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

[alsa-devel] Opening a PCM device in shared mode
by Sayem Ahmed 20 Nov '11

20 Nov '11

Hello all, I am a beginner in ALSA API. I am developing a sound application where I need to open a PCM device for recording/playback audio data. I want to open this device in shared mode so that other applications can use the same device. I have downloaded the example ALSA program from here <http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html> and run it on my PC. It works fine when it runs alone. But when I start a song on a media player and then run this example, it shows an error saying - |Playback open error: Device or resource busy | Looking at the source code of this example I could say that thesnd_pcm_open <http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#gb7eaaec6f…>function at line 882 is throwing this error. It finds the device busy since another process is currently using it. I also tried the reverse way - first started the example then tried to start a song. In this case the media player stays idle, showing "idle" just beside the progress bar (I am using Banshee). I am assuming that snd_pcm_open gains exclusive right of the device resource so that no other process can use it. But I don't want that. I want to play sound to a audio device without requiring any exclusive rights so that other processes in the PC can share the same device for outputting audio data. How can I do that? How can I open a PCM device so that other processes can also share the same device? P.S. : I have two sound cards on my system, one is built-in in motherboard, another one is a USB Card. I tried opening the USB (by specifying "plughw:1,0" as argument in the snd_pcm_open) card and then got this error. Thank you in advance. Regards, Sayem Ahmed

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[alsa-devel] [PATCH 0/5] RFC for snd-usb: rework usb endpoint logic
by Daniel Mack 18 Nov '11

18 Nov '11

It's been a while since the last round of patches for this topic, so here is a new version. In contrast to earlier versions, I did another bunch of cleanips and can now use input and output simultaniously. The patches that follow are structured in a way that should make reviews easier, as they first implement a new streaming model while leaving the old one working, and then switch to it in a separate patch. This way, the series is also fully bisectable. The problems that remain is a regression in handling data streams with feedback endpoints in full duplex mode and a hard lockup when changing audio parameters while two streams are running. I guess the latter is caused by missing locking. I didn't sign-off the patches on purpose, as I would really like to get them reviewed before they go in. Can people have a look and state whether the whole idea is at all sane? Thanks a lot, Daniel Daniel Mack (5): ALSA: snd-usb: implement new endpoint streaming model ALSA: snd-usb: switch over to new endpoint streaming logic ALSA: snd-usb: remove old streaming logic ALSA: snd-usb: set MAX_URBS to 16 ALSA: snd-usb: add support for implicit feedback sound/usb/card.c | 6 +- sound/usb/card.h | 56 ++- sound/usb/endpoint.c | 1347 ++++++++++++++++++++++++-------------------------- sound/usb/endpoint.h | 31 +- sound/usb/pcm.c | 412 +++++++++++++--- sound/usb/stream.c | 31 +- sound/usb/usbaudio.h | 1 + 7 files changed, 1081 insertions(+), 803 deletions(-) -- 1.7.5.4

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[alsa-devel] dmix issue when used with hot-plugged devices
by Arnaud Mouiche 18 Nov '11

18 Nov '11

Hi all, I'm using USB audio devices with dmix. I notice that the system can lock when the device gets unplugged. ex: $ aplay -D "dmix device on top of usb device" -f S16_LE /dev/zero now unplug the USB device, and aplay is now waiting forever. Two root causes of the problem in alsa-lib, with some proposed patch (I'm not able to upgrade to upstream, but I didn't see any change which can fix my issue) A) the sound timer used by dmix doesn't get the notification when the card was removed. so the poll done to wait is locking here is a patch to solve this: ________________________________________________________________________ commit 2ac49899d6a45ef594a99ad409a378119f0fa157 Author: Arnaud Mouiche <arnaud.mouiche(a)invoxia.com> Date: Fri Nov 18 15:03:29 2011 +0100 timer: use SND_TIMER_EVENT_MSTOP as interesting filter for timers, otherwise we miss the event when the hotplug device is removed and we set a timer of it diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 88609e5..c38da70 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -1137,6 +1137,7 @@ int snd_pcm_direct_initialize_poll_fd(snd_pcm_direct_t *dmix) dmix->timer_events = (1<<SND_TIMER_EVENT_MSUSPEND) | (1<<SND_TIMER_EVENT_MRESUME) | + (1<<SND_TIMER_EVENT_MSTOP) | (1<<SND_TIMER_EVENT_STOP); /* ________________________________________________________________________ B) when snd_pcm_direct_poll_revents() check for snd_pcm_state(dmix->spcm), it doesn't care about negative values so it just consider POLLERR should not be set, and on return, we can be sure to poll again I see 2 issues here: 1)how can snd_pcm_state returns negatives values (simply consider that for gcc, "if (snd_pcm_state() < 0)" can't be false So may be "snd_pcm_state" should check for negative values, and return an associated ALSA_state (like SND_PCM_STATE_DISCONNECTED) 2) POLLERR, at least, should be set for negative values: here is the patch for the second: ________________________________________________________________________ commit 4463226b9e14e17e84492c5d59c0920876b614c0 Author: Arnaud Mouiche <arnaud.mouiche(a)invoxia.com> Date: Fri Nov 18 15:00:36 2011 +0100 snd_pcm_direct_poll_revents set POLLERR is the snd_pcm_state result is invalid this happened for example when the real pcm device (in case of dmix) has been removed. snd_pcm_state(dmix->spcm) returns -ENODEV diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 0a9047d..227602f 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -548,6 +548,7 @@ int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned in snd_pcm_direct_t *dmix = pcm->private_data; unsigned short events; int empty = 0; + int state; assert(pfds && nfds == 1 && revents); events = pfds[0].revents; @@ -563,28 +564,33 @@ int snd_pcm_direct_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, unsigned in } empty = avail < pcm->avail_min; } - switch (snd_pcm_state(dmix->spcm)) { - case SND_PCM_STATE_XRUN: - case SND_PCM_STATE_SUSPENDED: - case SND_PCM_STATE_SETUP: + state = snd_pcm_state(dmix->spcm); + if (((int) state) < 0) { events |= POLLERR; - break; - default: - if (empty) { - snd_pcm_direct_clear_timer_queue(dmix); - events &= ~(POLLOUT|POLLIN); - /* additional check */ - switch (snd_pcm_state(pcm)) { - case SND_PCM_STATE_XRUN: - case SND_PCM_STATE_SUSPENDED: - case SND_PCM_STATE_SETUP: - events |= POLLERR; - break; - default: - break; + } else { + switch (state) { + case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: + case SND_PCM_STATE_SETUP: + events |= POLLERR; + break; + default: + if (empty) { + snd_pcm_direct_clear_timer_queue(dmix); + events &= ~(POLLOUT | POLLIN); + /* additional check */ + switch (snd_pcm_state(pcm)) { + case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: + case SND_PCM_STATE_SETUP: + events |= POLLERR; + break; + default: + break; + } } + break; } - break; } *revents = events; return 0;

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[alsa-devel] newbie questions on alsa architecture (e.g. who reads .asoundrc)
by 俞晓磊 18 Nov '11

18 Nov '11

hello i recently found that a few android devices build their audio stack upon alsa (SGSII for example), and become interested in creating an audio output capture application through alsa api (since android stack does not support such feature) Some tutorials suggest .asoundrc tweaks work for pc, so i want to know what happens when a copy device line is added to the conf file, and is that possible to achieve the goal through plain api calls. i'm ok if a kernel rebuild is necessary, just hoping to dig into alsa a little deeper.

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Re: [alsa-devel] [PATCH v5 1/3] ARM: mxs: add saif clkmux functions
by Dong Aisheng-B29396 18 Nov '11

18 Nov '11

> -----Original Message----- > From: Lothar Waßmann [mailto:LW@KARO-electronics.de] > Sent: Friday, November 18, 2011 4:30 PM > To: Dong Aisheng-B29396 > Cc: Marek Vasut; alsa-devel(a)alsa-project.org; s.hauer(a)pengutronix.de; > broonie(a)opensource.wolfsonmicro.com; w.sang(a)pengutronix.de; Guo Shawn- > R65073; kernel(a)pengutronix.de; u.kleine-koenig(a)pengutronix.de; lrg(a)ti.com; > linux-arm-kernel(a)lists.infradead.org > Subject: RE: [alsa-devel] [PATCH v5 1/3] ARM: mxs: add saif clkmux > functions > > Hi, > > Dong Aisheng-B29396 writes: > [...] > > > > +#define HW_DIGCTL_CTRL 0x0 > > > > +#define BP_DIGCTL_CTRL_SAIF_CLKMUX (10) > > > > > > Do you need the parenthesis above ? > > > > > I'm not sure about his. > > HW_DIGCTL_CTRL is the register offset. > > Do we have to do this? > > > You need parens around macro parameters that are used in an expression > inside the macro. But you never ever need to enclose a bare number in > parens! > Yep! I will remove it. Saving two bytes in code size! :-) Regards Dong Aisheng

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[alsa-devel] smatch stuff in new Cirrus Logic CS42L73 driver
by Dan Carpenter 18 Nov '11

18 Nov '11

Hi Brian, Smatch complains about some stuff in sound/soc/codecs/cs42l73.c. I don't know the code well enough to say what was intended. sound/soc/codecs/cs42l73.c +1030 cs42l73_set_dai_fmt(53) error: inv is never equal to 1024 (wrong type 0 - 255). sound/soc/codecs/cs42l73.c +1032 cs42l73_set_dai_fmt(55) error: inv is never equal to 768 (wrong type 0 - 255). sound/soc/codecs/cs42l73.c +1036 cs42l73_set_dai_fmt(59) error: inv is never equal to 1024 (wrong type 0 - 255). sound/soc/codecs/cs42l73.c +1105 cs42l73_pcm_hw_params(18) info: why not propagate 'mclk_coeff' from cs42l73_get_mclk_coeff() instead of -22? regards, dan carpenter

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[alsa-devel] [PATCH v5 0/3] ARM: mxs: add recording support for saif
by Dong Aisheng 18 Nov '11

18 Nov '11

Changes since v4: * use new added master_id and master_mode in platform_data. The new changes depend on another patch: 0001-ASoC-mxs-saif-remove-function-in-platform_data.patch Changes since v3: * remove unneeded locking according to Sascha. The patches are based on imx-features branch since commit: ca4e419c2. Changes since v2: * separate clkmux code into another patch according to Uwe * add lock according to Wolfram * Other minus fixes suggested by Uwe and Wolfram. * remove Wolfram's fix saif clock setting patch which is in v2 series since Wolfram will reform the clock code and send it out himself. For test purpose, user still needs that patch. People can get it from v2 series. The patch is: [PATCH v2 3/3] arm: mxs: disable clock-gates when setting saif-clocks The patches are based on imx-features branch since commit: f4f01e31835f. Changes since v1: The main changes are move mach-specific code(clkmux in DIGCTL) from saif driver to mach-specific layer based on Wolfram's suggestion. Note that the last patch is a RFC patch and sent out for testing since without that patch the saif may not work. Dong Aisheng (3): ARM: mxs: add saif clkmux functions ARM: mx28evk: add platform data for saif ARM: mx28evk: set a initial clock rate for saif arch/arm/mach-mxs/clock-mx28.c | 38 +++++++++++++++++++++++ arch/arm/mach-mxs/devices-mx28.h | 3 +- arch/arm/mach-mxs/devices/platform-mxs-saif.c | 5 ++- arch/arm/mach-mxs/include/mach/common.h | 1 + arch/arm/mach-mxs/include/mach/devices-common.h | 4 ++- arch/arm/mach-mxs/include/mach/digctl.h | 21 ++++++++++++ arch/arm/mach-mxs/mach-mx28evk.c | 18 +++++++++- 7 files changed, 84 insertions(+), 6 deletions(-) create mode 100644 arch/arm/mach-mxs/include/mach/digctl.h

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[alsa-devel] Request for new alsa-plugins release
by Tony Vroon 17 Nov '11

17 Nov '11

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good morning, Can I ask that a new release of alsa-plugins is rolled, incorporating the PulseAudio stability fixes since 1.0.24? Gentoo has now stabilised a newer version of PulseAudio, and I would have to start rolling a distro patchset otherwise. Generally I prefer to stay as close as possible to an upstream release, to avoid our users seeking support upstream with our downstream bugs. A version 1.0.24.1 of plugins would be ideal. Is this in the cards please? Regards, - -- Tony Vroon UNIX systems administrator London Internet Exchange Ltd, Trinity Court, Trinity Street, Peterborough, PE1 1DA Registered in England number 3137929 E-Mail: tony(a)linx.net -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.18 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk67qjAACgkQp5vW4rUFj5qaRwCgkZRYjqeUssmEXP3oVRWw9AFW pwEAoKMJlUGFGw12MQ9CUpTsYY3VFqb1 =iqMX -----END PGP SIGNATURE-----

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[alsa-devel] [PATCH] ASoC: alc5632: Remove volatile registers from regmap defaults
by Leon Romanovsky 17 Nov '11

17 Nov '11

There is no need to provide defaults for the volatile registers and doing so might cause confusion. Signed-off-by: Leon Romanovsky <leon(a)leon.nu> --- sound/soc/codecs/alc5632.c | 7 ------- 1 files changed, 0 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index c32eade..2d77665 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -36,7 +36,6 @@ * ALC5632 register cache */ static struct reg_default alc5632_reg_defaults[] = { - { 0, 0x59B4 }, /* R0 - Reset */ { 2, 0x8080 }, /* R2 - Speaker Output Volume */ { 4, 0x8080 }, /* R4 - Headphone Output Volume */ { 6, 0x8080 }, /* R6 - AUXOUT Volume */ @@ -52,7 +51,6 @@ static struct reg_default alc5632_reg_defaults[] = { { 34, 0x0000 }, /* R34 - Microphone Control */ { 36, 0x00C0 }, /* R36 - Codec Digital MIC/Digital Boost Control */ - { 38, 0xEF00 }, /* R38 - Power Down Control/Status */ { 46, 0x0000 }, /* R46 - Stereo DAC/Voice DAC/Stereo ADC Function Select */ { 52, 0x8000 }, /* R52 - Main Serial Data Port Control @@ -70,9 +68,7 @@ static struct reg_default alc5632_reg_defaults[] = { { 78, 0xBE3E }, /* R78 - GPIO Pin Polarity */ { 80, 0x0000 }, /* R80 - GPIO Pin Sticky */ { 82, 0x0000 }, /* R82 - GPIO Pin Wake Up */ - { 84, 0x803A }, /* R84 - GPIO Pin Status */ { 86, 0x0000 }, /* R86 - Pin Sharing */ - { 88, 0x0000 }, /* R88 - Over-Temp/Current Status */ { 90, 0x0009 }, /* R90 - Soft Volume Control Setting */ { 92, 0x0000 }, /* R92 - GPIO_Output Pin Control */ { 94, 0x3000 }, /* R94 - MISC Control */ @@ -82,9 +78,6 @@ static struct reg_default alc5632_reg_defaults[] = { { 104, 0x0553 }, /* R104 - Pseudo Stereo and Spatial Effect Block Control */ { 106, 0x0000 }, /* R106 - Private Register Address */ - { 108, 0x0000 }, /* R108 - Private Register Data */ - { 110, 0x0000 }, /* R110 - EQ Control and Status/ADC - HPF Control */ }; /* codec private data */ -- 1.7.3.4

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[alsa-devel] [PATCH v2 1/3] ASoC: alc5632: Remove unrelevant registers and name the relevant
by Leon Romanovsky 17 Nov '11

17 Nov '11

Signed-off-by: Leon Romanovsky <leon(a)leon.nu> --- sound/soc/codecs/alc5632.c | 161 +++++++++++++------------------------------ 1 files changed, 49 insertions(+), 112 deletions(-) diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 6bfbbc7..9660542 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -36,118 +36,55 @@ * ALC5632 register cache */ static struct reg_default alc5632_reg_defaults[] = { - { 0, 0x59B4 }, - { 1, 0x0000 }, - { 2, 0x8080 }, - { 3, 0x0000 }, - { 4, 0x8080 }, - { 5, 0x0000 }, - { 6, 0x8080 }, - { 7, 0x0000 }, - { 8, 0xC800 }, - { 9, 0x0000 }, - { 10, 0xE808 }, - { 11, 0x0000 }, - { 12, 0x1010 }, - { 13, 0x0000 }, - { 14, 0x0808 }, - { 15, 0x0000 }, - { 16, 0xEE0F }, - { 17, 0x0000 }, - { 18, 0xCBCB }, - { 19, 0x0000 }, - { 20, 0x7F7F }, - { 21, 0x0000 }, - { 22, 0x0000 }, - { 23, 0x0000 }, - { 24, 0xE010 }, - { 25, 0x0000 }, - { 26, 0x0000 }, - { 27, 0x0000 }, - { 28, 0x8008 }, - { 29, 0x0000 }, - { 30, 0x0000 }, - { 31, 0x0000 }, - { 32, 0x0000 }, - { 33, 0x0000 }, - { 34, 0x0000 }, - { 35, 0x0000 }, - { 36, 0x00C0 }, - { 37, 0x0000 }, - { 38, 0xEF00 }, - { 39, 0x0000 }, - { 40, 0x0000 }, - { 41, 0x0000 }, - { 42, 0x0000 }, - { 43, 0x0000 }, - { 44, 0x0000 }, - { 45, 0x0000 }, - { 46, 0x0000 }, - { 47, 0x0000 }, - { 48, 0x0000 }, - { 49, 0x0000 }, - { 50, 0x0000 }, - { 51, 0x0000 }, - { 52, 0x8000 }, - { 53, 0x0000 }, - { 54, 0x0000 }, - { 55, 0x0000 }, - { 56, 0x0000 }, - { 57, 0x0000 }, - { 58, 0x0000 }, - { 59, 0x0000 }, - { 60, 0x0000 }, - { 61, 0x0000 }, - { 62, 0x8000 }, - { 63, 0x0000 }, - { 64, 0x0C0A }, - { 65, 0x0000 }, - { 66, 0x0000 }, - { 67, 0x0000 }, - { 68, 0x0000 }, - { 69, 0x0000 }, - { 70, 0x0000 }, - { 71, 0x0000 }, - { 72, 0x0000 }, - { 73, 0x0000 }, - { 74, 0x0000 }, - { 75, 0x0000 }, - { 76, 0xBE3E }, - { 77, 0x0000 }, - { 78, 0xBE3E }, - { 79, 0x0000 }, - { 80, 0x0000 }, - { 81, 0x0000 }, - { 82, 0x0000 }, - { 83, 0x0000 }, - { 84, 0x803A }, - { 85, 0x0000 }, - { 86, 0x0000 }, - { 87, 0x0000 }, - { 88, 0x0000 }, - { 89, 0x0000 }, - { 90, 0x0009 }, - { 91, 0x0000 }, - { 92, 0x0000 }, - { 93, 0x0000 }, - { 94, 0x3000 }, - { 95, 0x0000 }, - { 96, 0x3075 }, - { 97, 0x0000 }, - { 98, 0x1010 }, - { 99, 0x0000 }, - { 100, 0x3110 }, - { 101, 0x0000 }, - { 102, 0x0000 }, - { 103, 0x0000 }, - { 104, 0x0553 }, - { 105, 0x0000 }, - { 106, 0x0000 }, - { 107, 0x0000 }, - { 108, 0x0000 }, - { 109, 0x0000 }, - { 110, 0x0000 }, - { 111, 0x0000 }, + { 0, 0x59B4 }, /* R0 - Reset */ + { 2, 0x8080 }, /* R2 - Speaker Output Volume */ + { 4, 0x8080 }, /* R4 - Headphone Output Volume */ + { 6, 0x8080 }, /* R6 - AUXOUT Volume */ + { 8, 0xC800 }, /* R8 - Phone Input */ + { 10, 0xE808 }, /* R10 - LINE_IN Volume */ + { 12, 0x1010 }, /* R12 - STEREO DAC Input Volume */ + { 14, 0x0808 }, /* R14 - MIC Input Volume */ + { 16, 0xEE0F }, /* R16 - Stereo DAC and MIC Routing Control */ + { 18, 0xCBCB }, /* R18 - ADC Record Gain */ + { 20, 0x7F7F }, /* R20 - ADC Record Mixer Control */ + { 24, 0xE010 }, /* R24 - Voice DAC Volume */ + { 28, 0x8008 }, /* R28 - Output Mixer Control */ + { 34, 0x0000 }, /* R34 - Microphone Control */ + { 36, 0x00C0 }, /* R36 - Codec Digital MIC/Digital Boost + Control */ + { 38, 0xEF00 }, /* R38 - Power Down Control/Status */ + { 46, 0x0000 }, /* R46 - Stereo DAC/Voice DAC/Stereo ADC + Function Select */ + { 52, 0x8000 }, /* R52 - Main Serial Data Port Control + (Stereo I2S) */ + { 54, 0x0000 }, /* R54 - Extend Serial Data Port Control + (VoDAC_I2S/PCM) */ + { 58, 0x0000 }, /* R58 - Power Management Addition 1 */ + { 60, 0x0000 }, /* R60 - Power Management Addition 2 */ + { 62, 0x8000 }, /* R62 - Power Management Addition 3 */ + { 64, 0x0C0A }, /* R64 - General Purpose Control Register 1 */ + { 66, 0x0000 }, /* R66 - General Purpose Control Register 2 */ + { 68, 0x0000 }, /* R68 - PLL1 Control */ + { 70, 0x0000 }, /* R70 - PLL2 Control */ + { 76, 0xBE3E }, /* R76 - GPIO Pin Configuration */ + { 78, 0xBE3E }, /* R78 - GPIO Pin Polarity */ + { 80, 0x0000 }, /* R80 - GPIO Pin Sticky */ + { 82, 0x0000 }, /* R82 - GPIO Pin Wake Up */ + { 84, 0x803A }, /* R84 - GPIO Pin Status */ + { 86, 0x0000 }, /* R86 - Pin Sharing */ + { 88, 0x0000 }, /* R88 - Over-Temp/Current Status */ + { 90, 0x0009 }, /* R90 - Soft Volume Control Setting */ + { 92, 0x0000 }, /* R92 - GPIO_Output Pin Control */ + { 94, 0x3000 }, /* R94 - MISC Control */ + { 96, 0x3075 }, /* R96 - Stereo DAC Clock Control_1 */ + { 98, 0x1010 }, /* R98 - Stereo DAC Clock Control_2 */ + { 100, 0x3110 }, /* R100 - VoDAC_PCM Clock Control_1 */ + { 104, 0x0553 }, /* R104 - Pseudo Stereo and Spatial Effect + Block Control */ + { 106, 0x0000 }, /* R106 - Private Register Address */ + { 108, 0x0000 }, /* R108 - Private Register Data */ + { 110, 0x0000 }, /* R110 - EQ Control and Status/ADC + HPF Control */ }; /* codec private data */ -- 1.7.3.4

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