September 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: ASoC: add DT bindings for CS4271
by Daniel Mack 28 Sep '12

28 Sep '12

Apart from pure matching, the bindings also support setting the the reset gpio line. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Alexander Sverdlin <subaparts(a)yandex.ru> --- Documentation/devicetree/bindings/sound/cs4271.txt | 36 ++++++++++++++++++++++ sound/soc/codecs/cs4271.c | 24 +++++++++++++-- 2 files changed, 57 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cs4271.txt diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt new file mode 100644 index 0000000..c81b5fd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs4271.txt @@ -0,0 +1,36 @@ +Cirrus Logic CS4271 DT bindings + +This driver supports both the I2C and the SPI bus. + +Required properties: + + - compatible: "cirrus,cs4271" + +For required properties on SPI, please consult +Documentation/devicetree/bindings/spi/spi-bus.txt + +Required properties on I2C: + + - reg: the i2c address + + +Optional properties: + + - reset-gpio: a GPIO spec to define which pin is connected to the chip's + !RESET pin + +Examples: + + codec_i2c: cs4271@10 { + compatible = "cirrus,cs4271"; + reg = <0x10>; + reset-gpio = <&gpio 23 0>; + }; + + codec_spi: cs4271@0 { + compatible = "cirrus,cs4271"; + reg = <0x0>; + reset-gpio = <&gpio 23 0>; + spi-max-frequency = <6000000>; + }; + diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 9eb01d7..f994af3 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -22,12 +22,14 @@ #include <linux/module.h> #include <linux/slab.h> #include <linux/delay.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/tlv.h> #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/spi/spi.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/tlv.h> #include <sound/cs4271.h> #define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ @@ -458,6 +460,14 @@ static int cs4271_soc_resume(struct snd_soc_codec *codec) #define cs4271_soc_resume NULL #endif /* CONFIG_PM */ +#ifdef CONFIG_OF +static const struct of_device_id cs4271_dt_ids[] = { + { .compatible = "cirrus,cs4271", }, + { } +}; +MODULE_DEVICE_TABLE(of, cs4271_dt_ids); +#endif + static int cs4271_probe(struct snd_soc_codec *codec) { struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); @@ -465,6 +475,12 @@ static int cs4271_probe(struct snd_soc_codec *codec) int ret; int gpio_nreset = -EINVAL; +#ifdef CONFIG_OF + if (of_match_device(cs4271_dt_ids, codec->dev)) + gpio_nreset = of_get_named_gpio(codec->dev->of_node, + "reset-gpio", 0); +#endif + if (cs4271plat && gpio_is_valid(cs4271plat->gpio_nreset)) gpio_nreset = cs4271plat->gpio_nreset; @@ -569,6 +585,7 @@ static struct spi_driver cs4271_spi_driver = { .driver = { .name = "cs4271", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(cs4271_dt_ids), }, .probe = cs4271_spi_probe, .remove = __devexit_p(cs4271_spi_remove), @@ -608,6 +625,7 @@ static struct i2c_driver cs4271_i2c_driver = { .driver = { .name = "cs4271", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(cs4271_dt_ids), }, .id_table = cs4271_i2c_id, .probe = cs4271_i2c_probe, -- 1.7.11.4

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Re: [alsa-devel] ASoC: dapm: Allow regulators to bypass as well as disable when idle
by Dan Carpenter 28 Sep '12

28 Sep '12

Hi Mark, I had a question about patch c05b84d14b23: "ASoC: dapm: Allow regulators to bypass as well as disable when idle" from Sep 7, 2012. 1022 if (SND_SOC_DAPM_EVENT_ON(event)) { 1023 if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 1024 ret = regulator_allow_bypass(w->regulator, true); 1025 if (ret != 0) 1026 dev_warn(w->dapm->dev, 1027 "Failed to bypass %s: %d\n", 1028 w->name, ret); 1029 } 1030 1031 return regulator_enable(w->regulator); 1032 } else { 1033 if (w->invert & SND_SOC_DAPM_REGULATOR_BYPASS) { ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ Right now ->invert is a single bit bitfield. We mostly test "if (w->invert) {...". Are we planning to add some more bits in there? 1034 ret = regulator_allow_bypass(w->regulator, false); 1035 if (ret != 0) 1036 dev_warn(w->dapm->dev, 1037 "Failed to unbypass %s: %d\n", 1038 w->name, ret); 1039 } regards, dan carpenter

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Re: [alsa-devel] [PATCH V2] ASoC: SAMSUNG: Add DT support for i2s
by Stephen Warren 28 Sep '12

28 Sep '12

On 09/27/2012 04:25 AM, Padmavathi Venna wrote: > Add support for device based discovery. > +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt > +i2s@03830000 { > + compatible = "samsung,samsung-i2s"; > + reg = <0x03830000 0x100>; > + tx-dma-channel-secondary = <&pdma0 8>; > + tx-dma-channel = <&pdma0 10>; > + rx-dma-channel = <&pdma0 9>; > + supports-6ch; > + supports-rstclr; > + supports-secdai; Those 6 properties are custom to this binding so should have a vendor prefix, e.g. "samsung,supports-6ch". Also, I wonder if this shouldn't wait for the forthcoming standardized DT DMA bindings (or perhaps document custom bindings that are identical to the new standardized bindings)? Otherwise, the DMA stuff will all have to be redefined again. Apparently it's hoped the bindings will be in 3.7, or if not there, in 3.8.

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Re: [alsa-devel] [PATCH V2] ASoC: SAMSUNG: Add DT support for i2s
by Sangbeom Kim 28 Sep '12

28 Sep '12

Hi! On Thu, Sep 27, 2012 at 07:26 PM, Padmavathi Venna wrote: > Subject: [PATCH V2] ASoC: SAMSUNG: Add DT support for i2s > > Add support for device based discovery. > > Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> Please describe the difference between v1 and v2. Thanks, Sangbeom.

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[alsa-devel] [PATCH] pcm: support for audio timestamps
by Pierre-Louis Bossart 28 Sep '12

28 Sep '12

add new snd_pcm_status_get_audio_htstamp() routine to query the audio timestamps provided by the kernel. This change provides applications with better ways to track elapsed time. Before this patch, applications would subtract queued samples (delay) from written samples, resulting in a 1-2 sample error. Also add snd_pcm_hw_params_supports_audio_wallclock_ts() to query what the hardware supports. TODO: check protocol compatibility? Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/pcm.h | 2 ++ include/sound/asound.h | 7 +++++-- src/pcm/pcm.c | 31 +++++++++++++++++++++++++++++++ test/audio_time.c | 27 +++++++++++++++++---------- 4 files changed, 55 insertions(+), 12 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 4997557..8d08471 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -550,6 +550,7 @@ int snd_pcm_hw_params_is_half_duplex(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_is_joint_duplex(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_can_sync_start(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_can_disable_period_wakeup(const snd_pcm_hw_params_t *params); +int snd_pcm_hw_params_supports_audio_wallclock_ts(const snd_pcm_hw_params_t *params); int snd_pcm_hw_params_get_rate_numden(const snd_pcm_hw_params_t *params, unsigned int *rate_num, unsigned int *rate_den); @@ -858,6 +859,7 @@ void snd_pcm_status_get_trigger_tstamp(const snd_pcm_status_t *obj, snd_timestam void snd_pcm_status_get_trigger_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); void snd_pcm_status_get_tstamp(const snd_pcm_status_t *obj, snd_timestamp_t *ptr); void snd_pcm_status_get_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); +void snd_pcm_status_get_audio_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr); snd_pcm_sframes_t snd_pcm_status_get_delay(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail(const snd_pcm_status_t *obj); snd_pcm_uframes_t snd_pcm_status_get_avail_max(const snd_pcm_status_t *obj); diff --git a/include/sound/asound.h b/include/sound/asound.h index 07c03fa..109302e 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -160,7 +160,7 @@ enum { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 10) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 11) typedef unsigned long sndrv_pcm_uframes_t; typedef long sndrv_pcm_sframes_t; @@ -285,6 +285,7 @@ enum sndrv_pcm_subformat { #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ +#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ enum sndrv_pcm_state { SNDRV_PCM_STATE_OPEN = 0, /* stream is open */ @@ -426,7 +427,8 @@ struct sndrv_pcm_status { sndrv_pcm_uframes_t avail_max; /* max frames available on hw since last status */ sndrv_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ int suspended_state; /* suspended stream state */ - unsigned char reserved[60]; /* must be filled with zero */ + struct timespec audio_tstamp; /* from sample counter or wall clock */ + unsigned char reserved[60-sizeof(struct timespec)]; /* must be filled with zero */ }; struct sndrv_pcm_mmap_status { @@ -435,6 +437,7 @@ struct sndrv_pcm_mmap_status { sndrv_pcm_uframes_t hw_ptr; /* RO: hw ptr (0...boundary-1) */ struct timespec tstamp; /* Timestamp */ int suspended_state; /* RO: suspended stream state */ + struct timespec audio_tstamp; /* from sample counter or wall clock */ }; struct sndrv_pcm_mmap_control { diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 18b43b3..748e8a6 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -3123,6 +3123,26 @@ int snd_pcm_hw_params_can_disable_period_wakeup(const snd_pcm_hw_params_t *param } /** + * \brief Check if hardware supports audio wallclock timestamps + * \param params Configuration space + * \retval 0 Hardware doesn't support audio wallclock timestamps + * \retval 1 Hardware supports audio wallclock timestamps + * + * This function should only be called when the configuration space + * contains a single configuration. Call #snd_pcm_hw_params to choose + * a single configuration from the configuration space. + */ +int snd_pcm_hw_params_supports_audio_wallclock_ts(const snd_pcm_hw_params_t *params) +{ + assert(params); + if (CHECK_SANITY(params->info == ~0U)) { + SNDMSG("invalid PCM info field"); + return 0; /* FIXME: should be a negative error? */ + } + return !!(params->info & SNDRV_PCM_INFO_HAS_WALL_CLOCK); +} + +/** * \brief Get rate exact info from a configuration space * \param params Configuration space * \param rate_num Pointer to returned rate numerator @@ -6213,6 +6233,17 @@ void snd_pcm_status_get_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *p use_default_symbol_version(__snd_pcm_status_get_htstamp, snd_pcm_status_get_htstamp, ALSA_0.9.0rc8); /** + * \brief Get "now" hi-res audio timestamp from a PCM status container + * \param obj pointer to #snd_pcm_status_t + * \param ptr Pointer to returned timestamp + */ +void snd_pcm_status_get_audio_htstamp(const snd_pcm_status_t *obj, snd_htimestamp_t *ptr) +{ + assert(obj && ptr); + *ptr = obj->audio_tstamp; +} + +/** * \brief Get delay from a PCM status container (see #snd_pcm_delay) * \return Delay in frames * diff --git a/test/audio_time.c b/test/audio_time.c index a910783..de8df17 100644 --- a/test/audio_time.c +++ b/test/audio_time.c @@ -33,6 +33,7 @@ long long timediff(snd_htimestamp_t t1, snd_htimestamp_t t2) void gettimestamp(snd_pcm_t *handle, snd_htimestamp_t *timestamp, snd_htimestamp_t *trigger_timestamp, + snd_htimestamp_t *audio_timestamp, snd_pcm_uframes_t *avail, snd_pcm_sframes_t *delay) { int err; @@ -45,6 +46,7 @@ void gettimestamp(snd_pcm_t *handle, snd_htimestamp_t *timestamp, } snd_pcm_status_get_trigger_htstamp(status, trigger_timestamp); snd_pcm_status_get_htstamp(status, timestamp); + snd_pcm_status_get_audio_htstamp(status, audio_timestamp); *avail = snd_pcm_status_get_avail(status); *delay = snd_pcm_status_get_delay(status); } @@ -65,6 +67,7 @@ int main(void) snd_pcm_sframes_t frames; snd_htimestamp_t tstamp_c, tstamp_p; snd_htimestamp_t trigger_tstamp_c, trigger_tstamp_p; + snd_htimestamp_t audio_tstamp_c, audio_tstamp_p; unsigned char buffer_p[PERIOD*4*4]; unsigned char buffer_c[PERIOD*4*4]; @@ -202,26 +205,30 @@ int main(void) frame_count_p += frames; #if defined(TRACK_PLAYBACK) - gettimestamp(handle_p, &tstamp_p, &trigger_tstamp_p, &avail_p, &delay_p); + gettimestamp(handle_p, &tstamp_p, &trigger_tstamp_p, &audio_tstamp_p, &avail_p, &delay_p); curr_count_p = frame_count_p - delay_p; /* written minus queued */ - printf("playback: systime: %lli nsec, sample time %lli nsec \tsystime delta %lli \n", - timediff(tstamp_p,trigger_tstamp_p), - (long long)round(((float)curr_count_p * 1000000000.0 / 48000.0)), - timediff(tstamp_p, trigger_tstamp_p) - (long long)round((double)curr_count_p * 1000000000.0 / 48000.0) + /* uncomment next line to see difference between old sample count and audio timestamps. + printf("curr_count %lli driver count %lli\n", (long long)curr_count_p * 1000000000LL / 48000, timestamp2ns(audio_tstamp_p)); + */ + + printf("playback: systime: %lli nsec, audio time %lli nsec, \tsystime delta %lli\n", + timediff(tstamp_p, trigger_tstamp_p), + timestamp2ns(audio_tstamp_p), + timediff(tstamp_p, trigger_tstamp_p) - timestamp2ns(audio_tstamp_p) ); #endif #if defined(TRACK_CAPTURE) - gettimestamp(handle_c, &tstamp_c, &trigger_tstamp_c, &avail_c, &delay_c); + gettimestamp(handle_c, &tstamp_c, &trigger_tstamp_c, &audio_tstamp_c, &avail_c, &delay_c); curr_count_c = frame_count_c + delay_c; /* read plus queued */ - printf("\t capture: systime: %lli nsec, sample time %lli nsec \tsystime delta %lli \n", - timediff(tstamp_c,trigger_tstamp_c), - (long long)round(((float)curr_count_c * 1000000000.0 / 48000.0)), - timediff(tstamp_c, trigger_tstamp_c) - (long long)round((double)curr_count_c * 1000000000.0 / 48000.0) + printf("\t capture: systime: %lli nsec, audio time %lli nsec, \tsystime delta %lli\n", + timediff(tstamp_c, trigger_tstamp_c), + timestamp2ns(audio_tstamp_c), + timediff(tstamp_c, trigger_tstamp_c) - timestamp2ns(audio_tstamp_c) ); #endif -- 1.7.9.5

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[alsa-devel] setup_dig_out_stream() resets the SPDIF steam causing clicks
by Laurence Darby 28 Sep '12

28 Sep '12

Hi, I'm playing audio via SPDIF, with a Realtek codec, and there's a problem that whenever audio is started, or seeking in every music player I tried, the DAC lost sync briefly, which made it 'click' when it muted and unmuted itself. I recently upgraded the DAC, and this problem only happened with the new DAC, but not the old one, I don't know (or really care) why - the old DAC had much worse problems. Here's what I did to fix this: diff -up linux-3.5.4/sound/pci/hda/hda_codec.c.orig linux-3.5.4/sound/pci/hda/hda_codec.c --- linux-3.5.4/sound/pci/hda/hda_codec.c.orig 2012-09-28 00:13:25.000000000 +0100 +++ linux-3.5.4/sound/pci/hda/hda_codec.c 2012-09-28 00:39:31.000000000 +0100 @@ -4693,9 +4693,11 @@ static void setup_dig_out_stream(struct format); } /* turn on again (if needed) */ - if (codec->spdif_status_reset && (spdif->ctls & AC_DIG1_ENABLE)) + if (codec->spdif_status_reset && (spdif->ctls & AC_DIG1_ENABLE)) { set_dig_out_convert(codec, nid, spdif->ctls & 0xff, -1); + codec->spdif_status_reset=0; + } } static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid) That makes SPDIF turn off and on just once when sound is first played (i.e. it clicks once then, then no more). In patch_realtek.c it's got this: /* FIXME: do we need this for all Realtek codec models? */ codec->spdif_status_reset = 1; If I set that to 0, then the left channel doesn't initialize and is silent, so the reset really does need to happen once (for my realtek model anyway, I don't know how to find out what that would be?) So, is the above an acceptable fix? Regards, Laurence

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[alsa-devel] [PATCH] ASoC: wm5110: Adding missing volume update bits.
by Mark Brown 27 Sep '12

27 Sep '12

From: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> The volume update bits were being set on all but one input and one output. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index b99451e..9211e41 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -916,6 +916,8 @@ static unsigned int wm5110_digital_vu[] = { ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_ADC_DIGITAL_VOLUME_3L, ARIZONA_ADC_DIGITAL_VOLUME_3R, + ARIZONA_ADC_DIGITAL_VOLUME_4L, + ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, @@ -927,6 +929,8 @@ static unsigned int wm5110_digital_vu[] = { ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_DAC_DIGITAL_VOLUME_5L, ARIZONA_DAC_DIGITAL_VOLUME_5R, + ARIZONA_DAC_DIGITAL_VOLUME_6L, + ARIZONA_DAC_DIGITAL_VOLUME_6R, }; static struct snd_soc_codec_driver soc_codec_dev_wm5110 = { -- 1.7.10.4

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[alsa-devel] ASoC updates for 3.6
by Mark Brown 27 Sep '12

27 Sep '12

The following changes since commit 5698bd757d55b1bb87edd1a9744ab09c142abfc2: Linux 3.6-rc6 (2012-09-16 14:58:51 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.6 for you to fetch changes up to d0e12f3ff3472cbd8f52d3c0e6ee07a841787c40: ASoC: wm2000: Correct register size (2012-09-26 12:06:20 +0100) Takashi is on vacation now so he asked me to send directly, it's a pretty bad bug with low regression risk. ---------------------------------------------------------------- ASoC: Updates for 3.6 One small and obvious driver-specific fix. ---------------------------------------------------------------- Mark Brown (1): ASoC: wm2000: Correct register size sound/soc/codecs/wm2000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-)

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[alsa-devel] ASoC on today's linux-next: dma_alloc_writecombine() failing at least on OMAP platforms
by Peter Ujfalusi 27 Sep '12

27 Sep '12

Hello, I tried today's linux-next on OMAP platform and audio is not probing because dma_alloc_writecombine() fails to get the area. My linux-next from two days ago works OK and I don't see anything directly connected to OMAP which can cause this failure. Can someone try linux-next on non OMAP platform to see if dma_alloc_writecombine() works there? Meanwhile I'll try to find the reason why it fails for me. Thanks, Péter

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[alsa-devel] OMAP4 McBSP AC97 link
by Sebastian 27 Sep '12

27 Sep '12

Hi All, i have to support a 2nd Audio Codec with AC97 Interface on McBSP Port2. Processor: OMAP4460 If any one have any idea on how to go about this. Please let me know. BR, Sebastian

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