreciated. thanks, Takashi

1 0

[alsa-devel] [PATCH] ALSA: hda - Always allow access for POWER_STATE verbs
by Takashi Iwai 09 Apr '15

09 Apr '15

The hdac regmap code checks whether the codec is powered on while accessing, but there must be an exception -- the verbs to control the power state. Currently HD-audio driver doesn't access them via regmap, so this patch doesn't fix any current behavior, but it's just for future. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_regmap.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 64876fa357c9..51f1b5c8a91c 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -239,7 +239,7 @@ static int hda_reg_read(void *context, unsigned int reg, unsigned int *val) int verb = get_verb(reg); int err; - if (!codec_is_running(codec)) + if (!codec_is_running(codec) && verb != AC_VERB_GET_POWER_STATE) return -EAGAIN; reg |= (codec->addr << 28); if (is_stereo_amp_verb(reg)) @@ -265,16 +265,16 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) unsigned int verb; int i, bytes, err; - if (!codec_is_running(codec)) - return codec->lazy_cache ? 0 : -EAGAIN; - reg &= ~0x00080000U; /* drop GET bit */ reg |= (codec->addr << 28); + verb = get_verb(reg); + + if (!codec_is_running(codec) && verb != AC_VERB_SET_POWER_STATE) + return codec->lazy_cache ? 0 : -EAGAIN; if (is_stereo_amp_verb(reg)) return hda_reg_write_stereo_amp(codec, reg, val); - verb = get_verb(reg); if (verb == AC_VERB_SET_PROC_COEF) return hda_reg_write_coef(codec, reg, val); -- 2.3.5

1 0

[alsa-devel] [PATCH 0/5 alsa-lib] amixer fixes for enumerated elements
by Takashi Sakamoto 09 Apr '15

09 Apr '15

The amixer still has issues according to sset operation to enumerated items. The issues are: - Fail to handle omitted value of each channel. - The name of enumerated element is truncated. - Unclear separators for values of channels. This patchset fixes these issues, including a small code refactoring for better reading. Takashi Sakamoto (5): amixer: gather local variables in the beginning of functions amixer: fix a bug to handle omitted value in parameter for enumerated element amixer: expand local storage for item name according to kernel code amixer: arrange validation logic amixer: use the same characters for separator. amixer/amixer.c | 60 ++++++++++++++++++++++++++++++++++++++++----------------- 1 file changed, 42 insertions(+), 18 deletions(-) -- 2.1.0

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Re: [alsa-devel] Audio Jack Out does not work
by Dan Carpenter 09 Apr '15

09 Apr '15

So it's 03ad6a8c93b6df2 ('ALSA: hda - Fix "PCM" name being used on one DAC when there are two DACs') which causes the problem? Have you tried to just revert that patch? git show 03ad6a8c93b6df2d65c305b5b5f9474068b45bfb | patch -p1 -R regards, dan carpenter

4 13

[alsa-devel] [PATCH] ALSA: hda - make power_save threshold per-codec
by Matthew Garrett 09 Apr '15

09 Apr '15

Modern hardware will often have multiple HDA devices, and the desired power saving configuration may vary depending on the codecs attached to each of them. Push the power_save value down to the individual codec structures, keeping the module parameter as a global control mechanism for compatibility purposes. Signed-off-by: Matthew Garrett <mjg59(a)coreos.com> --- Documentation/sound/alsa/HD-Audio.txt | 5 +++-- sound/pci/hda/hda_codec.c | 28 ++++++++++++++++++++-------- sound/pci/hda/hda_codec.h | 19 ++++--------------- sound/pci/hda/hda_controller.c | 6 +++--- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 7 +++---- sound/pci/hda/hda_sysfs.c | 30 ++++++++++++++++++++++++++++++ sound/pci/hda/hda_tegra.c | 4 ++-- 8 files changed, 66 insertions(+), 35 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 42a0a39..99f2fb8 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -569,8 +569,9 @@ Power-Saving The power-saving is a kind of auto-suspend of the device. When the device is inactive for a certain time, the device is automatically turned off to save the power. The time to go down is specified via -`power_save` module option, and this option can be changed dynamically -via sysfs. +`power_save` module option. This option can be changed dynamically +via sysfs, either globally (via /sys/module/snd_hda_*/parameters/power_save) +or per-codec (via the power_save sysfs parameter in the codec directory) The power-saving won't work when the analog loopback is enabled on some codecs. Make sure that you mute all unneeded signal routes when diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2fe86d2..5e8394a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -895,7 +895,6 @@ int snd_hda_bus_new(struct snd_card *card, bus->private_data = temp->private_data; bus->pci = temp->pci; bus->modelname = temp->modelname; - bus->power_save = temp->power_save; bus->ops = temp->ops; mutex_init(&bus->cmd_mutex); @@ -1436,8 +1435,9 @@ static void snd_hda_codec_dev_release(struct device *dev) * Returns 0 if successful, or a negative error code. */ int snd_hda_codec_new(struct hda_bus *bus, - unsigned int codec_addr, - struct hda_codec **codecp) + unsigned int codec_addr, + struct hda_codec **codecp, + int power_save) { struct hda_codec *codec; char component[31]; @@ -1495,6 +1495,7 @@ int snd_hda_codec_new(struct hda_bus *bus, INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); codec->depop_delay = -1; + codec->power_save = power_save; codec->fixup_id = HDA_FIXUP_ID_NOT_SET; #ifdef CONFIG_PM @@ -5112,23 +5113,34 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) codec->power_transition = 0; } -#define power_save(codec) \ - ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) - /* Transition to powered down */ static void __snd_hda_power_down(struct hda_codec *codec) { if (!codec->power_on || codec->power_count || codec->power_transition) return; - if (power_save(codec)) { + if (codec->power_save >= 0) { codec->power_transition = -1; /* avoid reentrance */ queue_delayed_work(codec->bus->workq, &codec->power_work, - msecs_to_jiffies(power_save(codec) * 1000)); + msecs_to_jiffies(codec->power_save * 1000)); } } /** + * + * snd_hda_power_save_set - set the power-down time for the codec + * + * @codec: HD-audio codec + * @value: seconds to wait before powering down the codec + */ +void snd_hda_power_save_set(struct hda_codec *codec, int value) +{ + codec->power_save = value; + snd_hda_power_save(codec, 0, false); +} +EXPORT_SYMBOL_GPL(snd_hda_power_save_set); + +/** * snd_hda_power_save - Power-up/down/sync the codec * @codec: HD-audio codec * @delta: the counter delta to change diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 9c8820f..faaab58 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -106,7 +106,6 @@ struct hda_bus_template { void *private_data; struct pci_dev *pci; const char *modelname; - int *power_save; struct hda_bus_ops ops; }; @@ -123,7 +122,6 @@ struct hda_bus { void *private_data; struct pci_dev *pci; const char *modelname; - int *power_save; struct hda_bus_ops ops; /* codec linked list */ @@ -398,6 +396,7 @@ struct hda_codec { struct snd_array jacks; #endif + int power_save; int depop_delay; /* depop delay in ms, -1 for default delay time */ /* fix-up list */ @@ -423,7 +422,7 @@ enum { int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, struct hda_bus **busp); int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, - struct hda_codec **codecp); + struct hda_codec **codecp, int power_save); int snd_hda_codec_configure(struct hda_codec *codec); int snd_hda_codec_update_widgets(struct hda_codec *codec); @@ -588,9 +587,11 @@ const char *snd_hda_get_jack_location(u32 cfg); * power saving */ #ifdef CONFIG_PM +void snd_hda_power_save_set(struct hda_codec *codec, int value); void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait); void snd_hda_update_power_acct(struct hda_codec *codec); #else +static inline void snd_hda_power_save_set(struct hda_codec *codec, int value) {} static inline void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait) {} #endif @@ -634,18 +635,6 @@ static inline void snd_hda_power_down(struct hda_codec *codec) snd_hda_power_save(codec, -1, false); } -/** - * snd_hda_power_sync - Synchronize the power-save status - * @codec: HD-audio codec - * - * Synchronize the actual power state with the power account; - * called when power_save parameter is changed - */ -static inline void snd_hda_power_sync(struct hda_codec *codec) -{ - snd_hda_power_save(codec, 0, false); -} - #ifdef CONFIG_SND_HDA_PATCH_LOADER /* * patch firmware diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 17c2637..4bb13ae 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1813,7 +1813,7 @@ static int get_jackpoll_interval(struct azx *chip) /* Codec initialization */ int azx_codec_create(struct azx *chip, const char *model, unsigned int max_slots, - int *power_save_to) + int power_save) { struct hda_bus_template bus_temp; int c, codecs, err; @@ -1827,7 +1827,6 @@ int azx_codec_create(struct azx *chip, const char *model, bus_temp.ops.attach_pcm = azx_attach_pcm_stream; bus_temp.ops.bus_reset = azx_bus_reset; #ifdef CONFIG_PM - bus_temp.power_save = power_save_to; bus_temp.ops.pm_notify = azx_power_notify; #endif #ifdef CONFIG_SND_HDA_DSP_LOADER @@ -1886,7 +1885,8 @@ int azx_codec_create(struct azx *chip, const char *model, for (c = 0; c < max_slots; c++) { if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) { struct hda_codec *codec; - err = snd_hda_codec_new(chip->bus, c, &codec); + err = snd_hda_codec_new(chip->bus, c, &codec, + power_save); if (err < 0) continue; codec->jackpoll_interval = get_jackpoll_interval(chip); diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index c90d10f..d187121 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -45,7 +45,7 @@ irqreturn_t azx_interrupt(int irq, void *dev_id); /* Codec interface */ int azx_codec_create(struct azx *chip, const char *model, unsigned int max_slots, - int *power_save_to); + int power_save); int azx_codec_configure(struct azx *chip); int azx_mixer_create(struct azx *chip); int azx_init_stream(struct azx *chip); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a8a1e14..d51779a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -174,7 +174,6 @@ static struct kernel_param_ops param_ops_xint = { #define param_check_xint param_check_int static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; -static int *power_save_addr = &power_save; module_param(power_save, xint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " "(in second, 0 = disable)."); @@ -187,7 +186,7 @@ static bool power_save_controller = 1; module_param(power_save_controller, bool, 0644); MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode."); #else -static int *power_save_addr; +static int power_save = -1; #endif /* CONFIG_PM */ static int align_buffer_size = -1; @@ -754,7 +753,7 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) if (!chip->bus || chip->disabled) continue; list_for_each_entry(c, &chip->bus->codec_list, list) - snd_hda_power_sync(c); + snd_hda_power_save_set(c, power_save); } mutex_unlock(&card_list_lock); return 0; @@ -1898,7 +1897,7 @@ static int azx_probe_continue(struct azx *chip) /* create codec instances */ err = azx_codec_create(chip, model[dev], azx_max_codecs[chip->driver_type], - power_save_addr); + power_save); if (err < 0) goto out_free; diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index ccc962a..69bbb0b 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -44,8 +44,37 @@ static ssize_t power_off_acct_show(struct device *dev, return sprintf(buf, "%u\n", jiffies_to_msecs(codec->power_off_acct)); } +static ssize_t power_save_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct hda_codec *codec = dev_get_drvdata(dev); + + return sprintf(buf, "%d\n", jiffies_to_msecs(codec->power_save)); +} + +static ssize_t power_save_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct hda_codec *codec = dev_get_drvdata(dev); + unsigned long val; + int err = kstrtol(buf, 0, &val); + + if (err < 0) + return err; + + if (val < -1) + return -EINVAL; + + snd_hda_power_save_set(codec, val); + return count; +} + static DEVICE_ATTR_RO(power_on_acct); static DEVICE_ATTR_RO(power_off_acct); +static DEVICE_ATTR_RW(power_save); + #endif /* CONFIG_PM */ #define CODEC_INFO_SHOW(type) \ @@ -759,6 +788,7 @@ static struct attribute *hda_dev_attrs[] = { #ifdef CONFIG_PM &dev_attr_power_on_acct.attr, &dev_attr_power_off_acct.attr, + &dev_attr_power_save.attr, #endif #ifdef CONFIG_SND_HDA_RECONFIG &dev_attr_init_verbs.attr, diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 375e94f..69b76fc 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -82,7 +82,7 @@ module_param(power_save, bint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout (in seconds, 0 = disable)."); #else -static int power_save = 0; +static int power_save = -1; #endif /* @@ -503,7 +503,7 @@ static int hda_tegra_probe(struct platform_device *pdev) goto out_free; /* create codec instances */ - err = azx_codec_create(chip, NULL, 0, &power_save); + err = azx_codec_create(chip, NULL, 0, power_save); if (err < 0) goto out_free; -- 2.3.4

2 1

[alsa-devel] [PATCH 1/4] ASoC: tegra_alc5632: Use card DAPM context to access widgets
by Lars-Peter Clausen 08 Apr '15

08 Apr '15

The dapm field of the snd_soc_codec struct will eventually be removed (replaced with the DAPM context from the component embedded inside the CODEC). Replace its usage with the card's DAPM context. The idea is that DAPM is hierarchical and with the card at the root it is possible to access widgets from other contexts through the card context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/tegra/tegra_alc5632.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 6dcd06a..ba272e2 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -101,9 +101,6 @@ static const struct snd_kcontrol_new tegra_alc5632_controls[] = { static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(rtd->card); snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET, @@ -118,7 +115,7 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) &tegra_alc5632_hp_jack_gpio); } - snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin(&rtd->card->dapm, "MICBIAS1"); return 0; } -- 1.8.0

2 4

[alsa-devel] [PATCH v4 0/5] ALSA: jack: Refactoring for jack kctls
by Jie Yang 08 Apr '15

08 Apr '15

Currently only hda will create kctls for hda jacks, for asoc, people may need create jack kctls in specific driver. Here we are introducing kctls for each jack, by creating kctls and add them to jack controls list (considering exist of combo jack). At the same time, we will report events for each control in the list. With this new implementation, both HDA and soc jack kctls works fine, they are compatible for old HDA jack kctls, too. For soc, a. snd_jack_new() with NULL jack_kctl; b. for each pin, call snd_jack_add_kctls() to add one kctl(it will call snd_ctl_add() to add kctl to card); For hda, a. call snd_jack_kctl_new()(in __snd_hda_jack_add_kctl()) to new a jack_kctl(also new kctl); and use snd_hda_ctl_add to add kctl to card; b. new a hda_jack_tbl, appoint jack_tbl->jack_kctl = jack_kctl; c. for non-phantom jack, call snd_jack_new() with this jack_kctl, then the jack_kctl will be added to the new created jack; Changes in v4: 1. use snd_ctl_find_id() to get avaliable index; 2. add initial kctl for snd_jack_new(compatible for HDA); 3. add struct snd_jack_kctl * field to struct hda_jack_tbl; 4. new kctls for soc jack during jack pins creating. 5. add a patch to remove exporting snd_kctl_jack_new(). Changes in v3: 1. replace bit index with bit mask in jack_kctl; 2. add exception for SND_JACK_HEADSET and SND_JACK_AVOUT, only create one jack kctl for these two combo jacks, respectively. 3. add NULL check, mem kfree, and fix some potential risk. Change in v2: 1. define jack_kctl struct, and put jack kctl related stuff there; 2. add a patch to remove the existing controls for HDA jack. Jie Yang (5): ALSA: jack: create jack kcontrols for every jack input ALSA: jack: add a parameter to pass kctl for snd_jack_new ALSA: hda - Update to use the new jack kctls method ASoC: jack: create kctls according to jack pins info ALSA: jack: remove export snd_kctl_jack_new() include/sound/jack.h | 25 ++++++- sound/core/ctljack.c | 1 - sound/core/jack.c | 152 +++++++++++++++++++++++++++++++++++++++- sound/pci/hda/hda_jack.c | 52 +++++++------- sound/pci/hda/hda_jack.h | 4 +- sound/pci/oxygen/xonar_wm87x6.c | 2 +- sound/soc/soc-jack.c | 3 +- 7 files changed, 200 insertions(+), 39 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rt5645: Adds push button support for rt5650
by Bard Liao 08 Apr '15

08 Apr '15

This patch exports rt5650_set_button_detect function for machine driver to support rt5650 headset button detection function. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 147 +++++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/rt5645.h | 7 ++- 2 files changed, 146 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 69528ae..e547a994 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2368,6 +2368,8 @@ static int rt5645_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, static int rt5645_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_PREPARE: if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { @@ -2398,8 +2400,9 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: snd_soc_write(codec, RT5645_DEPOP_M2, 0x1100); - snd_soc_update_bits(codec, RT5645_GEN_CTRL1, - RT5645_DIG_GATE_CTRL, 0); + if (!rt5645->en_button_func) + snd_soc_update_bits(codec, RT5645_GEN_CTRL1, + RT5645_DIG_GATE_CTRL, 0); snd_soc_update_bits(codec, RT5645_PWR_ANLG1, RT5645_PWR_VREF1 | RT5645_PWR_MB | RT5645_PWR_BG | RT5645_PWR_VREF2 | @@ -2414,6 +2417,37 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, return 0; } +static void rt5645_enable_push_button_irq(struct snd_soc_codec *codec, + bool enable) +{ + if (enable) { + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "ADC L power"); + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "ADC R power"); + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "LDO2"); + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync_unlocked(&codec->dapm); + snd_soc_update_bits(codec, RT5645_INT_IRQ_ST, 0x8, 0x8); + snd_soc_update_bits(codec, RT5650_4BTN_IL_CMD2, 0x8000, 0x8000); + snd_soc_read(codec, RT5650_4BTN_IL_CMD1); + } else { + snd_soc_update_bits(codec, RT5650_4BTN_IL_CMD2, 0x8000, 0x0); + snd_soc_update_bits(codec, RT5645_INT_IRQ_ST, 0x8, 0x0); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "ADC L power"); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "ADC R power"); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "LDO2"); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync_unlocked(&codec->dapm); + } +} + static int rt5645_jack_detect(struct snd_soc_codec *codec) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); @@ -2449,17 +2483,24 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec) val = snd_soc_read(codec, RT5645_IN1_CTRL3) & 0x7; dev_dbg(codec->dev, "val = %d\n", val); - if (val == 1 || val == 2) - jack_type = SND_JACK_HEADSET; - else - jack_type = SND_JACK_HEADPHONE; - snd_soc_dapm_disable_pin(&codec->dapm, "micbias1"); snd_soc_dapm_disable_pin(&codec->dapm, "micbias2"); if (rt5645->pdata.jd_mode == 0) snd_soc_dapm_disable_pin(&codec->dapm, "LDO2"); snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); snd_soc_dapm_sync(&codec->dapm); + + if (val == 1 || val == 2) { + jack_type = SND_JACK_HEADSET; + if (rt5645->en_button_func) + rt5645_enable_push_button_irq(codec, true); + } else { + jack_type = SND_JACK_HEADPHONE; + } + + } else { /* jack out */ + if (rt5645->en_button_func) + rt5645_enable_push_button_irq(codec, false); } snd_soc_jack_report(rt5645->hp_jack, jack_type, SND_JACK_HEADPHONE); @@ -2480,6 +2521,42 @@ int rt5645_set_jack_detect(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(rt5645_set_jack_detect); +int rt5650_set_button_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + int ret; + + rt5645->jack = jack; + rt5645->codec_irq.gpiod_dev = codec->dev; + rt5645->codec_irq.name = "rt5650 irq"; + rt5645->codec_irq.report = SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3; + rt5645->codec_irq.debounce_time = 150; + rt5645->codec_irq.wake = true; + rt5645->codec_irq.data = (struct rt5645_priv *)rt5645; + rt5645->codec_irq.jack_status_check = rt5645_irq_detection; + + ret = snd_soc_jack_add_gpios(rt5645->jack, 1, + &rt5645->codec_irq); + if (ret) { + dev_err(codec->dev, "Adding jack GPIO failed\n"); + return ret; + } + + regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, + RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ); + regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1, + RT5645_HP_CB_MASK, RT5645_HP_CB_PU); + regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1, + RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL); + rt5645->en_button_func = true; + + return 0; +} +EXPORT_SYMBOL_GPL(rt5650_set_button_detect); + static void rt5645_jack_detect_work(struct work_struct *work) { struct rt5645_priv *rt5645 = @@ -2498,6 +2575,62 @@ static irqreturn_t rt5645_irq(int irq, void *data) return IRQ_HANDLED; } +static int rt5645_button_detect(struct snd_soc_codec *codec) +{ + int btn_type, val; + + val = snd_soc_read(codec, RT5650_4BTN_IL_CMD1); + pr_debug("val=0x%x\n", val); + btn_type = val & 0xfff0; + snd_soc_write(codec, RT5650_4BTN_IL_CMD1, val); + + return btn_type; +} + +static int rt5645_irq_detection(void *data) +{ + struct rt5645_priv *rt5645 = (struct rt5645_priv *)data; + struct snd_soc_jack *jack = &rt5645->jack; + int btn_type, report = SND_JACK_HEADSET; + + btn_type = rt5645_button_detect(rt5645->codec); + switch (btn_type) { + case 0x8000: + case 0x4000: + case 0x2000: + report |= SND_JACK_BTN_0; + break; + case 0x1000: + case 0x0800: + case 0x0400: + report |= SND_JACK_BTN_1; + break; + case 0x0200: + case 0x0100: + case 0x0080: + report |= SND_JACK_BTN_2; + break; + case 0x0040: + case 0x0020: + case 0x0010: + report |= SND_JACK_BTN_3; + break; + case 0x0000: /* unpressed */ + break; + default: + dev_err(rt5645->codec->dev, + "Unexpected button code 0x%04x\n", + btn_type); + /* keep the jack status */ + report = jack->status; + break; + } + + pr_debug("report=%x\n", report); + + return report; +} + static int rt5645_probe(struct snd_soc_codec *codec) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index db78e94..bf29d8b 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2184,7 +2184,9 @@ struct rt5645_priv { struct i2c_client *i2c; struct snd_soc_jack *hp_jack; struct snd_soc_jack *mic_jack; + struct snd_soc_jack *jack; struct delayed_work jack_detect_work; + struct snd_soc_jack_gpio codec_irq; int codec_type; int sysclk; @@ -2196,9 +2198,12 @@ struct rt5645_priv { int pll_src; int pll_in; int pll_out; + + bool en_button_func; }; int rt5645_set_jack_detect(struct snd_soc_codec *codec, struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack); - +int rt5650_set_button_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); #endif /* __RT5645_H__ */ -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH v3] 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> --- Changes in v3: - modify the shutdown function before remove(..) - fix the `Serien-cc` 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..3e33ef2 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2605,8 +2605,24 @@ err_enable: return ret; } +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); +} + static int max98090_i2c_remove(struct i2c_client *client) { + max98090_i2c_shutdown(client); snd_soc_unregister_codec(&client->dev); return 0; } @@ -2696,6 +2712,7 @@ static struct i2c_driver max98090_i2c_driver = { .acpi_match_table = ACPI_PTR(max98090_acpi_match), }, .probe = max98090_i2c_probe, + .shutdown = max98090_i2c_shutdown, .remove = max98090_i2c_remove, .id_table = max98090_i2c_id, }; -- 1.9.1

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