August 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] sound: fix some checkpatch errors and warnings.
by Peng Donglin 18 Aug '17

18 Aug '17

ERROR: trailing whitespace WARNING: please, no spaces at the start of a line ERROR: open brace '{' following struct go on the same line ERROR: do not initialise statics to 0 ERROR: "foo * bar" should be "foo *bar" ERROR: spaces required around that '=' ERROR: switch and case should be at the same indent WARNING: Statements should start on a tabstop WARNING: Block comments use * on subsequent lines Signed-off-by: Peng Donglin <dolinux.peng(a)gmail.com> --- sound/last.c | 2 +- sound/sound_core.c | 163 ++++++++++++++++++++++++----------------------------- 2 files changed, 76 insertions(+), 89 deletions(-) diff --git a/sound/last.c b/sound/last.c index 43f2228..a47bbe0 100644 --- a/sound/last.c +++ b/sound/last.c @@ -25,7 +25,7 @@ static int __init alsa_sound_last_init(void) { int idx, ok = 0; - + printk(KERN_INFO "ALSA device list:\n"); for (idx = 0; idx < SNDRV_CARDS; idx++) if (snd_cards[idx] != NULL) { diff --git a/sound/sound_core.c b/sound/sound_core.c index 99b73c6..f99bb7a 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -68,7 +68,7 @@ static void __exit cleanup_soundcore(void) #ifdef CONFIG_SOUND_OSS_CORE /* * OSS sound core handling. Breaks out sound functions to submodules - * + * * Author: Alan Cox <alan(a)lxorguk.ukuu.org.uk> * * Fixes: @@ -80,9 +80,9 @@ static void __exit cleanup_soundcore(void) * 2 of the License, or (at your option) any later version. * * -------------------- - * + * * Top level handler for the sound subsystem. Various devices can - * plug into this. The fact they don't all go via OSS doesn't mean + * plug into this. The fact they don't all go via OSS doesn't mean * they don't have to implement the OSS API. There is a lot of logic * to keeping much of the OSS weight out of the code in a compatibility * module, but it's up to the driver to rember to load it... @@ -111,8 +111,7 @@ static void __exit cleanup_soundcore(void) #define SOUND_STEP 16 -struct sound_unit -{ +struct sound_unit { int unit_minor; const struct file_operations *unit_fops; struct sound_unit *next; @@ -151,15 +150,14 @@ struct sound_unit #ifdef CONFIG_SOUND_OSS_CORE_PRECLAIM static int preclaim_oss = 1; #else -static int preclaim_oss = 0; +static int preclaim_oss; #endif module_param(preclaim_oss, int, 0444); static int soundcore_open(struct inode *, struct file *); -static const struct file_operations soundcore_fops = -{ +static const struct file_operations soundcore_fops = { /* We must have an owner or the module locking fails */ .owner = THIS_MODULE, .open = soundcore_open, @@ -171,70 +169,68 @@ struct sound_unit * join into it. Called with the lock asserted */ -static int __sound_insert_unit(struct sound_unit * s, struct sound_unit **list, const struct file_operations *fops, int index, int low, int top) +static int __sound_insert_unit(struct sound_unit *s, struct sound_unit **list, + const struct file_operations *fops, int index, + int low, int top) { - int n=low; + int n = low; if (index < 0) { /* first free */ - while (*list && (*list)->unit_minor<n) - list=&((*list)->next); + while (*list && (*list)->unit_minor < n) + list = &((*list)->next); - while(n<top) - { + while (n < top) { /* Found a hole ? */ - if(*list==NULL || (*list)->unit_minor>n) + if (*list == NULL || (*list)->unit_minor > n) break; - list=&((*list)->next); - n+=SOUND_STEP; + list = &((*list)->next); + n += SOUND_STEP; } - if(n>=top) + if (n >= top) return -ENOENT; } else { n = low+(index*16); while (*list) { - if ((*list)->unit_minor==n) + if ((*list)->unit_minor == n) return -EBUSY; - if ((*list)->unit_minor>n) + if ((*list)->unit_minor > n) break; - list=&((*list)->next); + list = &((*list)->next); } - } - + } + /* * Fill it in */ - - s->unit_minor=n; - s->unit_fops=fops; - + + s->unit_minor = n; + s->unit_fops = fops; + /* * Link it */ - - s->next=*list; - *list=s; - - + + s->next = *list; + *list = s; + return n; } /* * Remove a node from the chain. Called with the lock asserted */ - + static struct sound_unit *__sound_remove_unit(struct sound_unit **list, int unit) { - while(*list) - { - struct sound_unit *p=*list; - if(p->unit_minor==unit) - { - *list=p->next; + while (*list) { + struct sound_unit *p = *list; + if (p->unit_minor == unit) { + *list = p->next; return p; } - list=&(p->next); + list = &(p->next); } printk(KERN_ERR "Sound device %d went missing!\n", unit); return NULL; @@ -251,7 +247,9 @@ static struct sound_unit *__sound_remove_unit(struct sound_unit **list, int unit * list. Acquires locks as needed */ -static int sound_insert_unit(struct sound_unit **list, const struct file_operations *fops, int index, int low, int top, const char *name, umode_t mode, struct device *dev) +static int sound_insert_unit(struct sound_unit **list, const struct file_operations *fops, + int index, int low, int top, const char *name, + umode_t mode, struct device *dev) { struct sound_unit *s = kmalloc(sizeof(*s), GFP_KERNEL); int r; @@ -263,7 +261,7 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati retry: r = __sound_insert_unit(s, list, fops, index, low, top); spin_unlock(&sound_loader_lock); - + if (r < 0) goto fail; else if (r < SOUND_STEP) @@ -305,7 +303,7 @@ static int sound_insert_unit(struct sound_unit **list, const struct file_operati * completed the removal before their file operations become * invalid. */ - + static void sound_remove_unit(struct sound_unit **list, int unit) { struct sound_unit *p; @@ -357,7 +355,7 @@ static void sound_remove_unit(struct sound_unit **list, int unit) * Return: The allocated number is returned on success. On failure, * a negative error code is returned. */ - + int register_sound_special_device(const struct file_operations *fops, int unit, struct device *dev) { @@ -367,69 +365,67 @@ int register_sound_special_device(const struct file_operations *fops, int unit, char _name[16]; switch (chain) { - case 0: + case 0: name = "mixer"; break; - case 1: + case 1: name = "sequencer"; if (unit >= SOUND_STEP) goto __unknown; max_unit = unit + 1; break; - case 2: + case 2: name = "midi"; break; - case 3: + case 3: name = "dsp"; break; - case 4: + case 4: name = "audio"; break; - case 5: + case 5: name = "dspW"; break; - case 8: + case 8: name = "sequencer2"; if (unit >= SOUND_STEP) goto __unknown; max_unit = unit + 1; break; - case 9: + case 9: name = "dmmidi"; break; - case 10: + case 10: name = "dmfm"; break; - case 12: + case 12: name = "adsp"; break; - case 13: + case 13: name = "amidi"; break; - case 14: + case 14: name = "admmidi"; break; - default: - { - __unknown: + default: + { + __unknown: sprintf(_name, "unknown%d", chain); - if (unit >= SOUND_STEP) - strcat(_name, "-"); - name = _name; + if (unit >= SOUND_STEP) + strcat(_name, "-"); + name = _name; } break; } return sound_insert_unit(&chains[chain], fops, -1, unit, max_unit, name, S_IRUSR | S_IWUSR, dev); } - EXPORT_SYMBOL(register_sound_special_device); int register_sound_special(const struct file_operations *fops, int unit) { return register_sound_special_device(fops, unit, NULL); } - EXPORT_SYMBOL(register_sound_special); /** @@ -449,7 +445,6 @@ int register_sound_mixer(const struct file_operations *fops, int dev) return sound_insert_unit(&chains[0], fops, dev, 0, 128, "mixer", S_IRUSR | S_IWUSR, NULL); } - EXPORT_SYMBOL(register_sound_mixer); /** @@ -469,14 +464,13 @@ int register_sound_midi(const struct file_operations *fops, int dev) return sound_insert_unit(&chains[2], fops, dev, 2, 130, "midi", S_IRUSR | S_IWUSR, NULL); } - EXPORT_SYMBOL(register_sound_midi); /* * DSP's are registered as a triple. Register only one and cheat * in open - see below. */ - + /** * register_sound_dsp - register a DSP device * @fops: File operations for the driver @@ -497,7 +491,6 @@ int register_sound_dsp(const struct file_operations *fops, int dev) return sound_insert_unit(&chains[3], fops, dev, 3, 131, "dsp", S_IWUSR | S_IRUSR, NULL); } - EXPORT_SYMBOL(register_sound_dsp); /** @@ -514,7 +507,6 @@ void unregister_sound_special(int unit) { sound_remove_unit(&chains[unit % SOUND_STEP], unit); } - EXPORT_SYMBOL(unregister_sound_special); /** @@ -529,7 +521,6 @@ void unregister_sound_mixer(int unit) { sound_remove_unit(&chains[0], unit); } - EXPORT_SYMBOL(unregister_sound_mixer); /** @@ -544,7 +535,6 @@ void unregister_sound_midi(int unit) { sound_remove_unit(&chains[2], unit); } - EXPORT_SYMBOL(unregister_sound_midi); /** @@ -561,20 +551,17 @@ void unregister_sound_dsp(int unit) { sound_remove_unit(&chains[3], unit); } - - EXPORT_SYMBOL(unregister_sound_dsp); static struct sound_unit *__look_for_unit(int chain, int unit) { struct sound_unit *s; - - s=chains[chain]; - while(s && s->unit_minor <= unit) - { - if(s->unit_minor==unit) + + s = chains[chain]; + while (s && s->unit_minor <= unit) { + if (s->unit_minor == unit) return s; - s=s->next; + s = s->next; } return NULL; } @@ -586,14 +573,13 @@ static int soundcore_open(struct inode *inode, struct file *file) struct sound_unit *s; const struct file_operations *new_fops = NULL; - chain=unit&0x0F; - if(chain==4 || chain==5) /* dsp/audio/dsp16 */ - { - unit&=0xF0; - unit|=3; - chain=3; + chain = unit&0x0F; + if (chain == 4 || chain == 5) { /* dsp/audio/dsp16 */ + unit &= 0xF0; + unit |= 3; + chain = 3; } - + spin_lock(&sound_loader_lock); s = __look_for_unit(chain, unit); if (s) @@ -636,7 +622,7 @@ static int soundcore_open(struct inode *inode, struct file *file) replace_fops(file, new_fops); if (file->f_op->open) - err = file->f_op->open(inode,file); + err = file->f_op->open(inode, file); return err; } @@ -648,7 +634,8 @@ static int soundcore_open(struct inode *inode, struct file *file) static void cleanup_oss_soundcore(void) { /* We have nothing to really do here - we know the lists must be - empty */ + * empty + */ unregister_chrdev(SOUND_MAJOR, "sound"); } -- 1.9.1

1 0

[alsa-devel] non-blocking mode example
by ThomasJF 18 Aug '17

18 Aug '17

Hi, could anyone please link me to or show me a small demo example, like "pcm_min.c" with working snd_pcm_open in SND_PCM_NONBLOCK - mode? Standard I/O transfers or Event waiting routines. I get both working - but only in blocking mode. I want to stop/pause with commands inside a tcp-server. Regards Thomas

2 14

[alsa-devel] [PATCH 00/14] constify asoc snd_soc_dai_ops structures
by Arvind Yadav 18 Aug '17

18 Aug '17

snd_soc_dai_ops are not supposed to change at runtime. All functions working with snd_soc_dai_ops provided by <sound/soc-dai.h> work with const snd_soc_dai_ops. So mark the non-const structs as const. Arvind Yadav (14): [PATCH 01/14] ASoC: blackfin: constify snd_soc_dai_ops structures [PATCH 02/14] ASoC: codecs: constify snd_soc_dai_ops structures [PATCH 03/14] ASoC: dwc: constify snd_soc_dai_ops structures [PATCH 04/14] ASoC: fsl: constify snd_soc_dai_ops structures [PATCH 05/14] ASoC: hisilicon: constify snd_soc_dai_ops structures [PATCH 06/14] ASoC: intel: constify snd_soc_dai_ops structures [PATCH 07/14] ASoC: mediatek: constify snd_soc_dai_ops structures [PATCH 08/14] ASoC: pxa: constify snd_soc_dai_ops structures [PATCH 09/14] ASoC: rockchip: constify snd_soc_dai_ops structures [PATCH 10/14] ASoC: spear: constify snd_soc_dai_ops structures [PATCH 11/14] ASoC: sunxi: constify snd_soc_dai_ops structures [PATCH 12/14] ASoC: tegra: constify snd_soc_dai_ops structures [PATCH 13/14] ASoC: ux500: constify snd_soc_dai_ops structures [PATCH 14/14] ASoC: zte: constify snd_soc_dai_ops structures sound/soc/blackfin/bf6xx-i2s.c | 2 +- sound/soc/codecs/cs42l42.c | 2 +- sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/hdac_hdmi.c | 2 +- sound/soc/codecs/inno_rk3036.c | 2 +- sound/soc/codecs/max9867.c | 2 +- sound/soc/codecs/max98926.c | 2 +- sound/soc/codecs/msm8916-wcd-analog.c | 2 +- sound/soc/codecs/msm8916-wcd-digital.c | 2 +- sound/soc/codecs/rt5616.c | 2 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/tas5720.c | 2 +- sound/soc/codecs/zx_aud96p22.c | 2 +- sound/soc/dwc/dwc-i2s.c | 2 +- sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_spdif.c | 2 +- sound/soc/hisilicon/hi6210-i2s.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +++--- sound/soc/intel/skylake/skl-pcm.c | 8 ++++---- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/spear/spdif_in.c | 2 +- sound/soc/sunxi/sun8i-codec.c | 2 +- sound/soc/tegra/tegra30_i2s.c | 2 +- sound/soc/ux500/ux500_msp_dai.c | 2 +- sound/soc/zte/zx-i2s.c | 2 +- sound/soc/zte/zx-spdif.c | 2 +- sound/soc/zte/zx-tdm.c | 2 +- 30 files changed, 35 insertions(+), 35 deletions(-) -- 1.9.1

2 21

[alsa-devel] Applied "ASoC: codecs: constify snd_soc_dai_ops structures" to the asoc tree
by Mark Brown 18 Aug '17

18 Aug '17

The patch ASoC: codecs: constify snd_soc_dai_ops structures has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From eb59d73cb535ba32df928f210fb9a8529bf465c0 Mon Sep 17 00:00:00 2001 From: Arvind Yadav <arvind.yadav.cs(a)gmail.com> Date: Fri, 18 Aug 2017 17:35:59 +0530 Subject: [PATCH] ASoC: codecs: constify snd_soc_dai_ops structures snd_soc_dai_ops are not supposed to change at runtime. All functions working with snd_soc_dai_ops provided by <sound/soc-dai.h> work with const snd_soc_dai_ops. So mark the non-const structs as const. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cs42l42.c | 2 +- sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/inno_rk3036.c | 2 +- sound/soc/codecs/max9867.c | 2 +- sound/soc/codecs/max98926.c | 2 +- sound/soc/codecs/msm8916-wcd-analog.c | 2 +- sound/soc/codecs/rt5616.c | 2 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/tas5720.c | 2 +- sound/soc/codecs/zx_aud96p22.c | 2 +- 10 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 55e4520cdcaf..9e860dfa0163 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -911,7 +911,7 @@ static int cs42l42_digital_mute(struct snd_soc_dai *dai, int mute) SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops cs42l42_ops = { +static const struct snd_soc_dai_ops cs42l42_ops = { .hw_params = cs42l42_pcm_hw_params, .set_fmt = cs42l42_set_dai_fmt, .set_sysclk = cs42l42_set_sysclk, diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 4f35af6a5d3a..da2d353af5ba 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -502,7 +502,7 @@ static int es8316_mute(struct snd_soc_dai *dai, int mute) #define ES8316_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops es8316_ops = { +static const struct snd_soc_dai_ops es8316_ops = { .startup = es8316_pcm_startup, .hw_params = es8316_pcm_hw_params, .set_fmt = es8316_set_dai_fmt, diff --git a/sound/soc/codecs/inno_rk3036.c b/sound/soc/codecs/inno_rk3036.c index 64b0be94bea3..6b59b6f08298 100644 --- a/sound/soc/codecs/inno_rk3036.c +++ b/sound/soc/codecs/inno_rk3036.c @@ -310,7 +310,7 @@ static int rk3036_codec_dai_hw_params(struct snd_pcm_substream *substream, SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops rk3036_codec_dai_ops = { +static const struct snd_soc_dai_ops rk3036_codec_dai_ops = { .set_fmt = rk3036_codec_dai_set_fmt, .hw_params = rk3036_codec_dai_hw_params, }; diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 6c0c0d6e8f3c..2f60924fe919 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -350,7 +350,7 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, return 0; } -static struct snd_soc_dai_ops max9867_dai_ops = { +static const struct snd_soc_dai_ops max9867_dai_ops = { .set_fmt = max9867_dai_set_fmt, .set_sysclk = max9867_set_dai_sysclk, .prepare = max9867_prepare, diff --git a/sound/soc/codecs/max98926.c b/sound/soc/codecs/max98926.c index 7a39bfb9e0f9..59d03d036945 100644 --- a/sound/soc/codecs/max98926.c +++ b/sound/soc/codecs/max98926.c @@ -459,7 +459,7 @@ static int max98926_dai_hw_params(struct snd_pcm_substream *substream, #define MAX98926_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops max98926_dai_ops = { +static const struct snd_soc_dai_ops max98926_dai_ops = { .set_fmt = max98926_dai_set_fmt, .hw_params = max98926_dai_hw_params, }; diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 4cc52cecf3b7..f07674eddc52 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -753,7 +753,7 @@ static void pm8916_wcd_analog_shutdown(struct snd_pcm_substream *substream, RST_CTL_DIG_SW_RST_N_MASK, 0); } -static struct snd_soc_dai_ops pm8916_wcd_analog_dai_ops = { +static const struct snd_soc_dai_ops pm8916_wcd_analog_dai_ops = { .startup = pm8916_wcd_analog_startup, .shutdown = pm8916_wcd_analog_shutdown, }; diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index 3c5f555ae6d0..c94e94fe8297 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -1265,7 +1265,7 @@ static int rt5616_resume(struct snd_soc_codec *codec) #define RT5616_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -static struct snd_soc_dai_ops rt5616_aif_dai_ops = { +static const struct snd_soc_dai_ops rt5616_aif_dai_ops = { .hw_params = rt5616_hw_params, .set_fmt = rt5616_set_dai_fmt, .set_sysclk = rt5616_set_dai_sysclk, diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index ebc5e3b2b62b..1bbe36338cd3 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -2860,7 +2860,7 @@ static int rt5663_resume(struct snd_soc_codec *codec) #define RT5663_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -static struct snd_soc_dai_ops rt5663_aif_dai_ops = { +static const struct snd_soc_dai_ops rt5663_aif_dai_ops = { .hw_params = rt5663_hw_params, .set_fmt = rt5663_set_dai_fmt, .set_sysclk = rt5663_set_dai_sysclk, diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index d14b78b1917b..a736a2a6976c 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -507,7 +507,7 @@ static const struct snd_soc_codec_driver soc_codec_dev_tas5720 = { #define TAS5720_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE |\ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_dai_ops tas5720_speaker_dai_ops = { +static const struct snd_soc_dai_ops tas5720_speaker_dai_ops = { .hw_params = tas5720_hw_params, .set_fmt = tas5720_set_dai_fmt, .set_tdm_slot = tas5720_set_dai_tdm_slot, diff --git a/sound/soc/codecs/zx_aud96p22.c b/sound/soc/codecs/zx_aud96p22.c index 36d0a7087a54..51fad48892d3 100644 --- a/sound/soc/codecs/zx_aud96p22.c +++ b/sound/soc/codecs/zx_aud96p22.c @@ -312,7 +312,7 @@ static int aud96p22_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static struct snd_soc_dai_ops aud96p22_dai_ops = { +static const struct snd_soc_dai_ops aud96p22_dai_ops = { .set_fmt = aud96p22_set_fmt, }; -- 2.13.2

1 0

[alsa-devel] [PATCH 1/3] sound: ASoC: tegra: Remove SoC-specific Kconfig depends and selects
by Paul Kocialkowski 18 Aug '17

18 Aug '17

This removes the SoC-specific dependencies on the platform drivers, as well as SoC-specific selections of platform drivers for the machine drivers. The rationale behind this change is that the dependencies are not actual build dependencies but run-time ones. The previously listed SoCs were also incomplete: for instance, tegra124 uses the tegra30 platform drivers, which could not be built without ARCH_TEGRA_3x_SOC set. Descriptions of the tristates are also added to allow these options to be set via defconfig/menuconfig. Signed-off-by: Paul Kocialkowski <contact(a)paulk.fr> --- sound/soc/tegra/Kconfig | 42 ++++++++++++++---------------------------- 1 file changed, 14 insertions(+), 28 deletions(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index efbe8d4c019e..6875fc39a575 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -9,8 +9,8 @@ config SND_SOC_TEGRA Say Y or M here if you want support for SoC audio on Tegra. config SND_SOC_TEGRA20_AC97 - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + tristate "Tegra20 AC97 interface" + depends on SND_SOC_TEGRA select SND_SOC_AC97_BUS select SND_SOC_TEGRA20_DAS help @@ -19,16 +19,16 @@ config SND_SOC_TEGRA20_AC97 machine drivers to support below. config SND_SOC_TEGRA20_DAS - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + tristate "Tegra20 DAS module" + depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra20 DAS module. You will also need to select the individual machine drivers to support below. config SND_SOC_TEGRA20_I2S - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + tristate "Tegra20 I2S interface" + depends on SND_SOC_TEGRA select SND_SOC_TEGRA20_DAS help Say Y or M if you want to add support for codecs attached to the @@ -36,8 +36,8 @@ config SND_SOC_TEGRA20_I2S machine drivers to support below. config SND_SOC_TEGRA20_SPDIF - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC + tristate "Tegra20 SPDIF interface" + depends on SND_SOC_TEGRA default m help Say Y or M if you want to add support for the Tegra20 SPDIF interface. @@ -45,16 +45,16 @@ config SND_SOC_TEGRA20_SPDIF below. config SND_SOC_TEGRA30_AHUB - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC + tristate "Tegra30 AHUB module" + depends on SND_SOC_TEGRA help - Say Y or M if you want to add support for the Tegra20 AHUB module. + Say Y or M if you want to add support for the Tegra30 AHUB module. You will also need to select the individual machine drivers to support below. config SND_SOC_TEGRA30_I2S - tristate - depends on SND_SOC_TEGRA && ARCH_TEGRA_3x_SOC + tristate "Tegra30 I2S interface" + depends on SND_SOC_TEGRA select SND_SOC_TEGRA30_AHUB help Say Y or M if you want to add support for codecs attached to the @@ -64,8 +64,6 @@ config SND_SOC_TEGRA30_I2S config SND_SOC_TEGRA_RT5640 tristate "SoC Audio support for Tegra boards using an RT5640 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_RT5640 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -74,8 +72,6 @@ config SND_SOC_TEGRA_RT5640 config SND_SOC_TEGRA_WM8753 tristate "SoC Audio support for Tegra boards using a WM8753 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_WM8753 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -84,8 +80,6 @@ config SND_SOC_TEGRA_WM8753 config SND_SOC_TEGRA_WM8903 tristate "SoC Audio support for Tegra boards using a WM8903 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_WM8903 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -94,7 +88,7 @@ config SND_SOC_TEGRA_WM8903 config SND_SOC_TEGRA_WM9712 tristate "SoC Audio support for Tegra boards using a WM9712 codec" - depends on SND_SOC_TEGRA && ARCH_TEGRA_2x_SOC && GPIOLIB + depends on SND_SOC_TEGRA && GPIOLIB select SND_SOC_TEGRA20_AC97 select SND_SOC_WM9712 help @@ -104,7 +98,6 @@ config SND_SOC_TEGRA_WM9712 config SND_SOC_TEGRA_TRIMSLICE tristate "SoC Audio support for TrimSlice board" depends on SND_SOC_TEGRA && I2C - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_TLV320AIC23_I2C help Say Y or M here if you want to add support for SoC audio on the @@ -113,7 +106,6 @@ config SND_SOC_TEGRA_TRIMSLICE config SND_SOC_TEGRA_ALC5632 tristate "SoC Audio support for Tegra boards using an ALC5632 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC select SND_SOC_ALC5632 help Say Y or M here if you want to add support for SoC audio on the @@ -122,8 +114,6 @@ config SND_SOC_TEGRA_ALC5632 config SND_SOC_TEGRA_MAX98090 tristate "SoC Audio support for Tegra boards using a MAX98090 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_MAX98090 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -132,8 +122,6 @@ config SND_SOC_TEGRA_MAX98090 config SND_SOC_TEGRA_RT5677 tristate "SoC Audio support for Tegra boards using a RT5677 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_RT5677 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -142,8 +130,6 @@ config SND_SOC_TEGRA_RT5677 config SND_SOC_TEGRA_SGTL5000 tristate "SoC Audio support for Tegra boards using a SGTL5000 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB - select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC select SND_SOC_SGTL5000 help Say Y or M here if you want to add support for SoC audio on Tegra -- 2.13.3

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[alsa-devel] [PATCH v1 1/2] ASoC: mediatek: Add interrupt-names property in binding text
by Ryder Lee 18 Aug '17

18 Aug '17

Add interrupt-names property in DT - if future revisions of the IP have different sets of interrupts the binding can adapt gracefully. Signed-off-by: Ryder Lee <ryder.lee(a)mediatek.com> CC: Garlic Tseng <garlic.tseng(a)mediatek.com> --- Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt index 9800a56..77a57f8 100644 --- a/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt +++ b/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt @@ -3,7 +3,8 @@ Mediatek AFE PCM controller for mt2701 Required properties: - compatible = "mediatek,mt2701-audio"; - reg: register location and size -- interrupts: Should contain AFE interrupt +- interrupts: should contain AFE and ASYS interrupts +- interrupt-names: should be "afe" and "asys" - power-domains: should define the power domain - clock-names: should have these clock names: "infra_sys_audio_clk", @@ -59,6 +60,7 @@ Example: <0 0x112A0000 0 0x20000>; interrupts = <GIC_SPI 104 IRQ_TYPE_LEVEL_LOW>, <GIC_SPI 132 IRQ_TYPE_LEVEL_LOW>; + interrupt-names = "afe", "asys"; power-domains = <&scpsys MT2701_POWER_DOMAIN_IFR_MSC>; clocks = <&infracfg CLK_INFRA_AUDIO>, <&topckgen CLK_TOP_AUD_MUX1_SEL>, -- 1.9.1

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[alsa-devel] [PATCHv3 0/6] Motorola Droid 4 Audio Support
by Sebastian Reichel 18 Aug '17

18 Aug '17

Hi, This adds audio support to Motorola Droid 4. Tested: - Playing via both DACs using Speaker, Earpiece, Headphone - Recording using internal Mic - Volume Controls Known not to work: - 3.5mm detection support (requires closed source firmware, needs further investigation) - Modem / Bluetooth Audio (actually untested, but probably needs some quirks) Changes since PATCHv2: * Fix a whitespace issue * Fix const notes Takashi provided * Fix a DAPM route issue I accidently introduced in v2 Changes since PATCHv1: * Add patch from Tony exporting soc_dpcm_runtime_update * Integrate DT patch for vaudio initial mode * Split dt-binding from codec patch and add Ack from Rob * Fix CPCAP position in Kconfig/Makefile * Avoid "err +=" constructs * Simplify reset function * Drop cpcap_audio_read/write helpers * Do not use tertiary operator for mute register value * Update Input Mux logic * Switch from simple-audio-card to audio-graph-card -- Sebastian Sebastian Reichel (5): dt-bindings: sound: add motorola,cpcap-audio-codec ASoC: codec: cpcap: new codec ARM: dts: motorola-cpcap-mapphone: add audio-codec ARM: dts: motorola-cpcap-mapphone: set initial mode for vaudio ARM: dts: omap4-droid4: add soundcard Tony Lindgren (1): ALSA: pcm: Export soc_dpcm_runtime_update .../bindings/sound/motorola,cpcap-audio-codec.txt | 19 + arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi | 7 + arch/arm/boot/dts/omap4-droid4-xt894.dts | 84 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cpcap.c | 1594 ++++++++++++++++++++ sound/soc/soc-pcm.c | 2 + 7 files changed, 1713 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt create mode 100644 sound/soc/codecs/cpcap.c -- 2.13.2

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[alsa-devel] [PATCH] ALSA: emu10k1: Fix forgotten user-copy conversion in init code
by Takashi Iwai 18 Aug '17

18 Aug '17

The commit d42fe63d5839 ("ALSA: emu10k1: Get rid of set_fs() usage") converted the user-space copy hack with set_fs() to the direct memcpy(), but one place was forgotten. This resulted in the error from snd_emu10k1_init_efx(), eventually failed to load the driver. Fix the missing piece. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=196687 Fixes: d42fe63d5839 ("ALSA: emu10k1: Get rid of set_fs() usage") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/emu10k1/emufx.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index dc585959ca32..a2b56b188be4 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -698,10 +698,18 @@ static int copy_gctl(struct snd_emu10k1 *emu, { struct snd_emu10k1_fx8010_control_old_gpr __user *octl; - if (emu->support_tlv) - return copy_from_user(gctl, &_gctl[idx], sizeof(*gctl)); + if (emu->support_tlv) { + if (in_kernel) + memcpy(gctl, (void *)&_gctl[idx], sizeof(*gctl)); + else if (copy_from_user(gctl, &_gctl[idx], sizeof(*gctl))) + return -EFAULT; + return 0; + } + octl = (struct snd_emu10k1_fx8010_control_old_gpr __user *)_gctl; - if (copy_from_user(gctl, &octl[idx], sizeof(*octl))) + if (in_kernel) + memcpy(gctl, (void *)&octl[idx], sizeof(*octl)); + else if (copy_from_user(gctl, &octl[idx], sizeof(*octl))) return -EFAULT; gctl->tlv = NULL; return 0; -- 2.14.1

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[alsa-devel] [PATCH] ALSA: usb-audio: add DSD support for new Amanero PID
by Jussi Laako 18 Aug '17

18 Aug '17

Add DSD support for new Amanero Combo384 firmware version with a new PID. This firmware uses DSD_U32_BE. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/quirks.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2eb6e0d..c09a28b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1374,6 +1374,10 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, } } break; + case USB_ID(0x16d0, 0x0a23): + if (fp->altsetting == 2) + return SNDRV_PCM_FMTBIT_DSD_U32_BE; + break; default: break; -- 2.7.4

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[alsa-devel] [PATCH 0/6] ASoC: make snd_pcm_hardware const
by Bhumika Goyal 18 Aug '17

18 Aug '17

Make these const. Done using Coccinelle Bhumika Goyal (6): ASoC: fsl: make snd_pcm_hardware const ASoC: Intel: Atom: make snd_pcm_hardware const ASoC: Intel: Skylake: make snd_pcm_hardware const ASoC: kirkwood: make snd_pcm_hardware const ASoC: sh: make snd_pcm_hardware const ASoC: qcom: make snd_pcm_hardware const sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/imx-pcm-fiq.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/sh/dma-sh7760.c | 2 +- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/rcar/core.c | 2 +- sound/soc/sh/siu_dai.c | 2 +- 10 files changed, 10 insertions(+), 10 deletions(-) -- 1.9.1

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