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

[alsa-devel] dmix - unable to open slave
by Nenad Sljivic 24 Oct '11

24 Oct '11

Hi, I posted the same questions on alsa-user group, but no one could help me, so I might have more luck here. I am trying to enable a software mixer on an MIPS based embedded target board using alsa-lib v1.0.23. When I try to play two audio files using these commands: #aplay -vDplug:dmix song1.wav & #aplay -vDplug:dmix song2.wav mixing works fine. For some strange reason when I try the same thing using the "nenad_dmix" device I am getting the following error: ALSA lib pcm_dmix.c:1068:(snd_pcm_dmix_open) unable to open slave aplay: main:654: audio open error: Invalid argument The error occurs while trying to open device for the second time (song2.wav playback). At the same time, song1.wav plays fine. The content of my .asoundrc is given bellow: pcm.nenad_dmix { type plug slave.pcm "dmixer" } pcm.dmixer { type dmix ipc_key 1024 ipc_key_add_uid false ipc_perm 0666 slave { pcm "hw:0,0" period_time 0 period_size 3840 buffer_size 15360 rate 48000 } } I narrowed down the problem using the strace tool and it showed me this: open("/dev/snd/pcmC0D0p", O_RDWR|O_APPEND|O_NONBLOCK|O_CLOEXEC) = -1 EINVAL (Invalid argument) What can be the difference between using the plug:dmix and some virtual device like nenad_dmix in this case ?? Could some of you help me to resolve this issue? Thanks, Nenad

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Re: [alsa-devel] Is locking problem in snd_intel8x0_pcm_pointer() ?
by Takashi Iwai 24 Oct '11

24 Oct '11

At Mon, 24 Oct 2011 15:48:44 +0400, Konstantin Ozerkov wrote: > > Seems that locking is broken: snd_intel8x0_pcm_pointer() uses spin_lock()/spin_unlock() semantic > for lock which is shared in ISR ( snd_intel8x0_update). As result we got possible deadlock inside ISR > on UP system. And needed special kludge inside snd_intel8x0_pcm_pointer() to prevent moving > backward (really this is race with snd_intel8x0_update). Hm, could you elaborate how can it a problem? snd_intel8x0_update() releases the lock before calling snd_pcm_period_update() that calls the pointer callback. Takashi

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Re: [alsa-devel] [PATCH] Add SoundCard driver for OKI SEMICONDUCTOR ML7213 IOH
by Mark Brown 24 Oct '11

24 Oct '11

On Mon, Oct 24, 2011 at 09:12:42PM +0900, Tomoya MORINAGA wrote: > 1. PCI interface function. > Any current ASoC drivers don't have PCI interface function. > So I don't know where the function should be in machine driver or > platform driver. It depends on what the driver is for. Probably you want a driver which is some combination of machine driver and the various drivers that are normally part of the SoC - whatever roles in the system are filled by this hardware the driver ought to register subsystem drivers for those roles. > 2. Register Access > Can platform driver access register ? > According to the soc document, platform driver must not access hardware, > however, some drivers looks accessing their hardware. What makes you say this? A driver that can't access hardware would be rather useless...

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[alsa-devel] [PATCH 1/3] ASoC: wm8940: Fix setting PLL Output clock division ratio
by Axel Lin 24 Oct '11

24 Oct '11

According to the datasheet: The PLL Output clock division ratio is controlled by BIT[5:4] of WM8940_GPIO register(08h). Current code read/write the WM8940_ADDCNTRL(07h) register which is wrong. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8940.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index a4abfdf..3cc3bce 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -627,8 +627,8 @@ static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai, ret = snd_soc_write(codec, WM8940_CLOCK, reg | (div << 5)); break; case WM8940_OPCLKDIV: - reg = snd_soc_read(codec, WM8940_ADDCNTRL) & 0xFFCF; - ret = snd_soc_write(codec, WM8940_ADDCNTRL, reg | (div << 4)); + reg = snd_soc_read(codec, WM8940_GPIO) & 0xFFCF; + ret = snd_soc_write(codec, WM8940_GPIO, reg | (div << 4)); break; } return ret; -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: wm8996: Avoid a redundant i2c_get_clientdata call in wm8996_i2c_remove
by Axel Lin 24 Oct '11

24 Oct '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8996.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 645c980..32324c9 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -3144,7 +3144,7 @@ static __devexit int wm8996_i2c_remove(struct i2c_client *client) snd_soc_unregister_codec(&client->dev); if (wm8996->pdata.ldo_ena > 0) gpio_free(wm8996->pdata.ldo_ena); - kfree(i2c_get_clientdata(client)); + kfree(wm8996); return 0; } -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: wm8400: Fix setting Fout clock divider for FLL Control 4
by Axel Lin 24 Oct '11

24 Oct '11

What we want here is to clear the WM8400_FLL_OUTDIV_MASK bits then OR with factors.outdiv. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8400.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index dc13be2..f29bc26 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1059,7 +1059,7 @@ static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, wm8400_write(codec, WM8400_FLL_CONTROL_3, factors.n); reg = wm8400_read(codec, WM8400_FLL_CONTROL_4); - reg &= WM8400_FLL_OUTDIV_MASK; + reg &= ~WM8400_FLL_OUTDIV_MASK; reg |= factors.outdiv; wm8400_write(codec, WM8400_FLL_CONTROL_4, reg); -- 1.7.5.4

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[alsa-devel] [PATCH] ALSA: HDA: Add new revision for ALC662
by David Henningsson 24 Oct '11

24 Oct '11

We found a new chip, with Vendor Id: 0x10ec0662 and Revision Id: 0x100300. It seems to work well with patch_alc662, but I don't know if patch_alc882 had been better, and actually, I don't know if there's much of a difference between the two patch functions these days anyway. I'm cc:ing Kailang in case there's more to it (codec renaming, processing coefficients etc) that we don't know of. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] Use of _hint() functions and older machines
by Randell Jesup 24 Oct '11

24 Oct '11

[ I initially posted this to the -users list, but it may be more appropriate here ] At Mozilla, we're in the process of adding support for WebRTC (http://webrtc.org/), which is being standardized by the IETF (their part is 'rtcweb'), and the W3C. This adds real-time audio and video (and data) communication to browsers, peer-to-peer over encrypted channels. We have a sound library that can load either Pulse or Alsa. However, for Alsa, it wants to look at snd_device_name_hint() and also _get_hint() and _free_hint(). It lazy-binds to libasound, so it will dlopen() it and then dlsym() all the symbols it uses; if any fail it unloads the lib and says it's not there. It uses the hint functions to build a device list, for example for presenting to the user. I have two problems: 1) Firefox is build on machines configured with I believe Centos5, and I'm told the machines run Alsa 1.0.12, while the hints() functions were added in 1.0.14 (released June 2007). Right now I can't build release or 'try' builds on the build servers because of this. 2) We'd like to run on older machines if possible, and official release builds are made on those servers. On older machines, _hint() aren't available, so even if I make them optional to dlsym-loading, I would need some other method to get the information I assume using older, now-deprecated-or-gone interfaces. Any suggestions? Tips? See http://code.google.com/p/webrtc/source/browse/trunk/src/modules/audio_devic… for how they're used currently. Thanks! -- Randell Jesup randell-ietf(a)jesup.org

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[alsa-devel] Register cache different from the real register values
by Leon Romanovsky 24 Oct '11

24 Oct '11

Hi, I need your suggestions, I'm trying to add new codec to use the asoc subsystem, all read/write operations done via standard snd_soc_* functions with cache enabled, but register cache is different from the real register values. Can you point me to what can I check? Our code is based on 2.6.38 release. Thanks -- Leon Romanovsky | Independent Linux Consultant www.leon.nu | leon(a)leon.nu

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[alsa-devel] [PATCH v2 0/2] ASoC: core: Ignoring pmdown_time for playback stream power down
by Peter Ujfalusi 24 Oct '11

24 Oct '11

Hello Mark, Changes since v1 (RFC): - move the flag to per codec (snd_soc_codec) - twl6040 to request to ignore the pmdown_time Add support in core to ignore the pmdown_time when stopping the playback stream. This will inline the DAPM stream stop event, so it is going to be executed without delay at stream close time (for playback stream). Backround: with the McPDM protocol we are sendning not only the pure audio stream, but OMAP McPDM also transmits additional information (for example offset cancellation). If McPDM is stopped prior to the DAC this information will be not sent to the codec, which can result noise rendered by the twl6040 codec. Regards, Peter --- Peter Ujfalusi (2): ASoC: core: Add flag to ignore pmdown_time at pcm_close ASoC: twl6040: Request core to inline the DAPM sequence include/sound/soc.h | 1 + sound/soc/codecs/twl6040.c | 1 + sound/soc/soc-pcm.c | 15 +++++++++++---- 3 files changed, 13 insertions(+), 4 deletions(-) -- 1.7.7

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