March 2010 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] mx31 snd and mc13783 codec status
by Sascha Hauer 31 Mar '10

31 Mar '10

Hi Valentin, On Tue, Mar 30, 2010 at 06:23:18PM +0200, Valentin Longchamp wrote: > Hi Sascha, > > For one of our experiments where our robots need to produce sound, I > would need a sound driver for mx31moboard platform. > > I have quickly hacked something based on what you had done for pcm037 > with the MC13783 codec in some of your first sound patches. I know that > this is not ready for mainline and I'm still struggling to debug my > code, but I would like to have results quite soon so that my colleague > could see what sounds (especially regarding to the volume) our system is > capable to produce. > > I would like to know if by any chance you have some updates on this code > ? Find the latest version of my code here: The following changes since commit 01e77706cdde7c0b47e5ca1f4284a795504c7c40: Linus Torvalds (1): Merge branch 'for-linus' of git://gitorious.org/linux-omap-dss2/linux are available in the git repository at: git://git.pengutronix.de/git/sha/linux-2.6.git mc13783 Sascha Hauer (3): add a mc13783 codec driver add phycore-mc13783 sound support pcm038: add sound support arch/arm/mach-mx2/mach-pcm038.c | 23 ++- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mc13783.c | 727 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/mc13783.h | 32 ++ sound/soc/imx/Kconfig | 9 + sound/soc/imx/Makefile | 3 + sound/soc/imx/phycore-mc13783.c | 160 +++++++++ 8 files changed, 959 insertions(+), 1 deletions(-) create mode 100644 sound/soc/codecs/mc13783.c create mode 100644 sound/soc/codecs/mc13783.h create mode 100644 sound/soc/imx/phycore-mc13783.c > And do you know if your initial mc13783 codec support coupled with > mx31 had some limitations ? Our setup is quite straightforward, we have > direct connection from the mx31 to the mc13783 on a single SSI. Our board uses both SSI channels of the MC13783 which we then mux into one channel in the DAM unit. I don't know how this affects you. Sascha -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

1 0

[alsa-devel] ASoC updates for 2.6.35
by Mark Brown 30 Mar '10

30 Mar '10

The following changes since commit 923a00427a141a6d1c83f1b151221a2458b2de8d: Tejun Heo (1): ASoC: update gfp/slab.h includes are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.35 Barry Song (1): ASoC: ad193x: move codec register/unregister to bus probe/remove Dan Carpenter (1): ASoC: wm8994: playback => capture Graham Gower (1): ASoC: Fix passing platform_data to ac97 bus users and fix a leak Kuninori Morimoto (4): ASoC: fsi: ensures process inside master lock ASoC: fsi: IRQ related process had be united ASoC: fsi: Add FIFO size calculate ASoC: fsi: Add FSI2 device support Mark Brown (2): Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-2.6.35 Merge branch 'for-2.6.34' into for-2.6.35 sound/soc/codecs/ac97.c | 15 ++-- sound/soc/codecs/ad193x.c | 102 ++++++++++-------------- sound/soc/codecs/wm8994.c | 2 +- sound/soc/sh/Kconfig | 3 +- sound/soc/sh/fsi.c | 191 ++++++++++++++++++++++++++++++++------------- sound/soc/soc-core.c | 3 +- 6 files changed, 192 insertions(+), 124 deletions(-)

2 1

[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 30 Mar '10

30 Mar '10

The following changes since commit 3cc4e53f86dab635166929bfa47cc68d59b28c26: Mark Brown (1): ASoC: Remove BROKEN from i.MX audio after dependencies merged are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 A couple of fairly straightforward bug fixes. The ac97 patch is mildly invasive but it tested OK for me (and should be very obvious if it fails). Dan Carpenter (1): ASoC: wm8994: playback => capture Graham Gower (1): ASoC: Fix passing platform_data to ac97 bus users and fix a leak sound/soc/codecs/ac97.c | 15 +++++++++------ sound/soc/codecs/wm8994.c | 2 +- sound/soc/soc-core.c | 3 ++- 3 files changed, 12 insertions(+), 8 deletions(-)

2 1

[alsa-devel] digiface vs. multiface vs. jack vs. other cards
by Fernando Lopez-Lezcano 30 Mar '10

30 Mar '10

At CCRMA we have several studios with linux computers in them (in addition to computers sprinkled throughout the building). Some of the studios have digiface interfaces, some multiface. All of them are connected to the sound system or mixer digitally through adat lightpipes. Workstations have Delta 66 or Gina3G soundcards. It would be nice for a user to be able to play, say, a stereo Ardour session in any of them without any changes to the session or routing. The digifaces send channels 1-24 out the adat ports. The multifaces on the other hand send 9-16 on the adat port. Deltas and Ginas send out stuff on channels 1-4 or 1-6. You can see the multiface interfaces are a problem. To get sound out of them you have to send on channels 9-16 which is different from _all_ others! I tried several workarounds without much success: a) fix this at the alsa level. Should be the way to go, right? Create an /etc/asound.conf that uses ttable to translate channels for the multifaces so that the digital out is actually sent through channels 1-8 (and create the same name plug interface in all other computers that does nothing so that the user can set that name as the default in qjackctl and will get the same behavior in all computers). Works, but not quite. When I try to use that "fake" plug interface with jack2 I get a warning that I'm using a plug interface (sort of fair) and jack only opens it with _2_ i/o channels (even though the ttable is 18 channels wide!). If I _force_ input and output channels in jack to 18 it does work fine. But now it will _not_ work on other computers that have less (or more) than 18 channels. Argh. b) fix it in hdspmixer. Well, it turns out that hdspmixer does not read the default matrix gains from a file so I can't really override the default. AFAICT the gains are calculated algorithmically in the code and there is no command line option to load presets from a file. Or I could write a small script that will set the right gains for 1-8 to go out through 9-16 but if a user starts hdspmixer that gets undone with no warning. And, of course: c) give up and connect an 8 channel analog snake to the analog outputs of the multifaces and do away with a digital connection to the mixer... Anyone has any other suggestions?? This should be easy to fix, right? :-) -- Fernando

2 1

[alsa-devel] [PATCH] ad193x: move codec register/unregister to bus probe/remove
by Barry Song 29 Mar '10

29 Mar '10

The way i've factored out the bus probe and removal functions so that there's no code in the individual I2C and SPI functions means that the register() and unregister() functions could just be squashed into the bus_probe() and bus_remove() functions. Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/codecs/ad193x.c | 102 +++++++++++++++++++------------------------- 1 files changed, 44 insertions(+), 58 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index d034464..e6fe8ac 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -34,8 +34,6 @@ static const u8 ad193x_reg[AD193X_NUM_REGS] = { static struct snd_soc_codec *ad193x_codec; struct snd_soc_codec_device soc_codec_dev_ad193x; -static int ad193x_register(struct ad193x_priv *ad193x, int bus_type); -static void ad193x_unregister(struct ad193x_priv *ad193x); /* * AD193X volume/mute/de-emphasis etc. controls @@ -289,69 +287,23 @@ static int ad193x_bus_probe(struct device *dev, void *ctrl_data, int bus_type) { struct snd_soc_codec *codec; struct ad193x_priv *ad193x; + int ret; + + if (ad193x_codec) { + dev_err(dev, "Another ad193x is registered\n"); + return -EINVAL; + } ad193x = kzalloc(sizeof(struct ad193x_priv), GFP_KERNEL); if (ad193x == NULL) return -ENOMEM; - codec = &ad193x->codec; - codec->control_data = ctrl_data; - codec->dev = dev; - dev_set_drvdata(dev, ad193x); - return ad193x_register(ad193x, bus_type); -} - -static int ad193x_bus_remove(struct device *dev) -{ - struct ad193x_priv *ad193x = dev_get_drvdata(dev); - - ad193x_unregister(ad193x); - return 0; -} - -static struct snd_soc_dai_ops ad193x_dai_ops = { - .hw_params = ad193x_hw_params, - .digital_mute = ad193x_mute, - .set_tdm_slot = ad193x_set_tdm_slot, - .set_fmt = ad193x_set_dai_fmt, -}; - -/* codec DAI instance */ -struct snd_soc_dai ad193x_dai = { - .name = "AD193X", - .playback = { - .stream_name = "Playback", - .channels_min = 2, - .channels_max = 8, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, - }, - .capture = { - .stream_name = "Capture", - .channels_min = 2, - .channels_max = 4, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, - }, - .ops = &ad193x_dai_ops, -}; -EXPORT_SYMBOL_GPL(ad193x_dai); - -static int ad193x_register(struct ad193x_priv *ad193x, int bus_type) -{ - int ret; - struct snd_soc_codec *codec = &ad193x->codec; - - if (ad193x_codec) { - dev_err(codec->dev, "Another ad193x is registered\n"); - return -EINVAL; - } - + codec = &ad193x->codec; mutex_init(&codec->mutex); + codec->control_data = ctrl_data; + codec->dev = dev; codec->private_data = ad193x; codec->reg_cache = ad193x->reg_cache; codec->reg_cache_size = AD193X_NUM_REGS; @@ -412,14 +364,48 @@ static int ad193x_register(struct ad193x_priv *ad193x, int bus_type) return 0; } -static void ad193x_unregister(struct ad193x_priv *ad193x) +static int ad193x_bus_remove(struct device *dev) { + struct ad193x_priv *ad193x = dev_get_drvdata(dev); + snd_soc_unregister_dai(&ad193x_dai); snd_soc_unregister_codec(&ad193x->codec); kfree(ad193x); ad193x_codec = NULL; + + return 0; } +static struct snd_soc_dai_ops ad193x_dai_ops = { + .hw_params = ad193x_hw_params, + .digital_mute = ad193x_mute, + .set_tdm_slot = ad193x_set_tdm_slot, + .set_fmt = ad193x_set_dai_fmt, +}; + +/* codec DAI instance */ +struct snd_soc_dai ad193x_dai = { + .name = "AD193X", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .ops = &ad193x_dai_ops, +}; +EXPORT_SYMBOL_GPL(ad193x_dai); + static int ad193x_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); -- 1.5.6.3

3 2

[alsa-devel] Volume levels with Geode LX/CS5536
by Valentin 29 Mar '10

29 Mar '10

Hi, the maximum volume level (100%) on my soundcard for the PCM mixer is 12.0dB instead of 0.0dB. Is there a way for me to fix this, or help fixing it? crimsun on freenode/#alsa told me to ask you guys :) My alsa-info.sh output is at http://0au.de/~apo/p/ai.txt Cheers, Valentin

2 1

[alsa-devel] bug in spi and i2c dual buses codecs
by Barry Song 29 Mar '10

29 Mar '10

Hi Mark/Liam, I think there is really a bug in spi and i2c dual buses codecs drivers, for example, for any codec with two buses, if you config kernel like this: SPI MASTER built-in I2C module Codec built-in You will get a link error like this: sound/built-in.o: In function `i2c_add_driver': include/linux/i2c.h:423: undefined reference to `i2c_register_driver' sound/built-in.o: In function `ad193x_modexit': sound/soc/codecs/ad193x.c:537: undefined reference to `i2c_del_driver' make: *** [.tmp_vmlinux1] Error 1 I remember that is also one reason why we have splitted this kind of driver into three parts(codec core, i2c bus, spi bus). In fact, if any bus is compiled as module and code is built-in, link error will be reported. Thanks Barry

3 4

[alsa-devel] Soundcard Detection problem (ALSA regression)
by Gregory T Flynn 29 Mar '10

29 Mar '10

So far I have tested the current stable drive and now the current snapshot, both to no avail. The last ALSA driver version that worked on this sound card was 1.0.18. General Info from the 1.0.18 driver: !!PCI Soundcards installed in the system !!-------------------------------------- 00:10.1 Audio device: nVidia Corporation MCP51 High Definition Audio (rev a2) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:10.1 0403: 10de:026c (rev a2) Subsystem: 1462:373d Currently I'm running Ubuntu 10.04 beta1 and this sounds regression (which was also in 9.10) is my only problem with the new system. So it would be awesome to get solved and packed into the beta before release. more details in bug report: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4967 -- - Greg Flynn

1 0

[alsa-devel] sound: patch_realtek: truncating bits
by Dan Carpenter 29 Mar '10

29 Mar '10

Smatch complains that we are truncating bits here: sound/pci/hda/patch_realtek.c +12459 alc268_aspire_one_speaker_automute(6) warn: value 28800 can't fit into 255 'bits' sound/pci/hda/patch_realtek.c +13482 alc269_quanta_fl1_speaker_automute(6) warn: value 28800 can't fit into 255 'bits' sound/pci/hda/patch_realtek.c +13511 alc269_lifebook_speaker_automute(11) warn: value 28800 can't fit into 255 'bits' sound/pci/hda/patch_realtek.c +13646 alc269_speaker_automute(8) warn: value 28800 can't fit into 255 'bits' The code looks like this: 13643 unsigned char bits; 13644 13645 present = snd_hda_jack_detect(codec, nid); 13646 bits = present ? AMP_IN_MUTE(0) : 0; "bits" is declared as an unsigned char but AMP_IN_MUTE(0) is 0x7080 so we are only using the last 0x80. I couldn't figure out if it was intended. This was first introduced by this commit which change "bits" from unsigned int to unsigned char. commit 60db6b53fb43421beb2ff3fe3e63412bf81620aa Author: Kailang Yang <kailang(a)realtek.com> Date: Tue Aug 26 13:13:00 2008 +0200 ALSA: hda - Add support of Quanta FL1 Added the support of Quanta FL1 with ALC269 code chip. Also a bit space clean-ups. If this is intentional, it would be cleaner to explicitly cast or mask away the bits we don't care about. regards, dan carpenter

2 1

[alsa-devel] [PATCH] ALSA: hda: Use LPIB for ga-ma770-ud3 board
by Daniel Chen 29 Mar '10

29 Mar '10

Hi, Attached is a patch against current sound-2.6 master HEAD (also applicable to 2.6.32.y and 2.6.33.y) that makes the OR's audio more tolerable with PulseAudio. Best, -Dan

2 1