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

[alsa-devel] Destruction of the sound after prolonged output
by Максим 18 Mar '11

18 Mar '11

Hi, My name is Max. I've got some problems using ALSA, a certain type of destruction of sound appears after a prolonged sound output. The destruction is like "bit crush", stutterings, interruptions, etc. Once started, it no longer stops. It starts almost always but only after a prolonged sound output, nearby 24 hours. To reproduce it shurely I use my test wav file and standard speaker-test utility with following options: speaker-test -t wav -w test.wav -r 16000 I've attached test.wav to this mail. To reproduce this I use a fresh minimal installation of Ubuntu 10.10, command line only, with standard Ubuntu 2.6.35 generic kernel. Minimal installation is made by installing Ubuntu from an "alternate" version of the installation CD. It provide option "command line minimal install" that allow install only base system with command line. So there isn't any desktop applications like Gnome, Pulseaudio, etc. I've used standard Ubuntu package of ALSA with version 1.0.23, and also with compiled 1.0.24. I've tried to reproduce this behavior on Centos and Fedora (both in single mode, command line only) with Intel HDA chipset, and on Ubuntu with a Realtek chipset. The results was the same, sound destruction after a day of sound output. Above is the output of speaker-test with parameters: speaker-test -d -c 2 -t wav Before this command my sound test was used during about 6 hours to reproduce the sound destruction: speaker-test -t wav -w test.wav -r 16000 Plug PCM: Linear conversion PCM (S32_LE) Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Soft volume PCM Control: PCM Playback Volume min_dB: -51 max_dB: 0 resolution: 256 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : ENABLE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1 stop_threshold : 1073741824 silence_threshold: 0 silence_size : 1073741824 boundary : 1073741824 appl_ptr : 0 hw_ptr : 926558209

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Re: [alsa-devel] Question about your DSP topic branch
by Patrick Lai 18 Mar '11

18 Mar '11

Hi Liam, I have two more questions about your DSP topic branch 7. I see in sdp4430.c, SDP4430 MODEM front-end dailink no_host_mode is set to SND_SOC_DAI_LINK_NO_HOST. What is the purpose of no_host_mode? Is it for use case that two physical devices can exchange audio data without host-processor intervention? If so, if user-space application tries to write to PCM buffer, will framework reject the buffer? 8. I see there is dmic codec(dmic.c) under sound/soc/codec which is pretty much just a dummy codec driver. I supposed the configuration of DMIC is done in other driver. Would it be better if we could have something like fixed voltage regulator? So, there is no need to duplicate the effort. Look forward to seeing your reply soon Thanks On 1/6/2011 3:39 PM, Patrick Lai wrote: > Hi Liam, > > I sync to your kernel DSP topic branch two days back in attempt to > understand the up-coming ASOC DSP design. I have few questions to get > clarification from you. > > 1. In the sdp4430.c, there are both FE dai link and BE dai link have > omap-aess-audio as platform driver. How is omap-aess-audio platform > driver used in both front-end and backend? > > 2. Front-end dai with stream name "Multimedia" has platform driver as > omap-pcm-audio which is the DMA based. This front-end dai is mapped to > backend (i.e PDM-DL1) with platform driver as omap-aess-audio. > Omap-aess-audio looks to me is DSP platform driver. If a stream is DMA > based, it seems strange to have DSP-based backend. > > 3. To best of my knowledge, I see in omap-abe.c which is the front end > implementation. Playback_trigger() seems to go ahead enable all > back-ends linked to the given front-end. However, front end may not be > routed to all back-ends. Where in the code to make sure BE is only > activated when a stream is being routed to that particular back-end? > > 4. omap-abe.c manages activation of BE DAIS and omap-abe-dsp.c manages > routing of front-end to back-end DAPM widgets and routing map. Am I > correct? This leads to previous question. How are two drivers working > together to make sure BEs are only activated if Front-end has been > routed to them? > > 5. Is there mechanism for front-end to switch between DMA and DSP > platform driver? It looks to me that mapping of frond-end and platform > driver is predetermined based on use case. For example, HDMI goes > through DMA as legacy dailink. > > 6. struct abe_data has array member called dapm. It looks to me that > this array simply tracks dapm components status but I don't see it's > really being used in meaningful way in omap-abe-adsp.c. > > Thanks > -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum. -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH] alsa-lib: fixed coverity reported issues under "REVERSE_INULL" checker.
by sudarshan.bisht＠nokia.com 17 Mar '11

17 Mar '11

From: Sudarshan <sudarshan.bisht(a)nokia.com> --- src/control/setup.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-) diff --git a/src/control/setup.c b/src/control/setup.c index eecda45..3606e5c 100644 --- a/src/control/setup.c +++ b/src/control/setup.c @@ -398,7 +398,7 @@ static int snd_config_get_ctl_elem_value(snd_config_t *conf, static int add_elem(snd_sctl_t *h, snd_config_t *_conf, snd_config_t *private_data) { - snd_config_t *conf; + snd_config_t *conf = NULL ; snd_config_iterator_t i, next; char *tmp; int iface = SND_CTL_ELEM_IFACE_MIXER; @@ -412,6 +412,9 @@ static int add_elem(snd_sctl_t *h, snd_config_t *_conf, snd_config_t *private_da snd_config_t *value = NULL, *mask = NULL; snd_sctl_elem_t *elem = NULL; int err; + /* The last parameter of snd_config_expand() is an output parameter, and + * it is guaranteed that this parameter is set when the function succeeds. + */ err = snd_config_expand(_conf, _conf, NULL, private_data, &conf); if (err < 0) return err; -- 1.7.0.4

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[alsa-devel] [PATCH 0/1] Fixed issues/defects reported by Coverity tool
by sudarshan.bisht＠nokia.com 17 Mar '11

17 Mar '11

From: Sudarshan <sudarshan.bisht(a)nokia.com> Coverity Static Analysis helps developers find hard-to-spot, yet potentially crash-causing defects early in the development phase, reducing the cost,time, and risk of software errors. This patch has a fix for a situation where variable can become NULL after certain function call, but later not been checked before passing to the next function call. Sudarshan (1): alsa-lib: fixed coverity reported issues under "REVERSE_INULL" checker. src/control/setup.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-)

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[alsa-devel] [PATCH] alsa-lib: fixed coverity reported issues under "USE_AFTER_FREE" checker.
by sudarshan.bisht＠nokia.com 17 Mar '11

17 Mar '11

From: Sudarshan <sudarshan.bisht(a)nokia.com> --- modules/mixer/simple/python.c | 1 + src/control/control_hw.c | 1 + src/mixer/mixer.c | 1 - src/mixer/simple_abst.c | 2 ++ 4 files changed, 4 insertions(+), 1 deletions(-) diff --git a/modules/mixer/simple/python.c b/modules/mixer/simple/python.c index c822c52..784d9a0 100644 --- a/modules/mixer/simple/python.c +++ b/modules/mixer/simple/python.c @@ -663,6 +663,7 @@ pymixer_attach_hctl(struct pymixer *pymixer, PyObject *args) return NULL; err = snd_mixer_attach_hctl(pymixer->mixer, hctl); if (err < 0) { + snd_hctl_close(hctl); PyErr_Format(PyExc_RuntimeError, "Cannot attach hctl: %s", snd_strerror(err)); return NULL; } diff --git a/src/control/control_hw.c b/src/control/control_hw.c index cf258b4..16c4987 100644 --- a/src/control/control_hw.c +++ b/src/control/control_hw.c @@ -414,6 +414,7 @@ int snd_ctl_hw_open(snd_ctl_t **handle, const char *name, int card, int mode) if (err < 0) { close(fd); free(hw); + return err; } ctl->ops = &snd_ctl_hw_ops; ctl->private_data = hw; diff --git a/src/mixer/mixer.c b/src/mixer/mixer.c index 85d843f..5e5789a 100644 --- a/src/mixer/mixer.c +++ b/src/mixer/mixer.c @@ -228,7 +228,6 @@ int snd_mixer_attach_hctl(snd_mixer_t *mixer, snd_hctl_t *hctl) return -ENOMEM; err = snd_hctl_nonblock(hctl, 1); if (err < 0) { - snd_hctl_close(hctl); free(slave); return err; } diff --git a/src/mixer/simple_abst.c b/src/mixer/simple_abst.c index 9e9aaf5..127d8d2 100644 --- a/src/mixer/simple_abst.c +++ b/src/mixer/simple_abst.c @@ -336,6 +336,8 @@ int snd_mixer_simple_basic_register(snd_mixer_t *mixer, err = find_module(class, top); if (err >= 0) err = snd_mixer_attach_hctl(mixer, priv->hctl); + if(err < 0) + goto __error; if (err >= 0) { priv->attach_flag = 1; err = snd_mixer_class_register(class, mixer); -- 1.7.0.4

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[alsa-devel] [PATCH] *** Fixed issues/defects reported by Coverity tool ***
by sudarshan.bisht＠nokia.com 17 Mar '11

17 Mar '11

From: Sudarshan <sudarshan.bisht(a)nokia.com> Coverity Static Analysis helps developers find hard-to-spot, yet potentially crash-causing defects early in the development phase, reducing the cost,time, and risk of software errors. This patch has a fix for conditions where 1) variable is used evern after getting freed thus causing junk assignment and 2) double free of memory. Sudarshan (1): alsa-lib: fixed coverity reported issues under "USE_AFTER_FREE" checker. src/control/control_hw.c | 1 + src/mixer/mixer.c | 2 +- 2 files changed, 2 insertions(+), 1 deletions(-)

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[alsa-devel] some questions about sound system
by loody 17 Mar '11

17 Mar '11

Dear all: I just install Ubuntu9.10 on my laptop and I try to update my sound system and recompile the utility. there are some questions I met: 1. when configure the utility it says checking panel.h presence... no checking for panel.h... no configure: error: required curses helper header not found and config.log is "conftest.c:10:42: error: CoreFoundation/CFPreferences.h: No such file or directory conftest.c: In function 'main': conftest.c:14: error: 'NULL' undeclared (first use in this function) conftest.c:14: error: (Each undeclared identifier is repor" the above error seems I miss install some package before. I google the name but I have no improvement. "conftest.c:20:28: error: ac_nonexistent.h: No such file or directory configure:7397: $? = 1" the above error seems the header located where the conftest.c doesn't expect. how could solve them all? 2. there are some configure file in share when I compile advance linux sound architecture. like alsa.conf. have anyone know where I can find the naming rule? appreciate your help, miloody

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[alsa-devel] ALC663 on ASUS G72GX Laptop
by nibs 16 Mar '11

16 Mar '11

I was advised to inquire about this here. Should the driver create two digital devices? _From the ALC663 datasheet: "In addition to the standard (primary) S/PDIF output function, the ALC663 features another independent (secondary) S/PDIF-OUT output and converters that transport digital audio output to a High Definition Media Interface (HDMI) transmitter (becoming more common in high-end PCs." More information here. https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5310

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[alsa-devel] UCM list 'modifiers' Seg-Fault when list is empty
by Alejandro Gonzalez 16 Mar '11

16 Mar '11

Hi, I've been testing the Alsa UCM and find that when listing 'modifiers' there is a segmentation fault if the list is empty. $ alsaucm -c Panda set _verb HiFi list _modifiers Im setting defaults list is empty Segmentation fault (core dumped) Using 'gdb' I see that the issue is trying to free a not NULL pointer. Program received signal SIGSEGV, Segmentation fault. 0x402a096c in free () from /lib/libc.so.6 (gdb) bt #0 0x402a096c in free () from /lib/libc.so.6 #1 0x40221224 in snd_use_case_free_list () from /usr/lib/libasound.so.2 #2 0x00008d8a in do_one (context=0x13008) at usecase.c:256 #3 do_commands (context=0x13008) at usecase.c:329 #4 0x000091aa in main (argc=<value optimized out>, argv=0xbe9986c4) at usecase.c:440 (gdb) quit But this can be easily resolved with the correct initialization of the '**list' pointer on the 'do_one' function in 'alsaucm/usecase.c'. You can find a patch attached to the mail. Regards, Alejandro

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[alsa-devel] [PATCH] sound/core: merge list_del()/list_add_tail() to list_move_tail()
by Nicolas Kaiser 16 Mar '11

16 Mar '11

Merge list_del() + list_add_tail() to list_move_tail(). Signed-off-by: Nicolas Kaiser <nikai(a)nikai.net> --- sound/core/timer.c | 8 +++----- 1 files changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index ed01632..7c1cbf0 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -186,9 +186,8 @@ static void snd_timer_check_slave(struct snd_timer_instance *slave) list_for_each_entry(master, &timer->open_list_head, open_list) { if (slave->slave_class == master->slave_class && slave->slave_id == master->slave_id) { - list_del(&slave->open_list); - list_add_tail(&slave->open_list, - &master->slave_list_head); + list_move_tail(&slave->open_list, + &master->slave_list_head); spin_lock_irq(&slave_active_lock); slave->master = master; slave->timer = master->timer; @@ -414,8 +413,7 @@ static void snd_timer_notify1(struct snd_timer_instance *ti, int event) static int snd_timer_start1(struct snd_timer *timer, struct snd_timer_instance *timeri, unsigned long sticks) { - list_del(&timeri->active_list); - list_add_tail(&timeri->active_list, &timer->active_list_head); + list_move_tail(&timeri->active_list, &timer->active_list_head); if (timer->running) { if (timer->hw.flags & SNDRV_TIMER_HW_SLAVE) goto __start_now; -- 1.7.3.4

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