August 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: omap: Fix build errors in ams-delta
by Jarkko Nikula 18 Aug '11

18 Aug '11

Fix "error: too few arguments to function 'ams_delta_set_bias_level'" build errors in ams-delta.c that were introduced after commit d4c6005 ("ASoC: Add context parameter to card DAPM callbacks") by adding dapm context to ams_delta_set_bias_level calls. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Cc: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- sound/soc/omap/ams-delta.c | 6 ++++-- 1 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 30fe0d0..0aa475f 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -514,7 +514,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) } /* Set codec bias level */ - ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY); + ams_delta_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY); /* Add hook switch - can be used to control the codec from userspace * even if line discipline fails */ @@ -649,7 +649,9 @@ static void __exit ams_delta_module_exit(void) ams_delta_hook_switch_gpios); /* Keep modem power on */ - ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY); + ams_delta_set_bias_level(&ams_delta_audio_card, + &ams_delta_audio_card.rtd[0].codec->dapm, + SND_SOC_BIAS_STANDBY); platform_device_unregister(cx20442_platform_device); platform_device_unregister(ams_delta_audio_platform_device); -- 1.7.5.4

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[alsa-devel] [PATCH 1/4] ASoC: DAPM: Allow multiple mixer sources to be routed via the same switch
by Lars-Peter Clausen 18 Aug '11

18 Aug '11

Currently it is only possible to route one source per switch into a mixer. This patch modifies the code, so that it is possible to route multiple sources into a mixer via the same switch. One use-case for this is routing a stereo channel pair into a mono-mixer via the same switch. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c265311..170c4ff 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -443,6 +443,11 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) if (path->name != (char *)w->kcontrol_news[i].name) continue; + if (w->kcontrols[i]) { + path->kcontrol = w->kcontrols[i]; + continue; + } + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + sizeof(struct snd_soc_dapm_widget *), wlist = kzalloc(wlistsize, GFP_KERNEL); @@ -1556,7 +1561,6 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, /* found, now check type */ found = 1; path->connect = connect; - break; } if (found) -- 1.7.2.5

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Re: [alsa-devel] [PATCH] ASoC: sta32x: Move resource allocation and release to the corresponding callback functions
by Liam Girdwood 18 Aug '11

18 Aug '11

On 18/08/11 08:31, Axel Lin wrote: > This patch includes below small fixes: > > 1. Move sta32x_set_bias_level() from sta32x_i2c_remove() to sta32x_remove(). > 2. Remove a redundant regulator_bulk_free() call in sta32x_i2c_remove(), > as we will call regulator_bulk_free() in sta32x_remove(). > 3. Remove unneeded snd_soc_codec_set_drvdata(codec, NULL) in sta32x_i2c_remove. > The i2c core will set the clientdata to NULL. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Acked-by: Liam Girdwood <lrg(a)ti.com>

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Re: [alsa-devel] [PATCH] ASoC: sta32x: Move resource allocation and release to the corresponding callback functions
by Johannes Stezenbach 18 Aug '11

18 Aug '11

On Thu, Aug 18, 2011 at 03:31:04PM +0800, Axel Lin wrote: > This patch includes below small fixes: > > 1. Move sta32x_set_bias_level() from sta32x_i2c_remove() to sta32x_remove(). > 2. Remove a redundant regulator_bulk_free() call in sta32x_i2c_remove(), > as we will call regulator_bulk_free() in sta32x_remove(). > 3. Remove unneeded snd_soc_codec_set_drvdata(codec, NULL) in sta32x_i2c_remove. > The i2c core will set the clientdata to NULL. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > hi, > I don't have this hardware handy. I appreciate if someone can test this patch. > > Axel Your patch looks good, but I also don't have the hardware currently. Maybe Sven or Daniel could test? But even though it's untested: Acked-by: Johannes Stezenbach <js(a)sig21.net> Thanks Johannes > sound/soc/codecs/sta32x.c | 13 ++----------- > 1 files changed, 2 insertions(+), 11 deletions(-) > > diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c > index 3d155f5..5c7def3 100644 > --- a/sound/soc/codecs/sta32x.c > +++ b/sound/soc/codecs/sta32x.c > @@ -812,6 +812,7 @@ static int sta32x_remove(struct snd_soc_codec *codec) > { > struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); > > + sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); > regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); > regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); > > @@ -871,18 +872,8 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, > static __devexit int sta32x_i2c_remove(struct i2c_client *client) > { > struct sta32x_priv *sta32x = i2c_get_clientdata(client); > - struct snd_soc_codec *codec = sta32x->codec; > - > - if (codec) > - sta32x_set_bias_level(codec, SND_SOC_BIAS_OFF); > - > - regulator_bulk_free(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); > - > - if (codec) { > - snd_soc_unregister_codec(&client->dev); > - snd_soc_codec_set_drvdata(codec, NULL); > - } > > + snd_soc_unregister_codec(&client->dev); > kfree(sta32x); > return 0; > } > -- > 1.7.4.1 > > > >

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Re: [alsa-devel] Correct way to set NONAUDIO flag
by Clemens Ladisch 17 Aug '11

17 Aug '11

Pierre-Louis Bossart wrote: > If you try to set the bits with ALSA controls in PulseAudio, somehow the > settings are overwritten when the iec958 device is opened and initialized > with default values. That's how the iec958 device is intended to work. The AES0..3 parameters of the iec958 device are the _only_ way to set these flags. If you don't set them, they get a static default value; see <http://git.alsa-project.org/?p=alsa-lib.git;a=blob;hb=HEAD;f=src/conf/pcm/i…>. If PA wants to set these flags, it must do so when opening the device; any other mechanism will not work. Regards, Clemens

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Re: [alsa-devel] Correct way to set NONAUDIO flag
by Arun Raghavan 17 Aug '11

17 Aug '11

On Wed, 2011-08-17 at 10:31 -0500, Pierre-Louis Bossart wrote: > > > > I'm trying to figure out the right way to set the NONAUDIO flag on > > an > > > > iec958 device. > > > > > > Use the device name "iec958:AES0=6". > > > > Thanks. As I understand it, this forces us to set all the flags instead > > of just the one that we're interested in. Is this the only way? > > I banged my head against the wall on this one for eAC3 support. For some > strange reason my receiver works fine with AC3 without touching the C bits, > but for eAC3 setting the NONAUDIO bit is required. > If you try to set the bits with ALSA controls in PulseAudio, somehow the > settings are overwritten when the iec958 device is opened and initialized > with default values. Clemens is right, setting the AES0=6 in the hdmi > profile proved to be the only solution. It'd be nice to find a solution in > which the iec958 device preserves the AES values set beforehand. That way we > could set the NONAUDIO bit only in passthrough mode. Well, given this is the best way to do it, my plan is to just close (suspend) and reopen (unsuspend) and tack on the the right flags to the device name when we're in passthrough mode. -- Arun

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[alsa-devel] Correct way to set NONAUDIO flag
by Arun Raghavan 17 Aug '11

17 Aug '11

Hello, I'm trying to figure out the right way to set the NONAUDIO flag on an iec958 device. The idea is that when we get a new stream in PulseAudio that signals a non-PCM format, we'd like to set the NONAUDIO flag to let receivers know that this isn't PCM data. I tried doing what iecset does in the PA alsa-sink code but this fails. As far as I can tell, this happens because: (a) if I set the flag before snd_pcm_open(), the AES0 value from iec958.conf gets used, and (b) if I set the flag after snd_pcm_open(), changing the value is no longer permitted. Any help/pointers would be appreciated. Thanks, Arun

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[PATCH 1/1] sgtl5000: fix record unwork issue
by Dong Aisheng 17 Aug '11

17 Aug '11

Signed-off-by: Dong Aisheng <b29396(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 32 ++++++++++++++++++-------------- 1 files changed, 18 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 5a0d8e4..ef52890 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -198,13 +198,13 @@ static int small_pop_event(struct snd_soc_dapm_widget= *w, { switch (event) { case SND_SOC_DAPM_PRE_PMU: - snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); +// snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, +// SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); break; =20 case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_VAG_POWERUP, 0); +// snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, +// SGTL5000_VAG_POWERUP, 0); msleep(400); break; default: @@ -1153,11 +1153,13 @@ static int sgtl5000_set_power_regs(struct snd_soc_c= odec *codec) SGTL5000_VDDC_CHRGPMP_POWERUP, 0); =20 /* VDDC use VDDIO rail */ - lreg_ctrl |=3D SGTL5000_VDDC_ASSN_OVRD; - lreg_ctrl |=3D SGTL5000_VDDC_MAN_ASSN_VDDIO << - SGTL5000_VDDC_MAN_ASSN_SHIFT; +// lreg_ctrl |=3D SGTL5000_VDDC_ASSN_OVRD; +// lreg_ctrl |=3D SGTL5000_VDDC_MAN_ASSN_VDDIO << +// SGTL5000_VDDC_MAN_ASSN_SHIFT; } =20 +// ana_pwr |=3D SGTL5000_VDDC_CHRGPMP_POWERUP | SGTL5000_VAG_POWERUP | SGT= L5000_HP_POWERUP; + ana_pwr |=3D 0x5afb; snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, lreg_ctrl); =20 snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, ana_pwr); @@ -1194,9 +1196,9 @@ static int sgtl5000_set_power_regs(struct snd_soc_cod= ec *codec) else vag =3D (vag - SGTL5000_ANA_GND_BASE) / SGTL5000_ANA_GND_STP; =20 - snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, - vag << SGTL5000_ANA_GND_SHIFT, - vag << SGTL5000_ANA_GND_SHIFT); +// snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, +// vag << SGTL5000_ANA_GND_SHIFT, +/// vag << SGTL5000_ANA_GND_SHIFT); =20 /* set line out VAG to vddio / 2, in range (0.8v, 1.675v) */ vag =3D vddio / 2; @@ -1208,14 +1210,15 @@ static int sgtl5000_set_power_regs(struct snd_soc_c= odec *codec) else vag =3D (vag - SGTL5000_LINE_OUT_GND_BASE) / SGTL5000_LINE_OUT_GND_STP; - +#if 1 snd_soc_update_bits(codec, SGTL5000_CHIP_LINE_OUT_CTRL, - vag << SGTL5000_LINE_OUT_GND_SHIFT | +// vag << SGTL5000_LINE_OUT_GND_SHIFT | SGTL5000_LINE_OUT_CURRENT_360u << SGTL5000_LINE_OUT_CURRENT_SHIFT, - vag << SGTL5000_LINE_OUT_GND_SHIFT | +// vag << SGTL5000_LINE_OUT_GND_SHIFT | SGTL5000_LINE_OUT_CURRENT_360u << SGTL5000_LINE_OUT_CURRENT_SHIFT); +#endif =20 return 0; } @@ -1378,7 +1381,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec= ) =20 snd_soc_write(codec, SGTL5000_CHIP_ANA_CTRL, SGTL5000_HP_ZCD_EN | - SGTL5000_ADC_ZCD_EN); + SGTL5000_ADC_ZCD_EN=20 + | SGTL5000_LINE_OUT_MUTE); =20 snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); =20 --=20 1.7.0.4

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[alsa-devel] How to pair Wine with ALSA? part1: intro & underruns (long)
by Joerg-Cyril.Hoehle＠t-systems.com 17 Aug '11

17 Aug '11

Dear knowledgeable ALSA developers, Wine currently undergoes a rewrite of its audio systems. That's a welcome point in time to reflect on what it needs from ALSA and how it should use the ALSA API. What combination of hw & sw_params makes sense? How to feed ALSA with samples? I've identified 3 uses cases and several other topics to help thinking about the issues: UC1 Intermittent Mouse Clicks UC2 Video UC3 Background Music T1 underrun T2 buffer and period size/time T3 blocking or not T4 mmap Due to size, I'll cover T2-T4 in a later e-mail. UC1 Intermittent Mouse Clicks An app may use the winmm API as follows: 1. open the device 2. from time to time, e.g. on mouse clicks, send wave data. The requirements are: R1 Nothing (or silence) is to be played when the app submits no data. R2 Once the app submits data, it should play ASAP. That is the app's view on it. It seems to translate to silence_threshold=0 and silence_size=boundary, doesn't it? Alternatively, Wine's internal periodic audio worker could as well send silence samples when it receives nothing from the app. If so, I fear additional complexity with snd_pcm_rewind to meet R2, e.g. in case where the app submits samples shortly after wine decided to play some silence. UC2 Video I've not much to say here because I basically ignore how apps synchronize audio with video. GetPosition returns "the stream position of the sample that is currently playing through the speakers". This begs for snd_pcm_delay, not avail_update. An app may select a particular device for output (e.g. 5:1 may not be available with the default mixer). R3 Wine must offer access to several sound cards if available. R4 Try and find hw/sw_params that work with the "default" device as well as individual cards (hw:x or perhaps plughw:x). mmdevapi has a notion of exclusive and shared modes. I expect shared mode to be asked with the "default" device (for mixing to work), and exclusive with others (e.g. 5:1 sound). OTOH, apps may also ask for the default device in exclusive mode, which is said to be granted if no other app is playing sound (much like I can grab hw:0 only when dmix has nothing to do). UC3 Background Music The app ought to send a continuous stream of samples, perhaps mixing sources itself (e.g. what games let the dsound API do). R5 No samples are ever skipped. If late, play late. See R1/R2. I've observed that "dmix"+"pulse" and the "hw:x" devices approach underruns radically differently. In case of a 10s underrun, the former silently skip over the next 10s worth of samples(!) -- if the app does not submit those *fast* to catch up, you'll hear no more sound! -- while the latter immediately write them to their HW buffer. IOW, dmix and pulse streaming semantics do not match winmm nor mmdevapi's semantics. To counter that, I'm considering using: /* call avail_update prior to every write: */ if (snd_pcm_avail_update() > buffer_size) snd_pcm_reset() /* skip over late samples */ /* should be equivalent to snd_pcm_forward(avail) */ snd_pcm_write() However, there's still a short underrun between reset() and write(). Topic T1 underrun I argue that the MS APIs winmm and mmdevapi have almost no notion of underrun during playback. You derive it indirectly at best ("buffer is empty", "all headers returned", "speaker position = amount of samples sent"). R5 GetPosition only counts submitted samples. Silence played during underruns must not count and is not reported to the app. This seems unlike snd_pcm_delay. I've observed snd_pcm_delay grow towards minus infinity during underruns. How to nevertheless make use of the ALSA API to compute that? I'm thinking about err = snd_pcm_avail_delay(&avail,&delay) if (err<0) ...? if (avail > buffer_size) /* underrun => everything was played */ position = sum_of_submitted_samples; else position = some_function_of(delay, submitted_samples) What do you recommend? Thank you for your help and for reading up to this point, Jörg Höhle

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Re: [alsa-devel] snd soc spi read/write
by Scott Jiang 17 Aug '11

17 Aug '11

> > Why are you replying off-list? You should implement 16x8 SPI reads in > the core stuff I'd expect, then disable cache for the CODEC. > Sorry, I just forgot to CC. I wonder if rbtree compress type can support this kind of sparse register? And should the patch aginst 3.0 or 3.2?

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