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

[alsa-devel] [PATCH] ASoC: Intel: move sysclk source setting to platform_clock_control for balance.
by Jin Yao 10 Mar '15

10 Mar '15

A playback noise happens after suspend/resume on Braswell. The issue is due to the codec PLL and codec ASRC are not enabled correctly due to the incorrect sysclk setting after resume. This patch resets the sysclk source setting in platform clock control widget handler. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Jin Yao <yao.jin(a)linux.intel.com> --- sound/soc/intel/cht_bsw_rt5672.c | 40 ++++++++++++++++++++++++++++------------ 1 file changed, 28 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index bc8dcac..279df4c 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -50,6 +50,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; struct snd_soc_dai *codec_dai; + int ret; codec_dai = cht_get_codec_dai(card); if (!codec_dai) { @@ -57,17 +58,31 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } - if (!SND_SOC_DAPM_EVENT_OFF(event)) - return 0; - - /* Set codec sysclk source to its internal clock because codec PLL will - * be off when idle and MCLK will also be off by ACPI when codec is - * runtime suspended. Codec needs clock for jack detection and button - * press. - */ - snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_RCCLK, - 0, SND_SOC_CLOCK_IN); - + if (SND_SOC_DAPM_EVENT_ON(event)) { + /* set codec PLL source to the 19.2MHz platform clock (MCLK) */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK, + CHT_PLAT_CLK_3_HZ, 48000 * 512); + if (ret < 0) { + dev_err(card->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + /* set codec sysclk source to PLL */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1, + 48000 * 512, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + } else { + /* Set codec sysclk source to its internal clock because codec + * PLL will be off when idle and MCLK will also be off by ACPI + * when codec is runtime suspended. Codec needs clock for jack + * detection and button press. + */ + snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_RCCLK, + 48000 * 512, SND_SOC_CLOCK_IN); + } return 0; } @@ -77,7 +92,8 @@ static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_POST_PMD), + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route cht_audio_map[] = { -- 1.9.1

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[alsa-devel] [PATCH 0/2] ASoC: rsnd: fixup patches for DMA/debugfs
by Kuninori Morimoto 10 Mar '15

10 Mar '15

Hi Mark These are rsnd fixup patches 1) is for new DMA method 2) is for debugfs these are based on mark/topic/rsnd Kuninori Morimoto (2): 1) ASoC: rsnd: recover PIO mode for new dma interface 2) ASoC: rsnd: add regmap_config::name for debugfs sound/soc/sh/rcar/dma.c | 19 ++++++++++++++++++- sound/soc/sh/rcar/gen.c | 1 + 2 files changed, 19 insertions(+), 1 deletion(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 00/14] Fix wrong ctl value references in ASoC codecs
by Takashi Iwai 10 Mar '15

10 Mar '15

Hi, while looking through the ASoC codec codes, I noticed that lots of drivers refer to the wrong union value type for boolean controls. They should be value.integer.value[], but the code accesses to value.enumerated.item[]. They haven't been noticed so far maybe just because both are same on 32bit architectures. But they are incompatible on 64bit architectures, thus must be fixed. Takashi === Takashi Iwai (14): ASoC: adav80x: Fix wrong value references for boolean kctl ASoC: ak4641: Fix wrong value references for boolean kctl ASoC: cs4271: Fix wrong value references for boolean kctl ASoC: es8238: Fix wrong value references for boolean kctl ASoC: pcm1681: Fix wrong value references for boolean kctl ASoC: tas5086: Fix wrong value references for boolean kctl ASoC: wm2000: Fix wrong value references for boolean kctl ASoC: wm8731: Fix wrong value references for boolean kctl ASoC: wm8903: Fix wrong value references for boolean kctl ASoC: wm8904: Fix wrong value references for boolean kctl ASoC: wm8955: Fix wrong value references for boolean kctl ASoC: wm8960: Fix wrong value references for boolean kctl ASoC: wm9712: Fix wrong value references for boolean kctl ASoC: wm9713: Fix wrong value references for boolean kctl sound/soc/codecs/adav80x.c | 4 ++-- sound/soc/codecs/ak4641.c | 4 ++-- sound/soc/codecs/cs4271.c | 4 ++-- sound/soc/codecs/es8328.c | 4 ++-- sound/soc/codecs/pcm1681.c | 4 ++-- sound/soc/codecs/tas5086.c | 4 ++-- sound/soc/codecs/wm2000.c | 8 ++++---- sound/soc/codecs/wm8731.c | 4 ++-- sound/soc/codecs/wm8903.c | 4 ++-- sound/soc/codecs/wm8904.c | 4 ++-- sound/soc/codecs/wm8955.c | 4 ++-- sound/soc/codecs/wm8960.c | 4 ++-- sound/soc/codecs/wm9712.c | 6 +++--- sound/soc/codecs/wm9713.c | 6 +++--- 14 files changed, 32 insertions(+), 32 deletions(-) -- 2.3.2

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[alsa-devel] [PATCH 0/2] simple-card: Add samplerate and samplewidth constraints
by Jyri Sarha 10 Mar '15

10 Mar '15

These constraints help to sort out the format and samplerate combinations that do not properly work on a specific HW. Jyri Sarha (2): ASoC: simple-card: Remove trailing whitespace ASoC: simple-card: Add support for samplerate and samplewidth constraints .../devicetree/bindings/sound/simple-card.txt | 6 ++ sound/soc/generic/simple-card.c | 89 +++++++++++++++++++++- 2 files changed, 94 insertions(+), 1 deletion(-) -- 1.9.1

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[alsa-devel] [PATCH] ALSA: firewire-lib: leave unit reference counting completely
by Takashi Sakamoto 10 Mar '15

10 Mar '15

With previous commit, this module managed to leave the counting to each drivers, but the isochronous resources functionality still increment/decrement the count. This commit purge such codes to leave the responsibility to each drivers. Fix: c6f224dc20ad ('ALSA: firewire-lib: remove reference counting') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/iso-resources.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/firewire/iso-resources.c b/sound/firewire/iso-resources.c index 5f17b77..f0e4d50 100644 --- a/sound/firewire/iso-resources.c +++ b/sound/firewire/iso-resources.c @@ -26,7 +26,7 @@ int fw_iso_resources_init(struct fw_iso_resources *r, struct fw_unit *unit) { r->channels_mask = ~0uLL; - r->unit = fw_unit_get(unit); + r->unit = unit; mutex_init(&r->mutex); r->allocated = false; @@ -42,7 +42,6 @@ void fw_iso_resources_destroy(struct fw_iso_resources *r) { WARN_ON(r->allocated); mutex_destroy(&r->mutex); - fw_unit_put(r->unit); } EXPORT_SYMBOL(fw_iso_resources_destroy); -- 2.1.0

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[alsa-devel] [PATCH 0/8] ALSA: control: refactoring core codes
by Takashi Sakamoto 10 Mar '15

10 Mar '15

This patchset is for refactoring core control codes. This is also a preparation for my further work for ALSA control functionality. Takashi Sakamoto (8): ALSA: control: change type from long due to the definition of sizeof operator ALSA: control: obsolete switch statement with const value table ALSA: control: gathering evaluations to access ALSA: control: add a comment about locking values after creating ALSA: control: rename loop index to i ALSA: control: fix over 80 characters lines ALSA: control: rename to appropriate macro name ALSA: control: arrange snd_ctl_new() as a local function sound/core/control.c | 223 ++++++++++++++++++++++++++------------------------- 1 file changed, 113 insertions(+), 110 deletions(-) -- 2.1.0

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[alsa-devel] [PATCH] ALSA: dice: fix wrong offsets for Dice interface
by Takashi Sakamoto 10 Mar '15

10 Mar '15

For received packet stream, the offset of 'RX_SEQ_START' locates after the offset of 'RX_NUMBER_MIDI', although current macro and proc output includes wrong offsets. Fortunately, this bug doesn't affect streaming functionality because these macro is not used. This commit fixes these wrong macro and outputs. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-interface.h | 18 +++++++++--------- sound/firewire/dice/dice-proc.c | 4 ++-- 2 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/firewire/dice/dice-interface.h b/sound/firewire/dice/dice-interface.h index 27b044f..de7602b 100644 --- a/sound/firewire/dice/dice-interface.h +++ b/sound/firewire/dice/dice-interface.h @@ -299,23 +299,23 @@ #define RX_ISOCHRONOUS 0x008 /* - * Index of first quadlet to be interpreted; read/write. If > 0, that many - * quadlets at the beginning of each data block will be ignored, and all the - * audio and MIDI quadlets will follow. - */ -#define RX_SEQ_START 0x00c - -/* * The number of audio channels; read-only. There will be one quadlet per * channel. */ -#define RX_NUMBER_AUDIO 0x010 +#define RX_NUMBER_AUDIO 0x00c /* * The number of MIDI ports, 0-8; read-only. If > 0, there will be one * additional quadlet in each data block, following the audio quadlets. */ -#define RX_NUMBER_MIDI 0x014 +#define RX_NUMBER_MIDI 0x010 + +/* + * Index of first quadlet to be interpreted; read/write. If > 0, that many + * quadlets at the beginning of each data block will be ignored, and all the + * audio and MIDI quadlets will follow. + */ +#define RX_SEQ_START 0x014 /* * Names of all audio channels; read-only. Quadlets are byte-swapped. Names diff --git a/sound/firewire/dice/dice-proc.c b/sound/firewire/dice/dice-proc.c index f5c1d1b..ecfe20f 100644 --- a/sound/firewire/dice/dice-proc.c +++ b/sound/firewire/dice/dice-proc.c @@ -99,9 +99,9 @@ static void dice_proc_read(struct snd_info_entry *entry, } tx; struct { u32 iso; - u32 seq_start; u32 number_audio; u32 number_midi; + u32 seq_start; char names[RX_NAMES_SIZE]; u32 ac3_caps; u32 ac3_enable; @@ -204,10 +204,10 @@ static void dice_proc_read(struct snd_info_entry *entry, break; snd_iprintf(buffer, "rx %u:\n", stream); snd_iprintf(buffer, " iso channel: %d\n", (int)buf.rx.iso); - snd_iprintf(buffer, " sequence start: %u\n", buf.rx.seq_start); snd_iprintf(buffer, " audio channels: %u\n", buf.rx.number_audio); snd_iprintf(buffer, " midi ports: %u\n", buf.rx.number_midi); + snd_iprintf(buffer, " sequence start: %u\n", buf.rx.seq_start); if (quadlets >= 68) { dice_proc_fixup_string(buf.rx.names, RX_NAMES_SIZE); snd_iprintf(buffer, " names: %s\n", buf.rx.names); -- 2.1.0

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[alsa-devel] [PATCH 1/3] ALSA: seq_oss: Drop superfluous error/debug messages after malloc failures
by Takashi Iwai 10 Mar '15

10 Mar '15

The kernel memory allocators already report the errors when the requested allocation fails, thus we don't need to warn it again in each caller side. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/oss/seq_oss_init.c | 4 +--- sound/core/seq/oss/seq_oss_midi.c | 5 ++--- sound/core/seq/oss/seq_oss_readq.c | 9 ++++----- sound/core/seq/oss/seq_oss_synth.c | 6 ++---- 4 files changed, 9 insertions(+), 15 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index b0e32e161dd1..2de3feff70d0 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -188,10 +188,8 @@ snd_seq_oss_open(struct file *file, int level) struct seq_oss_devinfo *dp; dp = kzalloc(sizeof(*dp), GFP_KERNEL); - if (!dp) { - pr_err("ALSA: seq_oss: can't malloc device info\n"); + if (!dp) return -ENOMEM; - } dp->cseq = system_client; dp->port = -1; diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index e79cc44b1394..96e8395ae586 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -173,10 +173,9 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) /* * allocate midi info record */ - if ((mdev = kzalloc(sizeof(*mdev), GFP_KERNEL)) == NULL) { - pr_err("ALSA: seq_oss: can't malloc midi info\n"); + mdev = kzalloc(sizeof(*mdev), GFP_KERNEL); + if (!mdev) return -ENOMEM; - } /* copy the port information */ mdev->client = pinfo->addr.client; diff --git a/sound/core/seq/oss/seq_oss_readq.c b/sound/core/seq/oss/seq_oss_readq.c index 654d17a5023c..c080c73cea04 100644 --- a/sound/core/seq/oss/seq_oss_readq.c +++ b/sound/core/seq/oss/seq_oss_readq.c @@ -47,13 +47,12 @@ snd_seq_oss_readq_new(struct seq_oss_devinfo *dp, int maxlen) { struct seq_oss_readq *q; - if ((q = kzalloc(sizeof(*q), GFP_KERNEL)) == NULL) { - pr_err("ALSA: seq_oss: can't malloc read queue\n"); + q = kzalloc(sizeof(*q), GFP_KERNEL); + if (!q) return NULL; - } - if ((q->q = kcalloc(maxlen, sizeof(union evrec), GFP_KERNEL)) == NULL) { - pr_err("ALSA: seq_oss: can't malloc read queue buffer\n"); + q->q = kcalloc(maxlen, sizeof(union evrec), GFP_KERNEL); + if (!q->q) { kfree(q); return NULL; } diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index 835edc80f918..48e4fe1b68ab 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -106,10 +106,9 @@ snd_seq_oss_synth_probe(struct device *_dev) struct snd_seq_oss_reg *reg = SNDRV_SEQ_DEVICE_ARGPTR(dev); unsigned long flags; - if ((rec = kzalloc(sizeof(*rec), GFP_KERNEL)) == NULL) { - pr_err("ALSA: seq_oss: can't malloc synth info\n"); + rec = kzalloc(sizeof(*rec), GFP_KERNEL); + if (!rec) return -ENOMEM; - } rec->seq_device = -1; rec->synth_type = reg->type; rec->synth_subtype = reg->subtype; @@ -249,7 +248,6 @@ snd_seq_oss_synth_setup(struct seq_oss_devinfo *dp) if (info->nr_voices > 0) { info->ch = kcalloc(info->nr_voices, sizeof(struct seq_oss_chinfo), GFP_KERNEL); if (!info->ch) { - pr_err("ALSA: seq_oss: Cannot malloc voices\n"); rec->oper.close(&info->arg); module_put(rec->oper.owner); snd_use_lock_free(&rec->use_lock); -- 2.3.2

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[alsa-devel] [PATCH] sound/firewire: Fix trivial typos in comments
by Yannick Guerrini 10 Mar '15

10 Mar '15

Change 'propper' to 'proper' Change 'paramters' to 'parameters' Change 'SYT_INTEVAL' to 'SYT_INTERVAL' Change 'aligh'/'alighed' to 'align'/'aligned' Signed-off-by: Yannick Guerrini <yguerrini(a)tomshardware.fr> --- sound/firewire/amdtp.c | 8 ++++---- sound/firewire/fireworks/fireworks_transaction.c | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 5cc356d..e061355 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -166,10 +166,10 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, * One AMDTP packet can include some frames. In blocking mode, the * number equals to SYT_INTERVAL. So the number is 8, 16 or 32, * depending on its sampling rate. For accurate period interrupt, it's - * preferrable to aligh period/buffer sizes to current SYT_INTERVAL. + * preferrable to align period/buffer sizes to current SYT_INTERVAL. * - * TODO: These constraints can be improved with propper rules. - * Currently apply LCM of SYT_INTEVALs. + * TODO: These constraints can be improved with proper rules. + * Currently apply LCM of SYT_INTERVALs. */ err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32); @@ -270,7 +270,7 @@ static void amdtp_read_s32(struct amdtp_stream *s, * @s: the AMDTP stream to configure * @format: the format of the ALSA PCM device * - * The sample format must be set after the other paramters (rate/PCM channels/ + * The sample format must be set after the other parameters (rate/PCM channels/ * MIDI) and before the stream is started, and must not be changed while the * stream is running. */ diff --git a/sound/firewire/fireworks/fireworks_transaction.c b/sound/firewire/fireworks/fireworks_transaction.c index 2a85e42..f550808 100644 --- a/sound/firewire/fireworks/fireworks_transaction.c +++ b/sound/firewire/fireworks/fireworks_transaction.c @@ -13,7 +13,7 @@ * * Transaction substance: * At first, 6 data exist. Following to the data, parameters for each command - * exist. All of the parameters are 32 bit alighed to big endian. + * exist. All of the parameters are 32 bit aligned to big endian. * data[0]: Length of transaction substance * data[1]: Transaction version * data[2]: Sequence number. This is incremented by the device -- 1.9.5.msysgit.0

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[alsa-devel] [USB UAC2] How to route pcm stream for UAC2 virtual sound card for lowest latency?
by Xuebing Wang 10 Mar '15

10 Mar '15

Hi Community, I am developing a commercial product based on Freescale iMX6SL + UAC2. I can get UAC2 working for below command. 'hw:1,0' is the virtual sound card. 'hw:0,0' is the physical sound card. export RATE=44100; arecord -D hw:1,0 -c 2 -r $RATE -f S16_LE | aplay -c 2 -r $RATE -f S16_LE & Any suggestions on what is the best way to get PCM stream from UAC2 virtual sound card and feed to the physical sound card, for the lowest latency purpose? Thanks. -- Xuebing

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