[alsa-devel] "Resource temporarily unavailable" while reading although poll returns POLLIN event
Hi,
I wrote a small application that opens both streams (i.e. capture and playback) of a PCM in non blocking (i.e. SND_PCM_NONBLOCK) mode.
During device configuration I also set up the minimum number of frames to consider a PCM ready:
--------------------------------------------------------------------------------- // when to consider the device to be ready for the next data transfer operation if ((ret=snd_pcm_sw_params_set_avail_min(handle, swparams, pcm_period_size))<0) { fprintf(stderr, "could not set avail_min: %s\n", snd_strerror(ret)); return NULL; } ---------------------------------------------------------------------------------
As you can see it is set to the number of frames one period contains.
In other words, each time there are at least that many frames available for reading on the capture device, we should get a POLLIN event when using poll(). On the other side we should get a POLLOUT event if that many frames can be written to the playback device.
Then, later in my application, I use poll() on the fds of the PCM devices. As, according to the ALSA reference, poll events can be "mangeled", I use snd_pcm_poll_descriptors_revents() to "demangle" the events:
--------------------------------------------------------------------------------- if ((ret=snd_pcm_poll_descriptors_revents(device_handle, &poll_fds[i], 1, &revents))<0) { fprintf(stderr, "could not snd_pcm_poll_descriptors_revents: %s\n", snd_strerror(ret)); exit(EXIT_FAILURE); } ---------------------------------------------------------------------------------
After that, each time there was a POLLIN event, I am reading exactly one period from the capture PCM. On the other side, if there was a POLLOUT event, I am writing exactly one period to the playback PCM.
While this works for a small number of periods (which is not always the same), I always end up with a "Resource temporarily unavailable" error when trying to read one period from the capture PCM:
--------------------------------------------------------------------------------- demangled poll: POLLIN on capture device could not read from capture device: Resource temporarily unavailable ---------------------------------------------------------------------------------
Since a POLLIN event only occurs after at least a full period is available for reading (as set up by snd_pcm_sw_params_set_avail_min() above) and I only read after a POLLIN event occured on the capture device fd, I really do not understand why I get the above error.
So why does the above happen ? What am I doing wrong ?
If it helps, I pasted the source code here so that you can view it nicely: http://pastebin.com/fCicqctq
cheers, stefan
Stefan Schoenleitner wrote:
Since a POLLIN event only occurs after at least a full period is available for reading (as set up by snd_pcm_sw_params_set_avail_min() above) and I only read after a POLLIN event occured on the capture device fd, I really do not understand why I get the above error.
By using snd_pcm_avail_update() I found out that polling on the PCMs *does not work at all*.
Although I verified that avail_min is 160 frames, polling on the capture/playback PCMs returns a POLLIN/POLLOUT event even if the number of frames for reading/writing *is less than avail_main*.
I also tried snd_pcm_wait() which should also wait until there are at least avail_min frames available for reading/writing. The result is the same: snd_pcm_avail_update() shows that it returns even if there are far less than avail_min frames available for processing.
I suspect that this is a bug in ALSA ?
cheers, stefan
On Wed, 21 Apr 2010, Stefan Schoenleitner wrote:
Stefan Schoenleitner wrote:
Since a POLLIN event only occurs after at least a full period is available for reading (as set up by snd_pcm_sw_params_set_avail_min() above) and I only read after a POLLIN event occured on the capture device fd, I really do not understand why I get the above error.
By using snd_pcm_avail_update() I found out that polling on the PCMs *does not work at all*.
Although I verified that avail_min is 160 frames, polling on the capture/playback PCMs returns a POLLIN/POLLOUT event even if the number of frames for reading/writing *is less than avail_main*.
I also tried snd_pcm_wait() which should also wait until there are at least avail_min frames available for reading/writing. The result is the same: snd_pcm_avail_update() shows that it returns even if there are far less than avail_min frames available for processing.
I suspect that this is a bug in ALSA ?
It might be. Could you post snd_pcm_dump() when you read less frames?
Jaroslav
----- Jaroslav Kysela perex@perex.cz Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.
Jaroslav Kysela wrote:
On Wed, 21 Apr 2010, Stefan Schoenleitner wrote:
I suspect that this is a bug in ALSA ?
It might be. Could you post snd_pcm_dump() when you read less frames?
Sure, here's the output:
----------------------------------------------------------------- snd_pcm_avail_update(capture): 11 demangled poll: POLLIN on capture device could not read from capture device: Resource temporarily unavailable ALSA <-> PulseAudio PCM I/O Plugin Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 640 period_size : 160 period_time : 20000 tstamp_mode : NONE period_step : 1 avail_min : 160 period_event : 0 start_threshold : 160 stop_threshold : 320 silence_threshold: 0 silence_size : 0 boundary : 5764607523034234880 -----------------------------------------------------------------
As you can see I'm using pulseaudio. The version I'm using is 1:0.9.14-0ubuntu20 (on ubuntu jaunty).
For testing purposes I increased the period size to 320 so that the program works on my hardware sound card "hw". It turned out that the program is working fine there and the poll() behavior is as it should be.
cheers, stefan
2010/4/22 Stefan Schoenleitner dev.c0debabe@gmail.com
Jaroslav Kysela wrote:
On Wed, 21 Apr 2010, Stefan Schoenleitner wrote:
I suspect that this is a bug in ALSA ?
It might be. Could you post snd_pcm_dump() when you read less frames?
Sure, here's the output:
snd_pcm_avail_update(capture): 11 demangled poll: POLLIN on capture device could not read from capture device: Resource temporarily unavailable ALSA <-> PulseAudio PCM I/O Plugin Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 640 period_size : 160 period_time : 20000 tstamp_mode : NONE period_step : 1 avail_min : 160 period_event : 0 start_threshold : 160 stop_threshold : 320 silence_threshold: 0 silence_size : 0 boundary : 5764607523034234880
As you can see I'm using pulseaudio. The version I'm using is 1:0.9.14-0ubuntu20 (on ubuntu jaunty).
For testing purposes I increased the period size to 320 so that the program works on my hardware sound card "hw". It turned out that the program is working fine there and the poll() behavior is as it should be.
cheers, stefan
you will need to provide the full pulseaudio log if you are using pulseaudio
pulseaudio --kill; pulseaudio -vvvv
2010/4/22 Stefan Schoenleitner dev.c0debabe@gmail.com
For testing purposes I increased the period size to 320 so that the program works on my hardware sound card "hw". It turned out that the program is working fine there and the poll() behavior is as it should be.
cheers, stefan
Is it possible to post the output of your program when using your sound card "hw" since your program failed with XRUN (broken pipe) on my two sound cards ?
Raymond Yau wrote:
Is it possible to post the output of your program when using your sound card "hw" since your program failed with XRUN (broken pipe) on my two sound cards ?
Sure, no problem.
However, I had to change 2 settings to get it working on my soundcard:
* change period size from 160 to 320, as my soundcard does not support a period size of 160 frames
* change buffer size from 2 periods (1280 bytes) to 1 period (640 bytes), as my soundcard only supports a buffersize being equal to one period
After that the program works fine and runs forever (see below).
I'm looking forward to test it on my embedded target as well (with the original settings). If required I can post the output of that as well.
cheers, stefan
------------------------------------------------------------------------------- $ ./duplex hw could not sched_setscheduler: Operation not permitted PCM format is signed, linear, LE with 16 bits PCM rate 8000 - 48000 Hz PCM period size: 192 - 16384 PCM buffer size: 640 - 640 calculated buffer size: 640 avail min: 320 avail min after setup: 320 start threshold: 320 frames stop threshold: 640 frames PCM format is signed, linear, LE with 16 bits PCM rate 4000 - 96000 Hz PCM period size: 32 - 32768 PCM buffer size: 320 - 32640 calculated buffer size: 640 avail min: 320 avail min after setup: 320 start threshold: 320 frames stop threshold: 640 frames capture fds: 1, playback fds: 1 capture poll fd: 4, playback poll fd: 5 capture struct: fd: 4, events: POLLIN POLLERR , revents: 0 playback struct: fd: 5, events: POLLOUT POLLERR , revents: 0 capture avail min: 320 playback avail min: 320 snd_pcm_avail_update(capture): 0 demangled poll: on capture device snd_pcm_avail_update(playback): 640 demangled poll: POLLOUT on playback device wrote 320 frames snd_pcm_avail_update(capture): 0 demangled poll: on capture device snd_pcm_avail_update(playback): 320 demangled poll: POLLOUT on playback device wrote 320 frames snd_pcm_avail_update(capture): 320 demangled poll: POLLIN on capture device read 320 frames hexdump(): 640 bytes 0000 fe ff fe ff fe ff fe ff ff ff fe ff fe ff ff ff 0010 fe ff ff ff ff ff ff ff fe ff fe ff fe ff ff ff 0020 ff ff fe ff fe ff ff ff ff ff ff ff ff ff ff ff 0030 ff ff fe ff fe ff ff ff ff ff fe ff ff ff fe ff 0040 fe ff fe ff fe ff ff ff ff ff fe ff fe ff ff ff 0050 ff ff ff ff fe ff fe ff fe ff ff ff ff ff fe ff 0060 fe ff ff ff ff ff fe ff fe ff ff ff ff ff fe ff 0070 ff ff fe ff fe ff ff ff fe ff fe ff ff ff fe ff 0080 fe ff ff ff ff ff ff ff fe ff fe ff fe ff ff ff 0090 ff ff fe ff ff ff ff ff fe ff fe ff ff ff fe ff 00a0 ff ff ff ff fe ff fe ff ff ff fe ff fe ff ff ff 00b0 ff ff fe ff fe ff ff ff ff ff ff ff ff ff ff ff 00c0 ff ff ff ff ff ff ff ff ff ff ff ff ff ff fe ff 00d0 ff ff ff ff fe ff fe ff fe ff ff ff ff ff fe ff 00e0 fe ff fe ff ff ff ff ff fe ff fe ff ff ff fe ff 00f0 ff ff ff ff ff ff ff ff ff ff fe ff ff ff ff ff 0100 fe ff fe ff fe ff fe ff fe ff ff ff ff ff fe ff 0110 ff ff ff ff fe ff fe ff ff ff ff ff fe ff fe ff 0120 ff ff ff ff fe ff fe ff ff ff ff ff fe ff fe ff 0130 fe ff ff ff fe ff ff ff fe ff fe ff fe ff fe ff 0140 ff ff fe ff fe ff ff ff ff ff fe ff fe ff ff ff 0150 fe ff fe ff fe ff fe ff ff ff fe ff ff ff ff ff 0160 fe ff fe ff ff ff fe ff fe ff ff ff ff ff fe ff 0170 ff ff ff ff ff ff ff ff fe ff ff ff ff ff fe ff 0180 fe ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0190 ff ff fe ff fe ff ff ff ff ff ff ff ff ff ff ff 01a0 ff ff ff ff fe ff fe ff fe ff ff ff ff ff fe ff 01b0 fe ff fe ff ff ff fe ff fe ff ff ff ff ff ff ff 01c0 ff ff ff ff ff ff ff ff ff ff ff ff fe ff fe ff 01d0 fe ff fe ff fe ff ff ff ff ff fe ff fe ff fe ff 01e0 fe ff ff ff ff ff ff ff fe ff fe ff ff ff ff ff 01f0 fe ff fe ff ff ff ff ff ff ff ff ff ff ff fe ff 0200 fe ff ff ff ff ff ff ff ff ff ff ff ff ff fe ff 0210 ff ff ff ff fe ff fe ff ff ff ff ff ff ff ff ff 0220 ff ff fe ff fe ff ff ff ff ff ff ff fe ff ff ff 0230 ff ff ff ff fe ff ff ff ff ff ff ff ff ff ff ff 0240 ff ff ff ff ff ff ff ff ff ff fe ff fe ff ff ff 0250 ff ff fe ff ff ff ff ff ff ff ff ff ff ff fe ff 0260 fe ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0270 fe ff ff ff ff ff fe ff fe ff ff ff ff ff ff ff
snd_pcm_avail_update(playback): 320 demangled poll: on playback device snd_pcm_avail_update(capture): 0 demangled poll: on capture device snd_pcm_avail_update(playback): 320 demangled poll: POLLOUT on playback device wrote 320 frames snd_pcm_avail_update(capture): 320 demangled poll: POLLIN on capture device read 320 frames hexdump(): 640 bytes 0000 ff ff ff ff fe ff ff ff ff ff ff ff ff ff ff ff 0010 fe ff fe ff ff ff ff ff fe ff ff ff ff ff ff ff 0020 ff ff ff ff fe ff ff ff ff ff fe ff ff ff ff ff 0030 fe ff fe ff ff ff fe ff fe ff ff ff ff ff fe ff 0040 fe ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0050 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0060 ff ff fe ff fe ff ff ff ff ff ff ff ff ff ff ff 0070 ff ff fe ff fe ff fe ff ff ff ff ff ff ff fe ff 0080 fe ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0090 ff ff ff ff ff ff ff ff fe ff fe ff ff ff fe ff 00a0 ff ff fe ff fe ff ff ff ff ff ff ff ff ff fe ff 00b0 fe ff fe ff fe ff fe ff fe ff fe ff ff ff ff ff 00c0 ff ff ff ff ff ff fe ff ff ff ff ff ff ff fe ff 00d0 fe ff ff ff fe ff fe ff ff ff ff ff fe ff fe ff 00e0 fe ff ff ff fe ff fe ff ff ff ff ff ff ff ff ff 00f0 ff ff ff ff fe ff ff ff ff ff ff ff ff ff ff ff 0100 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0110 ff ff ff ff fe ff ff ff ff ff ff ff ff ff fe ff 0120 ff ff ff ff fe ff ff ff ff ff fe ff fe ff ff ff 0130 ff ff ff ff ff ff ff ff ff ff fe ff fe ff ff ff 0140 fe ff ff ff ff ff fe ff ff ff ff ff fe ff fe ff 0150 fe ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0160 ff ff ff ff ff ff fe ff fe ff fe ff ff ff ff ff 0170 ff ff ff ff ff ff ff ff ff ff ff ff fe ff fe ff 0180 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0190 ff ff ff ff ff ff ff ff ff ff fe ff fe ff ff ff 01a0 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 01b0 ff ff ff ff ff ff ff ff fe ff fe ff ff ff ff ff 01c0 ff ff ff ff ff ff ff ff ff ff ff ff ff ff fe ff 01d0 ff ff ff ff ff ff ff ff fe ff ff ff ff ff ff ff 01e0 ff ff fe ff fe ff ff ff ff ff ff ff ff ff ff ff 01f0 ff ff ff ff ff ff fe ff fe ff fe ff fe ff fe ff 0200 fe ff fe ff fe ff ff ff fe ff ff ff ff ff fe ff 0210 fe ff ff ff ff ff ff ff ff ff ff ff fe ff fe ff 0220 ff ff ff ff ff ff ff ff ff ff fe ff ff ff ff ff 0230 ff ff ff ff ff ff ff ff ff ff ff ff fe ff fe ff 0240 fe ff ff ff ff ff ff ff ff ff ff ff ff ff fe ff 0250 fe ff fe ff ff ff ff ff ff ff ff ff ff ff ff ff 0260 fe ff ff ff ff ff ff ff fe ff ff ff ff ff ff ff 0270 ff ff ff ff fe ff ff ff ff ff ff ff ff ff ff ff
snd_pcm_avail_update(playback): 320 demangled poll: on playback device snd_pcm_avail_update(capture): 0 demangled poll: on capture device snd_pcm_avail_update(playback): 320 demangled poll: POLLOUT on playback device wrote 320 frames snd_pcm_avail_update(capture): 320 demangled poll: POLLIN on capture device read 320 frames hexdump(): 640 bytes 0000 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0010 ff ff ff ff ff ff ff ff ff ff fe ff fe ff ff ff 0020 ff ff fe ff fe ff ff ff fe ff fe ff ff ff fe ff 0030 fe ff ff ff ff ff ff ff ff ff fe ff ff ff ff ff 0040 fe ff fe ff fe ff fe ff fe ff fe ff fe ff fe ff 0050 fe ff fe ff ff ff fe ff fe ff fe ff ff ff ff ff 0060 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0070 ff ff ff ff fe ff fe ff ff ff ff ff fe ff fe ff 0080 ff ff ff ff ff ff ff ff ff ff ff ff ff ff fe ff 0090 fe ff ff ff fe ff fe ff ff ff fe ff ff ff ff ff 00a0 ff ff fe ff ff ff ff ff fe ff ff ff ff ff ff ff 00b0 ff ff ff ff fe ff fe ff fe ff ff ff fe ff fe ff 00c0 fe ff fe ff fe ff fe ff fe ff fe ff ff ff fe ff 00d0 fe ff fe ff ff ff fe ff fe ff fe ff fe ff ff ff 00e0 ff ff ff ff ff ff fe ff fe ff fe ff fe ff fe ff 00f0 ff ff fe ff fe ff ff ff ff ff ff ff ff ff ff ff 0100 fe ff fe ff fe ff fe ff fe ff fe ff fe ff fe ff 0110 fe ff ff ff fe ff fe ff ff ff ff ff fe ff fe ff 0120 fe ff fe ff fe ff ff ff ff ff ff ff fe ff fe ff 0130 fe ff fe ff ff ff ff ff ff ff fe ff fe ff fe ff 0140 ff ff ff ff fe ff fe ff fe ff fe ff fe ff ff ff 0150 fe ff fe ff ff ff ff ff ff ff fe ff fe ff fe ff 0160 fe ff fe ff fe ff ff ff ff ff ff ff ff ff ff ff 0170 ff ff ff ff ff ff ff ff fe ff ff ff ff ff ff ff 0180 fe ff fe ff ff ff ff ff fe ff ff ff fe ff fe ff 0190 fe ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 01a0 ff ff ff ff fe ff ff ff ff ff fe ff fe ff ff ff 01b0 ff ff ff ff ff ff ff ff ff ff ff ff fe ff fe ff 01c0 ff ff ff ff fe ff fe ff ff ff ff ff ff ff ff ff 01d0 ff ff fe ff fe ff ff ff ff ff ff ff ff ff ff ff 01e0 ff ff fe ff ff ff ff ff ff ff fe ff ff ff fe ff 01f0 fe ff ff ff ff ff fe ff fe ff ff ff fe ff fe ff 0200 ff ff ff ff ff ff ff ff ff ff fe ff fe ff fe ff 0210 fe ff fe ff fe ff ff ff ff ff fe ff fe ff ff ff 0220 ff ff ff ff ff ff ff ff ff ff fe ff fe ff fe ff 0230 ff ff ff ff ff ff ff ff ff ff ff ff fe ff fe ff 0240 fe ff fe ff fe ff ff ff fe ff fe ff fe ff fe ff 0250 fe ff ff ff ff ff ff ff ff ff ff ff ff ff fe ff 0260 fe ff ff ff fe ff fe ff fe ff fe ff fe ff fe ff 0270 ff ff ff ff fe ff fe ff fe ff fe ff ff ff ff ff
snd_pcm_avail_update(playback): 320 demangled poll: POLLOUT on playback device wrote 320 frames snd_pcm_avail_update(capture): 320 demangled poll: POLLIN on capture device read 320 frames hexdump(): 640 bytes 0000 ff ff fe ff ff ff ff ff ff ff ff ff fe ff fe ff 0010 ff ff ff ff ff ff ff ff fe ff fe ff fe ff fe ff 0020 fe ff fe ff fe ff fe ff ff ff ff ff ff ff ff ff 0030 ff ff ff ff ff ff ff ff ff ff ff ff ff ff fe ff 0040 ff ff ff ff fe ff fe ff ff ff fe ff ff ff fe ff 0050 fe ff ff ff ff ff fe ff ff ff ff ff ff ff ff ff 0060 fe ff fe ff fe ff ff ff ff ff ff ff fe ff ff ff 0070 ff ff ff ff fe ff fe ff fe ff ff ff ff ff fe ff 0080 fe ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0090 fe ff fe ff ff ff ff ff ff ff ff ff ff ff ff ff 00a0 fe ff ff ff fe ff ff ff fe ff fe ff ff ff fe ff 00b0 fe ff ff ff ff ff ff ff fe ff fe ff ff ff ff ff 00c0 fe ff fe ff ff ff ff ff fe ff ff ff ff ff fe ff 00d0 ff ff ff ff fe ff ff ff ff ff ff ff ff ff ff ff 00e0 ff ff ff ff ff ff ff ff fe ff ff ff ff ff fe ff 00f0 fe ff ff ff fe ff fe ff ff ff ff ff fe ff fe ff 0100 ff ff ff ff fe ff fe ff ff ff ff ff ff ff fe ff 0110 fe ff fe ff ff ff ff ff ff ff ff ff fe ff fe ff 0120 ff ff ff ff fe ff fe ff fe ff ff ff ff ff fe ff 0130 fe ff fe ff ff ff fe ff ff ff ff ff fe ff fe ff 0140 fe ff fe ff ff ff ff ff ff ff ff ff ff ff fe ff 0150 fe ff ff ff ff ff ff ff fe ff fe ff fe ff fe ff 0160 ff ff ff ff ff ff ff ff fe ff fe ff ff ff ff ff 0170 fe ff fe ff ff ff ff ff ff ff fe ff fe ff fe ff 0180 ff ff ff ff ff ff fe ff fe ff ff ff fe ff fe ff 0190 fe ff fe ff fe ff ff ff fe ff fe ff fe ff fe ff 01a0 fe ff ff ff fe ff fe ff fe ff fe ff ff ff fe ff 01b0 ff ff fe ff fe ff fe ff ff ff ff ff fe ff ff ff 01c0 ff ff fe ff fe ff fe ff ff ff fe ff fe ff fe ff 01d0 fe ff ff ff fe ff fe ff ff ff fe ff fe ff fe ff 01e0 fe ff fe ff fe ff fe ff ff ff ff ff ff ff ff ff 01f0 ff ff ff ff ff ff fe ff fe ff ff ff ff ff fe ff 0200 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [...] -------------------------------------------------------------------------------
2010/4/24 Stefan Schoenleitner dev.c0debabe@gmail.com
Raymond Yau wrote:
Is it possible to post the output of your program when using your sound
card
"hw" since your program failed with XRUN (broken pipe) on my two sound
cards
?
Sure, no problem.
However, I had to change 2 settings to get it working on my soundcard:
- change period size from 160 to 320, as my soundcard does not support a
period size of 160 frames
- change buffer size from 2 periods (1280 bytes) to 1 period (640
bytes), as my soundcard only supports a buffersize being equal to one period
After that the program works fine and runs forever (see below).
I'm looking forward to test it on my embedded target as well (with the original settings). If required I can post the output of that as well.
cheers, stefan
$ ./duplex hw could not sched_setscheduler: Operation not permitted PCM format is signed, linear, LE with 16 bits PCM rate 8000 - 48000 Hz PCM period size: 192 - 16384 PCM buffer size: 640 - 640 calculated buffer size: 640 avail min: 320 avail min after setup: 320 start threshold: 320 frames stop threshold: 640 frames PCM format is signed, linear, LE with 16 bits PCM rate 4000 - 96000 Hz PCM period size: 32 - 32768 PCM buffer size: 320 - 32640 calculated buffer size: 640 avail min: 320 avail min after setup: 320 start threshold: 320 frames stop threshold: 640 frames capture fds: 1, playback fds: 1 capture poll fd: 4, playback poll fd: 5 capture struct: fd: 4, events: POLLIN POLLERR , revents: 0 playback struct: fd: 5, events: POLLOUT POLLERR , revents: 0 capture avail min: 320 playback avail min: 320 snd_pcm_avail_update(capture): 0 demangled poll: on capture device snd_pcm_avail_update(playback): 640 demangled poll: POLLOUT on playback device wrote 320 frames snd_pcm_avail_update(capture): 0 demangled poll: on capture device snd_pcm_avail_update(playback): 320 demangled poll: POLLOUT on playback device wrote 320 frames snd_pcm_avail_update(capture): 320 demangled poll: POLLIN on capture device read 320 frames hexdump(): 640 bytes 0000 fe ff fe ff fe ff fe ff ff ff fe ff fe ff ff ff 0010 fe ff ff ff ff ff ff ff fe ff fe ff fe ff ff ff 0020 ff ff fe ff fe ff ff ff ff ff ff ff ff ff ff ff 0030 ff ff fe ff fe ff ff ff ff ff fe ff ff ff fe ff 0040 fe ff fe ff fe ff ff ff ff ff fe ff fe ff ff ff 0050 ff ff ff ff fe ff fe ff fe ff ff ff ff ff fe ff 0060 fe ff ff ff ff ff fe ff fe ff ff ff ff ff fe ff 0070 ff ff fe ff fe ff ff ff fe ff fe ff ff ff fe ff 0080 fe ff ff ff ff ff ff ff fe ff fe ff fe ff ff ff 0090 ff ff fe ff ff ff ff ff fe ff fe ff ff ff fe ff 00a0 ff ff ff ff fe ff fe ff ff ff fe ff fe ff ff ff 00b0 ff ff fe ff fe ff ff ff ff ff ff ff ff ff ff ff 00c0 ff ff ff ff ff ff ff ff ff ff ff ff ff ff fe ff 00d0 ff ff ff ff fe ff fe ff fe ff ff ff ff ff fe ff 00e0 fe ff fe ff ff ff ff ff fe ff fe ff ff ff fe ff 00f0 ff ff ff ff ff ff ff ff ff ff fe ff ff ff ff ff 0100 fe ff fe ff fe ff fe ff fe ff ff ff ff ff fe ff 0110 ff ff ff ff fe ff fe ff ff ff ff ff fe ff fe ff 0120 ff ff ff ff fe ff fe ff ff ff ff ff fe ff fe ff 0130 fe ff ff ff fe ff ff ff fe ff fe ff fe ff fe ff 0140 ff ff fe ff fe ff ff ff ff ff fe ff fe ff ff ff 0150 fe ff fe ff fe ff fe ff ff ff fe ff ff ff ff ff 0160 fe ff fe ff ff ff fe ff fe ff ff ff ff ff fe ff 0170 ff ff ff ff ff ff ff ff fe ff ff ff ff ff fe ff 0180 fe ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0190 ff ff fe ff fe ff ff ff ff ff ff ff ff ff ff ff 01a0 ff ff ff ff fe ff fe ff fe ff ff ff ff ff fe ff 01b0 fe ff fe ff ff ff fe ff fe ff ff ff ff ff ff ff 01c0 ff ff ff ff ff ff ff ff ff ff ff ff fe ff fe ff 01d0 fe ff fe ff fe ff ff ff ff ff fe ff fe ff fe ff 01e0 fe ff ff ff ff ff ff ff fe ff fe ff ff ff ff ff 01f0 fe ff fe ff ff ff ff ff ff ff ff ff ff ff fe ff 0200 fe ff ff ff ff ff ff ff ff ff ff ff ff ff fe ff 0210 ff ff ff ff fe ff fe ff ff ff ff ff ff ff ff ff 0220 ff ff fe ff fe ff ff ff ff ff ff ff fe ff ff ff 0230 ff ff ff ff fe ff ff ff ff ff ff ff ff ff ff ff 0240 ff ff ff ff ff ff ff ff ff ff fe ff fe ff ff ff 0250 ff ff fe ff ff ff ff ff ff ff ff ff ff ff fe ff 0260 fe ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0270 fe ff ff ff ff ff fe ff fe ff ff ff ff ff ff ff
snd_pcm_avail_update(playback): 320 demangled poll: on playback device snd_pcm_avail_update(capture): 0 demangled poll: on capture device snd_pcm_avail_update(playback): 320 demangled poll: POLLOUT on playback device wrote 320 frames snd_pcm_avail_update(capture): 320 demangled poll: POLLIN on capture device read 320 frames hexdump(): 640 bytes 0000 ff ff ff ff fe ff ff ff ff ff ff ff ff ff ff ff 0010 fe ff fe ff ff ff ff ff fe ff ff ff ff ff ff ff 0020 ff ff ff ff fe ff ff ff ff ff fe ff ff ff ff ff 0030 fe ff fe ff ff ff fe ff fe ff ff ff ff ff fe ff 0040 fe ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0050 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0060 ff ff fe ff fe ff ff ff ff ff ff ff ff ff ff ff 0070 ff ff fe ff fe ff fe ff ff ff ff ff ff ff fe ff 0080 fe ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0090 ff ff ff ff ff ff ff ff fe ff fe ff ff ff fe ff 00a0 ff ff fe ff fe ff ff ff ff ff ff ff ff ff fe ff 00b0 fe ff fe ff fe ff fe ff fe ff fe ff ff ff ff ff 00c0 ff ff ff ff ff ff fe ff ff ff ff ff ff ff fe ff 00d0 fe ff ff ff fe ff fe ff ff ff ff ff fe ff fe ff 00e0 fe ff ff ff fe ff fe ff ff ff ff ff ff ff ff ff 00f0 ff ff ff ff fe ff ff ff ff ff ff ff ff ff ff ff 0100 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0110 ff ff ff ff fe ff ff ff ff ff ff ff ff ff fe ff 0120 ff ff ff ff fe ff ff ff ff ff fe ff fe ff ff ff 0130 ff ff ff ff ff ff ff ff ff ff fe ff fe ff ff ff 0140 fe ff ff ff ff ff fe ff ff ff ff ff fe ff fe ff 0150 fe ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0160 ff ff ff ff ff ff fe ff fe ff fe ff ff ff ff ff 0170 ff ff ff ff ff ff ff ff ff ff ff ff fe ff fe ff 0180 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0190 ff ff ff ff ff ff ff ff ff ff fe ff fe ff ff ff 01a0 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 01b0 ff ff ff ff ff ff ff ff fe ff fe ff ff ff ff ff 01c0 ff ff ff ff ff ff ff ff ff ff ff ff ff ff fe ff 01d0 ff ff ff ff ff ff ff ff fe ff ff ff ff ff ff ff 01e0 ff ff fe ff fe ff ff ff ff ff ff ff ff ff ff ff 01f0 ff ff ff ff ff ff fe ff fe ff fe ff fe ff fe ff 0200 fe ff fe ff fe ff ff ff fe ff ff ff ff ff fe ff 0210 fe ff ff ff ff ff ff ff ff ff ff ff fe ff fe ff 0220 ff ff ff ff ff ff ff ff ff ff fe ff ff ff ff ff 0230 ff ff ff ff ff ff ff ff ff ff ff ff fe ff fe ff 0240 fe ff ff ff ff ff ff ff ff ff ff ff ff ff fe ff 0250 fe ff fe ff ff ff ff ff ff ff ff ff ff ff ff ff 0260 fe ff ff ff ff ff ff ff fe ff ff ff ff ff ff ff 0270 ff ff ff ff fe ff ff ff ff ff ff ff ff ff ff ff
How about the qualtity of the playback ? did you speak to the mic and hear the result ?
Refer to your pulseaudio log , PA use "front" device for playback/capture but your test use "hw"
D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... I: module-alsa-source.c: Successfully opened device front:0. I: module-alsa-source.c: Successfully enabled mmap() mode. I: (alsa-lib)control.c: Invalid CTL front:0 I: alsa-util.c: Unable to attach to mixer front:0: No such file or directory I: alsa-util.c: Successfully attached to mixer 'hw:0' I: alsa-util.c: Cannot find mixer control "Capture" or mixer control is no combination of switch/volume. I: alsa-util.c: Using mixer control "Mic"
...
sink_name=alsa_output.pci_ 1102_4_sound_card_0_alsa_playback_0 tsched=0' D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying front:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:front:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:front:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem I: alsa-util.c: PCM device plug:front:0 refused our hw parameters: Device or resource busy ... D: alsa-util.c: Trying hw:0 as last resort... D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ... I: module-alsa-sink.c: Successfully opened device hw:0. I: module-alsa-sink.c: Successfully enabled mmap() mode. I: alsa-util.c: Successfully attached to mixer 'hw:0' I: alsa-util.c: Cannot find mixer control "Master" or mixer control is no combination of switch/volume. W: alsa-util.c: Cannot find fallback mixer control "PCM" or mixer control is no combination of switch/volume. I: alsa-util.c: Using mixer control "PCM".
Do you mean that you suspect the problem is related to alsa-pulse plugin or PA server ?
Raymond Yau wrote:
How about the qualtity of the playback ? did you speak to the mic and hear the result ?
Yes, when speaking into the mike I can hear the output on the speakers. The sound quality is reasonable for 8kHz S16_LE and there are no sound errors (e.g. no clicks or similar).
Refer to your pulseaudio log , PA use "front" device for playback/capture
I: module-alsa-sink.c: Successfully opened device hw:0. I: alsa-util.c: Successfully attached to mixer 'hw:0' I: alsa-util.c: Using mixer control "PCM".
Here you can see that PA uses the "hw" device. (It just tries to use the "front" device in the beginning which fails. The "hw" device is opened successfully after that (see log above).)
but your test use "hw"
In my tests I first used the "default" device which uses the PA plugin. The test clearly showed that poll() returns with POLLIN even though far less than avail_min frames are available for reading. Hence I suspect that there is a *bug* somewhere in the pulseaudio code (either the plugin or the daemon).
In order to show that my code is working fine and the erroneous poll behavior does not occur when PA is not used, I also tested with the "hw" device. The result, as mentioned, it that the POLLIN event is correctly returned after at least avail_min frames are available.
In short:
* test with "default" uses PA: erroneus poll() behavior, BUG * test with "hw" does *not* use PA: everything works fine
Do you mean that you suspect the problem is related to alsa-pulse plugin or PA server ?
Yes.
cheers, Stefan
2010/4/26 Stefan Schoenleitner dev.c0debabe@gmail.com
:
In my tests I first used the "default" device which uses the PA plugin.
The test clearly showed that poll() returns with POLLIN even though far
less than avail_min frames are available for reading.
Hence I suspect that there is a *bug* somewhere in the pulseaudio code (either the plugin or the daemon).
In order to show that my code is working fine and the erroneous poll behavior does not occur when PA is not used, I also tested with the "hw" device. The result, as mentioned, it that the POLLIN event is correctly returned after at least avail_min frames are available.
In short:
- test with "default" uses PA: erroneus poll() behavior, BUG
- test with "hw" does *not* use PA: everything works fine
http://0pointer.de/blog/projects/guide-to-sound-apis.html
Most likely , the PA developer will tell you
Do *not* touch buffering/period metrics unless you have specific latency needs. Develop defensively, handling correctly the case when the backend cannot fulfill your buffering metrics requests. Be aware that the buffering metrics of the playback buffer only indirectly influence the overall latency in many cases. i.e. setting the buffer size to a fixed value might actually result in practical latencies that are much higher.
Do *not* assume that the time when a PCM stream can receive new data is strictly dependant on the sampling and buffering parameters and the resulting average throughput. Always make sure to supply new audio data to the device when it asks for it by signalling "writability" on the fd. (And similarly for capturing)
Do you mean that you suspect the problem is related to alsa-pulse plugin
or
PA server ?
Yes.
cheers, Stefan
2010/4/24 Stefan Schoenleitner dev.c0debabe@gmail.com
Raymond Yau wrote:
Is it possible to post the output of your program when using your sound
card
"hw" since your program failed with XRUN (broken pipe) on my two sound
cards
?
Sure, no problem.
However, I had to change 2 settings to get it working on my soundcard:
- change period size from 160 to 320, as my soundcard does not support a
period size of 160 frames
- change buffer size from 2 periods (1280 bytes) to 1 period (640
bytes), as my soundcard only supports a buffersize being equal to one period
After that the program works fine and runs forever (see below).
I'm looking forward to test it on my embedded target as well (with the original settings). If required I can post the output of that as well.
cheers, stefan
Are you sure that you really need pulseaudio since your request latency is quite low ?
You have to ask PA developer whether PA support such low latecny ?
PA (tsched=0 ) configure your sound card 1792 frames per period but your application request for 160 frames per period
You will need the PA expert to answer how PA server capture 1792 frames from sound card and send it to your application
: module-alsa-source.c: Using 2 fragments of size 7168 bytes, buffer time is 81.27ms
D: alsa-util.c: Its setup is: D: alsa-util.c: stream : CAPTURE D: alsa-util.c: access : MMAP_INTERLEAVED D: alsa-util.c: format : S16_LE D: alsa-util.c: subformat : STD D: alsa-util.c: channels : 2 D: alsa-util.c: rate : 44100 D: alsa-util.c: exact rate : 44100 (44100/1) D: alsa-util.c: msbits : 16 D: alsa-util.c: buffer_size : 3584 D: alsa-util.c: period_size : 1792 D: alsa-util.c: period_time : 40634 D: alsa-util.c: tstamp_mode : ENABLE D: alsa-util.c: period_step : 1 D: alsa-util.c: avail_min : 1792 D: alsa-util.c: period_event : 0 D: alsa-util.c: start_threshold : -1 D: alsa-util.c: stop_threshold : 8070450532247928832 D: alsa-util.c: silence_threshold: 0 D: alsa-util.c: silence_size : 0 D: alsa-util.c: boundary : 8070450532247928832
Raymond Yau wrote:
Are you sure that you really need pulseaudio since your request latency is quite low ?
You mean the small period_size of 160 frames or rather the buffer size of 2*period_size?
In fact at 8kHz sampling rate a period_size of 160 equals a full 20ms of sound:
duration of one frame: 1000ms / 8000Hz = 0.125ms = 125 us 160 frames * 125us = 20000us = 20ms
This is by far more than a large period size holds at a higher sampling rate (e.g. 44.1 kHz).
To the question whether pulseaudio is needed: On the embedded target PA will not be used, but since development takes place on a PC, I need a soundcard that supports the mentioned audio format contraints. And at the moment this is pulseaudio which is the reason why I really need it for application development.
However, we should try to not drift away from the actual problem which is that *poll() returns an event even if far less than avail_min frames are available*.
You have to ask PA developer whether PA support such low latecny ?
PA (tsched=0 ) configure your sound card 1792 frames per period but your application request for 160 frames per period
You will need the PA expert to answer how PA server capture 1792 frames from sound card and send it to your application
: module-alsa-source.c: Using 2 fragments of size 7168 bytes, buffer time is 81.27ms
D: alsa-util.c: Its setup is: D: alsa-util.c: stream : CAPTURE D: alsa-util.c: access : MMAP_INTERLEAVED D: alsa-util.c: format : S16_LE D: alsa-util.c: subformat : STD D: alsa-util.c: channels : 2 D: alsa-util.c: rate : 44100 D: alsa-util.c: exact rate : 44100 (44100/1) D: alsa-util.c: msbits : 16 D: alsa-util.c: buffer_size : 3584 D: alsa-util.c: period_size : 1792 D: alsa-util.c: period_time : 40634 D: alsa-util.c: tstamp_mode : ENABLE D: alsa-util.c: period_step : 1 D: alsa-util.c: avail_min : 1792 D: alsa-util.c: period_event : 0 D: alsa-util.c: start_threshold : -1 D: alsa-util.c: stop_threshold : 8070450532247928832 D: alsa-util.c: silence_threshold: 0 D: alsa-util.c: silence_size : 0 D: alsa-util.c: boundary : 8070450532247928832
This is how pulseaudio works *internally*. Hence it opens my hardware sound card "hw" with the above format. If sound it recorded/played back at a different sampling rate, PA converts it. If you play a prerecorded audio file that has been recorded at a different sampling rate, you can see that behavior.
What happens is:
[sound application (e.g.. aplay)] ---(audio format of sound file)---> [ PA plugin] --> [PA daemon] ---(audio format of PA)--> [soundcard]
cheers, stefan
2010/4/26 Stefan Schoenleitner dev.c0debabe@gmail.com
Raymond Yau wrote:
Are you sure that you really need pulseaudio since your request latency
is
quite low ?
You mean the small period_size of 160 frames or rather the buffer size of 2*period_size?
In fact at 8kHz sampling rate a period_size of 160 equals a full 20ms of sound:
duration of one frame: 1000ms / 8000Hz = 0.125ms = 125 us 160 frames * 125us = 20000us = 20ms
This is by far more than a large period size holds at a higher sampling rate (e.g. 44.1 kHz).
But the period time of your sound card is 40.634ms which is more than double of your requested 20ms
D: alsa-util.c: rate : 44100 D: alsa-util.c: exact rate : 44100 (44100/1) D: alsa-util.c: msbits : 16 D: alsa-util.c: buffer_size : 3584 D: alsa-util.c: period_size : 1792 D: alsa-util.c: period_time : 40634
To the question whether pulseaudio is needed: On the embedded target PA will not be used, but since development takes place on a PC, I need a soundcard that supports the mentioned audio format contraints. And at the moment this is pulseaudio which is the reason why I really need it for application development.
However, we should try to not drift away from the actual problem which is that *poll() returns an event even if far less than avail_min frames are available*.
You have to ask PA developer whether PA support such low latecny ?
PA (tsched=0 ) configure your sound card 1792 frames per period but your application request for 160 frames per period
You will need the PA expert to answer how PA server capture 1792 frames
from
sound card and send it to your application
: module-alsa-source.c: Using 2 fragments of size 7168 bytes, buffer time is 81.27ms
D: alsa-util.c: Its setup is: D: alsa-util.c: stream : CAPTURE D: alsa-util.c: access : MMAP_INTERLEAVED D: alsa-util.c: format : S16_LE D: alsa-util.c: subformat : STD D: alsa-util.c: channels : 2 D: alsa-util.c: rate : 44100 D: alsa-util.c: exact rate : 44100 (44100/1) D: alsa-util.c: msbits : 16 D: alsa-util.c: buffer_size : 3584 D: alsa-util.c: period_size : 1792 D: alsa-util.c: period_time : 40634 D: alsa-util.c: tstamp_mode : ENABLE D: alsa-util.c: period_step : 1 D: alsa-util.c: avail_min : 1792 D: alsa-util.c: period_event : 0 D: alsa-util.c: start_threshold : -1 D: alsa-util.c: stop_threshold : 8070450532247928832 D: alsa-util.c: silence_threshold: 0 D: alsa-util.c: silence_size : 0 D: alsa-util.c: boundary : 8070450532247928832
This is how pulseaudio works *internally*. Hence it opens my hardware sound card "hw" with the above format. If sound it recorded/played back at a different sampling rate, PA converts it. If you play a prerecorded audio file that has been recorded at a different sampling rate, you can see that behavior.
After the sound driver capture 40.634 ms of audio , PA have to convert the 44100Hz stereo to 8000Hz mono
seem just add left + right to mono without halve the sum , you may hear clipping if you are using line in instead of mic or PA clamp the output
D: resampler.c: Channel matrix: D: resampler.c: I00 I01 D: resampler.c: +------------ D: resampler.c: O00 | 1.000 1.000
What happens is:
[sound application (e.g.. aplay)] ---(audio format of sound file)---> [ PA plugin] --> [PA daemon] ---(audio format of PA)--> [soundcard]
cheers, stefan
2010/4/22 Stefan Schoenleitner dev.c0debabe@gmail.com
Stefan Schoenleitner wrote:
Since a POLLIN event only occurs after at least a full period is
available for
reading (as set up by snd_pcm_sw_params_set_avail_min() above) and I only read after a POLLIN event occured on the capture device fd, I really do not understand why I get the above error.
By using snd_pcm_avail_update() I found out that polling on the PCMs *does not work at all*.
Although I verified that avail_min is 160 frames, polling on the capture/playback PCMs returns a POLLIN/POLLOUT event even if the number of frames for reading/writing *is less than avail_main*.
I also tried snd_pcm_wait() which should also wait until there are at least avail_min frames available for reading/writing. The result is the same: snd_pcm_avail_update() shows that it returns even if there are far less than avail_min frames available for processing.
I suspect that this is a bug in ALSA ?
cheers, stefan
your program expect the driver support 2 periods per buffer but does not expicitly set the period
8000 Hz , S16_LE and mono
I verified that avail_min is 160 frames
is there any specific reason to choose 160 frames ?
Raymond Yau wrote:
your program expect the driver support 2 periods per buffer but does not expicitly set the period
8000 Hz , S16_LE and mono
I am not sure why you think this is the case.
The period size is set at line 170 with snd_pcm_hw_params_set_period_size().
I'm setting up the sampling rate of 8000Hz in setup_pcm() starting at line 111. I either use snd_pcm_hw_params_set_rate_near() or snd_pcm_hw_params_set_rate(), depending on whether the PCM supports the exact rate or not.
I set up the audio format SND_PCM_FORMAT_S16_LE at line 76 with snd_pcm_hw_params_set_format().
And finally, I also set up the number of channels (mono) in line 85 with snd_pcm_hw_params_set_channels().
Last but not least, snd_pcm_dump() shows that exactly these settings are actively used:
------------------------------------------------------------------------ ALSA <-> PulseAudio PCM I/O Plugin Its setup is: stream : CAPTURE [...] format : S16_LE [...] channels : 1 rate : 8000 exact rate : 8000 (8000/1) [...] period_size : 160 [...] avail_min : 160 ------------------------------------------------------------------------
In the above output you can see that the format, number of channels, rate, period size and avail_min are indeed set to correct values.
I verified that avail_min is 160 frames
is there any specific reason to choose 160 frames ?
Yes there is:
The audio frames are used for processing by a DSP lateron, which requires each speech packet (i.e. period) to have exactly 160 frames. It is also required that the audio frames are in S16_LE format, they have a sampling rate of 8kHz and they arrive at the DSP each 20ms (which corresponds to period_time).
As my code will use the atmel-pcm on an embedded target, the above mentioned constraints should be no problem.
In fact a look at the PCM in the alsa kernel sources (sound/soc/atmel/atmel-pcm.c) reveals: ------------------------------------------------------------------------ static const struct snd_pcm_hardware atmel_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE, .formats = SNDRV_PCM_FMTBIT_S16_LE, .period_bytes_min = 32, .period_bytes_max = 8192, .periods_min = 2, .periods_max = 1024, .buffer_bytes_max = 32 * 1024, }; ------------------------------------------------------------------------
However, as development on a slow ARM target can be a real pain, I am developing the code *on a PC* which is why the poll() behavior really is an issue (and maybe even a bug in alsa but more likely in pulseaudio).
As soon as the code is working, it will be easy to port it the ARM target.
cheers, stefan
2010/4/22 Stefan Schoenleitner dev.c0debabe@gmail.com
Raymond Yau wrote:
your program expect the driver support 2 periods per buffer but does not expicitly set the period
8000 Hz , S16_LE and mono
I am not sure why you think this is the case.
The period size is set at line 170 with snd_pcm_hw_params_set_period_size().
I'm setting up the sampling rate of 8000Hz in setup_pcm() starting at line 111. I either use snd_pcm_hw_params_set_rate_near() or snd_pcm_hw_params_set_rate(), depending on whether the PCM supports the exact rate or not.
I set up the audio format SND_PCM_FORMAT_S16_LE at line 76 with snd_pcm_hw_params_set_format().
And finally, I also set up the number of channels (mono) in line 85 with snd_pcm_hw_params_set_channels().
Last but not least, snd_pcm_dump() shows that exactly these settings are actively used:
ALSA <-> PulseAudio PCM I/O Plugin Its setup is: stream : CAPTURE [...] format : S16_LE [...] channels : 1 rate : 8000 exact rate : 8000 (8000/1) [...] period_size : 160 [...] avail_min : 160
In the above output you can see that the format, number of channels, rate, period size and avail_min are indeed set to correct values.
you output did not show the values of buffer size
PCM format is signed, linear, LE with 16 bits PCM rate 1 - 192000 Hz PCM period size: 64 - 699051 PCM buffer size: 480 - 163840 calculated buffer size: 640 ALSA <-> PulseAudio PCM I/O Plugin Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 640 period_size : 160 period_time : 20000 tstamp_mode : NONE period_step : 1 avail_min : 160 period_event : 0 start_threshold : 1 stop_threshold : 640 silence_threshold: 0 silence_size : 0 boundary : 1342177280 avail min: 160 start threshold: 160 frames stop threshold: 320 frames PCM format is signed, linear, LE with 16 bits PCM rate 1 - 192000 Hz PCM period size: 64 - 699051 PCM buffer size: 480 - 163840 calculated buffer size: 640 ALSA <-> PulseAudio PCM I/O Plugin Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 640 period_size : 160 period_time : 20000 tstamp_mode : NONE period_step : 1 avail_min : 160 period_event : 0 start_threshold : 1 stop_threshold : 640 silence_threshold: 0 silence_size : 0 boundary : 1342177280 avail min: 160 start threshold: 160 frames stop threshold: 320 frames capture fds: 1, playback fds: 1 capture poll fd: 3, playback poll fd: 8 capture struct: fd: 3, events: POLLIN POLLERR , revents: 0 playback struct: fd: 8, events: POLLIN POLLERR , revents: 0 demangled poll: on capture device demangled poll: POLLOUT on playback device wrote 160 frames demangled poll: on capture device demangled poll: POLLOUT on playback device wrote 160 frames demangled poll: on capture device demangled poll: POLLOUT on playback device wrote 160 frames demangled poll: on capture device demangled poll: POLLOUT on playback device wrote 160 frames demangled poll: POLLIN on capture device read 160 frames hexdump(): 320 bytes 0000 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0010 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0020 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 0030 00 00 02 00 04 00 ff ff ff ff fb ff f8 ff f9 ff 0040 f9 ff fe ff fc ff f9 ff f5 ff f8 ff 02 00 fe ff 0050 f7 ff ed ff f4 ff 01 00 fe ff ff ff f9 ff f4 ff 0060 f9 ff f6 ff f9 ff fb ff f6 ff fa ff fa ff fe ff 0070 00 00 fd ff fe ff fe ff fd ff fb ff fb ff fc ff 0080 fc ff fa ff fa ff f8 ff fd ff fe ff fa ff 00 00 0090 fa ff f6 ff fc ff fa ff fe ff 00 00 ff ff 00 00 00a0 fe ff fd ff fe ff fe ff 00 00 ff ff 01 00 fd ff 00b0 fb ff fd ff f8 ff fb ff 00 00 03 00 01 00 01 00 00c0 00 00 fa ff fb ff fe ff fb ff fc ff ff ff 00 00 00d0 04 00 05 00 02 00 01 00 fe ff 00 00 03 00 02 00 00e0 01 00 00 00 01 00 01 00 fe ff fc ff fc ff fc ff 00f0 fd ff fd ff 00 00 01 00 02 00 03 00 01 00 05 00 0100 fd ff f8 ff fb ff f8 ff fb ff fc ff f8 ff fd ff 0110 fd ff fe ff 02 00 01 00 00 00 00 00 fd ff 02 00 0120 03 00 01 00 ff ff fb ff 02 00 01 00 ff ff 04 00 0130 04 00 03 00 01 00 02 00 09 00 0f 00 0d 00 0c 00
demangled poll: on playback device demangled poll: on capture device demangled poll: POLLOUT on playback device wrote 160 frames demangled poll: POLLIN on capture device read 160 frames hexdump(): 320 bytes 0000 0a 00 07 00 06 00 07 00 0d 00 08 00 05 00 05 00 0010 05 00 07 00 09 00 0b 00 07 00 04 00 03 00 03 00 0020 02 00 01 00 07 00 0a 00 0a 00 06 00 00 00 fc ff 0030 fe ff 05 00 05 00 08 00 0a 00 05 00 05 00 06 00 0040 06 00 08 00 06 00 05 00 03 00 00 00 02 00 02 00 0050 fb ff f7 ff fa ff ff ff ff ff 00 00 fd ff f7 ff 0060 f5 ff f9 ff fb ff ff ff 01 00 03 00 01 00 fe ff 0070 fe ff ff ff 00 00 04 00 02 00 f7 ff fa ff ff ff 0080 02 00 04 00 01 00 04 00 07 00 00 00 fc ff ff ff 0090 fc ff fb ff f9 ff fd ff ff ff ff ff 03 00 fe ff 00a0 f7 ff f7 ff f7 ff f6 ff f8 ff f7 ff f7 ff f8 ff 00b0 f6 ff fe ff fe ff fc ff fc ff fa ff fc ff fc ff 00c0 fb ff f7 ff ff ff 00 00 01 00 05 00 fb ff f8 ff 00d0 f2 ff f7 ff fa ff fa ff fd ff f9 ff f8 ff f9 ff 00e0 fe ff 00 00 01 00 01 00 ff ff fd ff ff ff 04 00 00f0 00 00 fb ff f7 ff f4 ff fa ff fe ff 03 00 01 00 0100 00 00 ff ff fc ff 01 00 03 00 fe ff f9 ff f8 ff 0110 fa ff fd ff 03 00 03 00 02 00 fe ff f9 ff fd ff 0120 00 00 fa ff f7 ff fd ff fe ff fd ff 01 00 00 00 0130 fa ff f8 ff f7 ff f4 ff f6 ff fa ff f9 ff fa ff
demangled poll: on playback device demangled poll: on capture device demangled poll: POLLOUT on playback device wrote 160 frames demangled poll: POLLIN on capture device read 160 frames hexdump(): 320 bytes 0000 fd ff ff ff 03 00 01 00 fb ff fb ff fd ff fd ff 0010 f9 ff fb ff fd ff ff ff 02 00 00 00 fe ff fb ff 0020 fb ff fc ff 00 00 01 00 fe ff fe ff ff ff ff ff 0030 00 00 02 00 ff ff ff ff 00 00 fc ff fe ff ff ff 0040 fe ff fe ff fd ff ff ff fe ff ff ff 00 00 01 00 0050 04 00 06 00 02 00 fd ff 03 00 02 00 02 00 01 00 0060 fd ff fd ff ff ff 03 00 02 00 05 00 04 00 01 00 0070 00 00 fc ff 00 00 05 00 06 00 05 00 03 00 00 00 0080 fd ff 01 00 04 00 06 00 06 00 01 00 03 00 fd ff 0090 fa ff 05 00 06 00 04 00 00 00 fd ff fe ff fa ff 00a0 f8 ff 01 00 0a 00 08 00 08 00 05 00 ff ff fe ff 00b0 01 00 05 00 0a 00 09 00 03 00 01 00 01 00 04 00 00c0 01 00 04 00 07 00 06 00 09 00 06 00 03 00 00 00 00d0 fd ff fc ff 01 00 04 00 02 00 06 00 07 00 06 00 00e0 05 00 03 00 02 00 06 00 0a 00 09 00 02 00 fd ff 00f0 fb ff fa ff fd ff ff ff 01 00 ff ff f9 ff f5 ff 0100 f3 ff f7 ff fb ff f7 ff f6 ff fc ff fb ff fd ff 0110 fd ff f6 ff f7 ff fc ff f9 ff fb ff fc ff f5 ff 0120 f6 ff f7 ff f3 ff f6 ff f8 ff f7 ff f6 ff f9 ff 0130 fc ff f6 ff f4 ff f6 ff f4 ff fa ff f8 ff f2 ff
demangled poll: on playback device demangled poll: on capture device demangled poll: POLLOUT on playback device wrote 160 frames demangled poll: POLLIN on capture device could not read from capture device: Resource temporarily unavailable
cheers, stefan
Hi,
Raymond Yau wrote:
you output did not show the values of buffer size [...] 00d0 fd ff fc ff 01 00 04 00 02 00 06 00 07 00 06 00 00e0 05 00 03 00 02 00 06 00 0a 00 09 00 02 00 fd ff 00f0 fb ff fa ff fd ff ff ff 01 00 ff ff f9 ff f5 ff 0100 f3 ff f7 ff fb ff f7 ff f6 ff fc ff fb ff fd ff 0110 fd ff f6 ff f7 ff fc ff f9 ff fb ff fc ff f5 ff 0120 f6 ff f7 ff f3 ff f6 ff f8 ff f7 ff f6 ff f9 ff 0130 fc ff f6 ff f4 ff f6 ff f4 ff fa ff f8 ff f2 ff
demangled poll: on playback device demangled poll: on capture device demangled poll: POLLOUT on playback device wrote 160 frames demangled poll: POLLIN on capture device could not read from capture device: Resource temporarily unavailable
thanks for posting the full output. I added a call to snd_pcm_avail_update() in the code which shows how many frames are available after poll() returns.
As you can see in the output below, although avail_min is set to 160 frames, the POLLIN event on the capture PCM is returned after only 11 frames are available for reading:
------------------------------------------------------------------------------ snd_pcm_avail_update(playback): 0 demangled poll: on playback device snd_pcm_avail_update(capture): 171 demangled poll: POLLIN on capture device read 160 frames hexdump(): 320 bytes 0000 fc ff fc ff fc ff fc ff fc ff fc ff fc ff fe ff 0010 fc ff fd ff fb ff fc ff fc ff fc ff fd ff fc ff 0020 fc ff fc ff fd ff fc ff fc ff fc ff fd ff fd ff 0030 fc ff fc ff fc ff fc ff fc ff fb ff fc ff fc ff 0040 fd ff fc ff fc ff fc ff fc ff fc ff fd ff fc ff 0050 fd ff fc ff fc ff fc ff fd ff fc ff fc ff fa ff 0060 fc ff fc ff fc ff fb ff fc ff fd ff fc ff fd ff 0070 fc ff fc ff fd ff fc ff fc ff fc ff fc ff fc ff 0080 fb ff fd ff fc ff fc ff fc ff fc ff fc ff fc ff 0090 fb ff fd ff fc ff fd ff fb ff fd ff fc ff fc ff 00a0 fd ff fd ff fc ff fc ff fc ff fc ff fc ff fc ff 00b0 fc ff fc ff fb ff fc ff fc ff fc ff fc ff fd ff 00c0 fc ff fc ff fd ff fc ff fc ff fc ff fc ff fc ff 00d0 fc ff fc ff fc ff fc ff fc ff fd ff fc ff fc ff 00e0 fd ff fc ff fc ff fc ff fd ff fd ff fc ff fc ff 00f0 fc ff fb ff fc ff fd ff fd ff fc ff fc ff fc ff 0100 fc ff fc ff fc ff fc ff fc ff fc ff fc ff fc ff 0110 fc ff fc ff fc ff fc ff fc ff fc ff fc ff fc ff 0120 fc ff fc ff fc ff fc ff fd ff fc ff fc ff fc ff 0130 fc ff fc ff fc ff fc ff fc ff fc ff fc ff fc ff
snd_pcm_avail_update(playback): 0 demangled poll: on playback device snd_pcm_avail_update(capture): 11 demangled poll: POLLIN on capture device could not read from capture device: Resource temporarily unavailable ALSA <-> PulseAudio PCM I/O Plugin Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 640 period_size : 160 period_time : 20000 tstamp_mode : NONE period_step : 1 avail_min : 160 period_event : 0 start_threshold : 160 stop_threshold : 320 silence_threshold: 0 silence_size : 0 boundary : 5764607523034234880 ------------------------------------------------------------------------------
In one of your previous posts you also said that the pulseaudio log would be helpful. You can see it below. In order to keep it brief, I included only the logs between pulseaudio startup and the end of the audio application.
------------------------------------------------------------------------------ $ pulseaudio -vvv I: caps.c: Limited capabilities successfully to CAP_SYS_NICE. I: caps.c: Dropping root privileges. I: caps.c: Limited capabilities successfully to CAP_SYS_NICE. D: main.c: Started as real root: no, suid root: yes I: main.c: We're in the group 'pulse-rt', allowing high-priority scheduling. I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted I: core-util.c: Successfully gained nice level -11. D: main.c: Can realtime: yes, can high-priority: yes I: main.c: Giving up CAP_NICE D: main.c: Can realtime: no, can high-priority: no I: main.c: This is PulseAudio 0.9.14 D: main.c: Compilation host: x86_64-pc-linux-gnu D: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pedantic -pipe -Wno-long-long -Wvla -Wno-overlength-strings -Wconversion -Wundef -Wformat -Wlogical-op -Wpacked -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wdeclaration-after-statement -Wfloat-equal -Wmissing-declarations -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-noreturn -Wshadow -Wendif-labels -Wpointer-arith -Wcast-align -Wwrite-strings -Wno-unused-parameter -ffast-math D: main.c: Running on host: Linux x86_64 2.6.28-18-generic #59-Ubuntu SMP Thu Jan 28 01:40:19 UTC 2010 I: main.c: Page size is 4096 bytes D: main.c: Compiled with Valgrind support: no D: main.c: Running in valgrind mode: no D: main.c: Optimized build: yes I: main.c: Machine ID is d05a505180ca4c5382d620fc498da356. I: main.c: Using runtime directory /home/mne/.pulse/d05a505180ca4c5382d620fc498da356:runtime. I: main.c: Using state directory /home/mne/.pulse. I: main.c: Running in system mode: no I: main.c: Fresh high-resolution timers available! Bon appetit! D: memblock.c: Using shared memory pool with 1024 slots of size 64.0 KiB each, total size is 64.0 MiB, maximum usable slot size is 65472 D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-gconf.so': success D: module-gconf.c: Loading module 'module-combine' with args '' due to GConf configuration. I: sink.c: Created sink 0 "combined" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: source.c: Created source 0 "combined.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right D: module-combine.c: Thread starting up D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+29 I: module.c: Loaded "module-combine" (index: #0; argument: ""). I: module.c: Loaded "module-gconf" (index: #1; argument: ""). D: module-suspend-on-idle.c: Sink combined becomes idle. D: module-suspend-on-idle.c: Source combined.monitor becomes idle. I: module.c: Loaded "module-suspend-on-idle" (index: #2; argument: ""). I: module-device-restore.c: Sucessfully opened database file '/home/mne/.pulse/d05a505180ca4c5382d620fc498da356:device-volumes.x86_64-pc-linux-gnu.gdbm'. I: module.c: Loaded "module-device-restore" (index: #3; argument: ""). I: module-stream-restore.c: Sucessfully opened database file '/home/mne/.pulse/d05a505180ca4c5382d620fc498da356:stream-volumes.x86_64-pc-linux-gnu.gdbm'. I: module.c: Loaded "module-stream-restore" (index: #4; argument: ""). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-hal-detect.so': success I: module-hal-detect.c: Trying capability alsa D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/computer_alsa_timer D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/computer_alsa_sequencer D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_playback_4 D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_midi_3 D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_playback_2 D: module-hal-detect.c: Loading module-alsa-source with arguments 'device_id=0 source_name=alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 tsched=0' D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... I: module-alsa-source.c: Successfully opened device front:0. I: module-alsa-source.c: Successfully enabled mmap() mode. I: (alsa-lib)control.c: Invalid CTL front:0 I: alsa-util.c: Unable to attach to mixer front:0: No such file or directory I: alsa-util.c: Successfully attached to mixer 'hw:0' I: alsa-util.c: Cannot find mixer control "Capture" or mixer control is no combination of switch/volume. I: alsa-util.c: Using mixer control "Mic". I: module-device-restore.c: Restoring volume for source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0. I: module-device-restore.c: Restoring mute state for source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0. I: source.c: Created source 1 "alsa_input.pci_1102_4_sound_card_0_alsa_capture_0" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: module-alsa-source.c: Using 2 fragments of size 7168 bytes, buffer time is 81.27ms D: module-alsa-source.c: hwbuf_unused=0 D: module-alsa-source.c: setting avail_min=1 I: module-alsa-source.c: Volume ranges from 0 to 31. I: module-alsa-source.c: Volume ranges from -34.50 dB to 12.00 dB. I: alsa-util.c: ALSA device lacks independant volume controls for each channel. I: module-alsa-source.c: Using hardware volume control. Hardware dB scale supported. D: alsa-util.c: snd_pcm_dump(): D: alsa-util.c: Hooks PCM D: alsa-util.c: Its setup is: D: alsa-util.c: stream : CAPTURE D: alsa-util.c: access : MMAP_INTERLEAVED D: alsa-util.c: format : S16_LE D: alsa-util.c: subformat : STD D: alsa-util.c: channels : 2 D: alsa-util.c: rate : 44100 D: alsa-util.c: exact rate : 44100 (44100/1) D: alsa-util.c: msbits : 16 D: alsa-util.c: buffer_size : 3584 D: alsa-util.c: period_size : 1792 D: alsa-util.c: period_time : 40634 D: alsa-util.c: tstamp_mode : ENABLE D: alsa-util.c: period_step : 1 D: alsa-util.c: avail_min : 1792 D: alsa-util.c: period_event : 0 D: alsa-util.c: start_threshold : -1 D: alsa-util.c: stop_threshold : 8070450532247928832 D: alsa-util.c: silence_threshold: 0 D: alsa-util.c: silence_size : 0 D: alsa-util.c: boundary : 8070450532247928832 D: alsa-util.c: Slave: Hardware PCM card 0 'Audigy 2 [SB0240]' device 0 subdevice 0 D: alsa-util.c: Its setup is: D: alsa-util.c: stream : CAPTURE D: alsa-util.c: access : MMAP_INTERLEAVED D: alsa-util.c: format : S16_LE D: alsa-util.c: subformat : STD D: alsa-util.c: channels : 2 D: alsa-util.c: rate : 44100 D: alsa-util.c: exact rate : 44100 (44100/1) D: alsa-util.c: msbits : 16 D: alsa-util.c: buffer_size : 3584 D: alsa-util.c: period_size : 1792 D: alsa-util.c: period_time : 40634 D: alsa-util.c: tstamp_mode : ENABLE D: alsa-util.c: period_step : 1 D: alsa-util.c: avail_min : 1792 D: alsa-util.c: period_event : 0 D: alsa-util.c: start_threshold : -1 D: alsa-util.c: stop_threshold : 807045053224 D: module-alsa-source.c: Thread starting up D: module-alsa-source.c: Requested volume: 0: 100% 1: 100% D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+28 D: module-alsa-source.c: Got hardware volume: 0: 100% 1: 100% D: module-alsa-source.c: Calculated software volume: 0: 100% 1: 100% D: module-suspend-on-idle.c: Source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 becomes idle. I: module.c: Loaded "module-alsa-source" (index: #5; argument: "device_id=0 source_name=alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 tsched=0"). D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_capture_4 D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_playback_3 D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_capture_2 D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_midi_2 D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_capture_1 D: module-hal-detect.c: Loading module-alsa-sink with arguments 'device_id=0 sink_name=alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 tsched=0' D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying front:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:front:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:front:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem I: alsa-util.c: PCM device plug:front:0 refused our hw parameters: Device or resource busy D: alsa-util.c: Trying surround40:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying surround40:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:surround40:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:surround40:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem I: alsa-util.c: PCM device plug:surround40:0 refused our hw parameters: Device or resource busy D: alsa-util.c: Trying surround41:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying surround41:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:surround41:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:surround41:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem I: alsa-util.c: PCM device plug:surround41:0 refused our hw parameters: Device or resource busy D: alsa-util.c: Trying surround50:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying surround50:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:surround50:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:surround50:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem I: alsa-util.c: PCM device plug:surround50:0 refused our hw parameters: Device or resource busy D: alsa-util.c: Trying surround51:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying surround51:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:surround51:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:surround51:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem I: alsa-util.c: PCM device plug:surround51:0 refused our hw parameters: Device or resource busy D: alsa-util.c: Trying surround71:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying surround71:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:surround71:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:surround71:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem I: alsa-util.c: PCM device plug:surround71:0 refused our hw parameters: Device or resource busy D: alsa-util.c: Trying hw:0 as last resort... D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ... I: module-alsa-sink.c: Successfully opened device hw:0. I: module-alsa-sink.c: Successfully enabled mmap() mode. I: alsa-util.c: Successfully attached to mixer 'hw:0' I: alsa-util.c: Cannot find mixer control "Master" or mixer control is no combination of switch/volume. W: alsa-util.c: Cannot find fallback mixer control "PCM" or mixer control is no combination of switch/volume. I: alsa-util.c: Using mixer control "PCM". I: module-device-restore.c: Restoring volume for sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_0. I: module-device-restore.c: Restoring mute state for sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_0. I: sink.c: Created sink 1 "alsa_output.pci_1102_4_sound_card_0_alsa_playback_0" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: module-device-restore.c: Restoring volume for source alsa_output.pci_1102_4_sound_card_0_alsa_playback_0.monitor. I: module-device-restore.c: Restoring mute state for source alsa_output.pci_1102_4_sound_card_0_alsa_playback_0.monitor. I: source.c: Created source 2 "alsa_output.pci_1102_4_sound_card_0_alsa_playback_0.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: module-alsa-sink.c: Using 8 fragments of size 1764 bytes, buffer time is 80.00ms D: module-alsa-sink.c: hwbuf_unused=0 D: module-alsa-sink.c: setting avail_min=1 I: module-alsa-sink.c: Volume ranges from 0 to 100. I: module-alsa-sink.c: Volume ranges from -40.00 dB to 0.00 dB. I: alsa-util.c: ALSA device lacks independant volume controls for each channel. I: module-alsa-sink.c: Using hardware volume control. Hardware dB scale supported. I: module-alsa-sink.c: Using software mute control. D: alsa-util.c: snd_pcm_dump(): D: alsa-util.c: Hardware PCM card 0 'Audigy 2 [SB0240]' device 0 subdevice 0 D: alsa-util.c: Its setup is: D: alsa-util.c: stream : PLAYBACK D: alsa-util.c: access : MMAP_INTERLEAVED D: alsa-util.c: format : S16_LE D: alsa-util.c: subformat : STD D: alsa-util.c: channels : 2 D: alsa-util.c: rate : 44100 D: alsa-util.c: exact rate : 44100 (44100/1) D: alsa-util.c: msbits : 16 D: alsa-util.c: buffer_size : 3528 D: alsa-util.c: period_size : 441 D: alsa-util.c: period_time : 10000 D: alsa-util.c: tstamp_mode : ENABLE D: alsa-util.c: period_step : 1 D: alsa-util.c: avail_min : 441 D: alsa-util.c: period_event : 0 D: alsa-util.c: start_threshold : -1 D: alsa-util.c: stop_threshold : 7944349742681554944 D: alsa-util.c: silence_threshold: 0 D: alsa-util.c: silence_size : 0 D: alsa-util.c: boundary : 7944349742681554944 D: module-alsa-sink.c: Thread starting up D: module-alsa-sink.c: Requested volume: 0: 77% 1: 77% D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+27 D: module-alsa-sink.c: Got hardware volume: 0: 77% 1: 77% D: module-alsa-sink.c: Calculated software volume: 0: 100% 1: 100% I: module-alsa-sink.c: Starting playback. D: module-suspend-on-idle.c: Source alsa_output.pci_1102_4_sound_card_0_alsa_playback_0.monitor becomes idle. D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 becomes idle. I: module-combine.c: Configuring new sink: alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 D: memblockq.c: memblockq requested: maxlength=16777216, tlength=16777216, base=4, prebuf=1, minreq=0 maxrewind=0 D: memblockq.c: memblockq sanitized: maxlength=16777216, tlength=16777216, base=4, prebuf=4, minreq=4 maxrewind=0 I: module-stream-restore.c: Not restore device for stream sink-input-by-media-role:filter, because already set. I: module-stream-restore.c: Restoring volume for sink input sink-input-by-media-role:filter. I: module-stream-restore.c: Restoring mute state for sink input sink-input-by-media-role:filter. D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 becomes busy. I: resampler.c: Using resampler 'trivial' I: resampler.c: Using s16le as working format. D: memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0 D: memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0 I: sink-input.c: Created input 0 "Simultaneous output on Audigy 2 [SB0240] - ADC Capture/Standard PCM Playback" on alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 with sample spec s16le 2ch 44100Hz and channel map front-left,front-right D: module-alsa-sink.c: hwbuf_unused=0 D: module-alsa-sink.c: setting avail_min=1 I: module.c: Loaded "module-alsa-sink" (index: #6; argument: "device_id=0 sink_name=alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 tsched=0"). D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_midi_1 D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_midi_0 D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_hw_specific_2 D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_hw_specific_0 D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_control__1 I: module-hal-detect.c: Loaded 2 modules. I: module.c: Loaded "module-hal-detect" (index: #7; argument: "tsched=0"). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-esound-protocol-unix.so': success I: module.c: Loaded "module-esound-protocol-unix" (index: #8; argument: ""). I: module.c: Loaded "module-native-protocol-unix" (index: #9; argument: ""). I: module-default-device-restore.c: Restored default sink 'alsa_output.pci_1102_4_sound_card_0_alsa_playback_0'. D: core-subscribe.c: Dropped redundant event due to change event. I: module-default-device-restore.c: Restored default source 'alsa_input.pci_1102_4_sound_card_0_alsa_capture_0'. I: module.c: Loaded "module-default-device-restore" (index: #10; argument: ""). I: module.c: Loaded "module-rescue-streams" (index: #11; argument: ""). I: module.c: Loaded "module-always-sink" (index: #12; argument: ""). I: client.c: Created 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session3" D: module-console-kit.c: Added new session /org/freedesktop/ConsoleKit/Session3 I: module.c: Loaded "module-console-kit" (index: #13; argument: ""). I: module.c: Loaded "module-position-event-sounds" (index: #14; argument: ""). I: main.c: Daemon startup complete. D: module-hal-detect.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired D: module-console-kit.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired D: module-hal-detect.c: dbus: interface=org.freedesktop.ConsoleKit.Seat, path=/org/freedesktop/ConsoleKit/Seat1, member=SessionAdded D: module-console-kit.c: dbus: interface=org.freedesktop.ConsoleKit.Seat, path=/org/freedesktop/ConsoleKit/Seat1, member=SessionAdded D: module-hal-detect.c: dbus: interface=org.freedesktop.ConsoleKit.Seat, path=/org/freedesktop/ConsoleKit/Seat1, member=SessionRemoved D: module-console-kit.c: dbus: interface=org.freedesktop.ConsoleKit.Seat, path=/org/freedesktop/ConsoleKit/Seat1, member=SessionRemoved ^CI: main.c: Got signal SIGINT. I: main.c: Exiting. I: main.c: Daemon shutdown initiated. I: module.c: Unloading "module-combine" (index: #0). D: module-alsa-sink.c: hwbuf_unused=0 D: module-alsa-sink.c: setting avail_min=1 D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 becomes idle. D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 becomes idle. I: sink-input.c: Freeing input 0 "Simultaneous output on Audigy 2 [SB0240] - ADC Capture/Standard PCM Playback" D: module-rescue-streams.c: No sink inputs to move away. D: module-rescue-streams.c: No source outputs to move away. D: core-subscribe.c: Dropped redundant event due to remove event. D: core-subscribe.c: Dropped redundant event due to remove event. D: module-combine.c: Thread shutting down I: sink.c: Freeing sink 0 "combined" I: source.c: Freeing source 0 "combined.monitor" I: module.c: Unloaded "module-combine" (index: #0). I: module.c: Unloading "module-gconf" (index: #1). D: module-gconf.c: Unloading module #0 I: module.c: Unloaded "module-gconf" (index: #1). I: module.c: Unloading "module-suspend-on-idle" (index: #2). I: module.c: Unloaded "module-suspend-on-idle" (index: #2). I: module.c: Unloading "module-device-restore" (index: #3). I: module.c: Unloaded "module-device-restore" (index: #3). I: module.c: Unloading "module-stream-restore" (index: #4). I: module.c: Unloaded "module-stream-restore" (index: #4). I: module.c: Unloading "module-alsa-source" (index: #5). D: module-rescue-streams.c: No source outputs to move away. D: module-alsa-source.c: Thread shutting down I: source.c: Freeing source 1 "alsa_input.pci_1102_4_sound_card_0_alsa_capture_0" I: module.c: Unloaded "module-alsa-source" (index: #5). I: module.c: Unloading "module-alsa-sink" (index: #6). D: module-always-sink.c: Autoloading null-sink as no other sinks detected. I: sink.c: Created sink 2 "auto_null" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: source.c: Created source 3 "auto_null.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right D: module-null-sink.c: Thread starting up D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+29 I: module.c: Loaded "module-null-sink" (index: #15; argument: "sink_name=auto_null"). D: module-rescue-streams.c: No sink inputs to move away. D: module-rescue-streams.c: No source outputs to move away. D: module-alsa-sink.c: Thread shutting down I: sink.c: Freeing sink 1 "alsa_output.pci_1102_4_sound_card_0_alsa_playback_0" I: source.c: Freeing source 2 "alsa_output.pci_1102_4_sound_card_0_alsa_playback_0.monitor" I: module.c: Unloaded "module-alsa-sink" (index: #6). I: module.c: Unloading "module-hal-detect" (index: #7). I: module.c: Unloaded "module-hal-detect" (index: #7). I: module.c: Unloading "module-esound-protocol-unix" (index: #8). I: module.c: Unloaded "module-esound-protocol-unix" (index: #8). I: module.c: Unloading "module-native-protocol-unix" (index: #9). I: module.c: Unloaded "module-native-protocol-unix" (index: #9). I: module.c: Unloading "module-default-device-restore" (index: #10). I: module.c: Unloaded "module-default-device-restore" (index: #10). I: module.c: Unloading "module-rescue-streams" (index: #11). I: module.c: Unloaded "module-rescue-streams" (index: #11). I: module.c: Unloading "module-always-sink" (index: #12). I: module.c: Unloaded "module-always-sink" (index: #12). I: module.c: Unloading "module-console-kit" (index: #13). D: module-console-kit.c: Removing session /org/freedesktop/ConsoleKit/Session3 I: client.c: Freed 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session3" I: module.c: Unloaded "module-console-kit" (index: #13). I: module.c: Unloading "module-position-event-sounds" (index: #14). I: module.c: Unloaded "module-position-event-sounds" (index: #14). I: module.c: Unloading "module-null-sink" (index: #15). D: core-subscribe.c: Dropped redundant event due to remove event. D: core-subscribe.c: Dropped redundant event due to remove event. D: module-null-sink.c: Thread shutting down I: sink.c: Freeing sink 2 "auto_null" I: source.c: Freeing source 3 "auto_null.monitor" I: module.c: Unloaded "module-null-sink" (index: #15). D: core-subscribe.c: Dropped redundant event due to remove event. I: main.c: Daemon terminated. mne@nanoflex:~$ pulseaudio -vvv I: caps.c: Limited capabilities successfully to CAP_SYS_NICE. I: caps.c: Dropping root privileges. I: caps.c: Limited capabilities successfully to CAP_SYS_NICE. D: main.c: Started as real root: no, suid root: yes I: main.c: We're in the group 'pulse-rt', allowing high-priority scheduling. I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted I: core-util.c: Successfully gained nice level -11. D: main.c: Can realtime: yes, can high-priority: yes I: main.c: Giving up CAP_NICE D: main.c: Can realtime: no, can high-priority: no I: main.c: This is PulseAudio 0.9.14 D: main.c: Compilation host: x86_64-pc-linux-gnu D: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pedantic -pipe -Wno-long-long -Wvla -Wno-overlength-strings -Wconversion -Wundef -Wformat -Wlogical-op -Wpacked -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wdeclaration-after-statement -Wfloat-equal -Wmissing-declarations -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-noreturn -Wshadow -Wendif-labels -Wpointer-arith -Wcast-align -Wwrite-strings -Wno-unused-parameter -ffast-math D: main.c: Running on host: Linux x86_64 2.6.28-18-generic #59-Ubuntu SMP Thu Jan 28 01:40:19 UTC 2010 I: main.c: Page size is 4096 bytes D: main.c: Compiled with Valgrind support: no D: main.c: Running in valgrind mode: no D: main.c: Optimized build: yes I: main.c: Machine ID is d05a505180ca4c5382d620fc498da356. I: main.c: Using runtime directory /home/mne/.pulse/d05a505180ca4c5382d620fc498da356:runtime. I: main.c: Using state directory /home/mne/.pulse. I: main.c: Running in system mode: no I: main.c: Fresh high-resolution timers available! Bon appetit! D: memblock.c: Using shared memory pool with 1024 slots of size 64.0 KiB each, total size is 64.0 MiB, maximum usable slot size is 65472 D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-gconf.so': success D: module-gconf.c: Loading module 'module-combine' with args '' due to GConf configuration. I: sink.c: Created sink 0 "combined" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: source.c: Created source 0 "combined.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right D: module-combine.c: Thread starting up D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+29 I: module.c: Loaded "module-combine" (index: #0; argument: ""). I: module.c: Loaded "module-gconf" (index: #1; argument: ""). D: module-suspend-on-idle.c: Sink combined becomes idle. D: module-suspend-on-idle.c: Source combined.monitor becomes idle. I: module.c: Loaded "module-suspend-on-idle" (index: #2; argument: ""). I: module-device-restore.c: Sucessfully opened database file '/home/mne/.pulse/d05a505180ca4c5382d620fc498da356:device-volumes.x86_64-pc-linux-gnu.gdbm'. I: module.c: Loaded "module-device-restore" (index: #3; argument: ""). I: module-stream-restore.c: Sucessfully opened database file '/home/mne/.pulse/d05a505180ca4c5382d620fc498da356:stream-volumes.x86_64-pc-linux-gnu.gdbm'. I: module.c: Loaded "module-stream-restore" (index: #4; argument: ""). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-hal-detect.so': success I: module-hal-detect.c: Trying capability alsa D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/computer_alsa_timer D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/computer_alsa_sequencer D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_playback_4 D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_midi_3 D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_playback_2 D: module-hal-detect.c: Loading module-alsa-source with arguments 'device_id=0 source_name=alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 tsched=0' D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... I: module-alsa-source.c: Successfully opened device front:0. I: module-alsa-source.c: Successfully enabled mmap() mode. I: (alsa-lib)control.c: Invalid CTL front:0 I: alsa-util.c: Unable to attach to mixer front:0: No such file or directory I: alsa-util.c: Successfully attached to mixer 'hw:0' I: alsa-util.c: Cannot find mixer control "Capture" or mixer control is no combination of switch/volume. I: alsa-util.c: Using mixer control "Mic". I: module-device-restore.c: Restoring volume for source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0. I: module-device-restore.c: Restoring mute state for source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0. I: source.c: Created source 1 "alsa_input.pci_1102_4_sound_card_0_alsa_capture_0" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: module-alsa-source.c: Using 2 fragments of size 7168 bytes, buffer time is 81.27ms D: module-alsa-source.c: hwbuf_unused=0 D: module-alsa-source.c: setting avail_min=1 I: module-alsa-source.c: Volume ranges from 0 to 31. I: module-alsa-source.c: Volume ranges from -34.50 dB to 12.00 dB. I: alsa-util.c: ALSA device lacks independant volume controls for each channel. I: module-alsa-source.c: Using hardware volume control. Hardware dB scale supported. D: alsa-util.c: snd_pcm_dump(): D: alsa-util.c: Hooks PCM D: alsa-util.c: Its setup is: D: alsa-util.c: stream : CAPTURE D: alsa-util.c: access : MMAP_INTERLEAVED D: alsa-util.c: format : S16_LE D: alsa-util.c: subformat : STD D: alsa-util.c: channels : 2 D: alsa-util.c: rate : 44100 D: alsa-util.c: exact rate : 44100 (44100/1) D: alsa-util.c: msbits : 16 D: alsa-util.c: buffer_size : 3584 D: alsa-util.c: period_size : 1792 D: alsa-util.c: period_time : 40634 D: alsa-util.c: tstamp_mode : ENABLE D: alsa-util.c: period_step : 1 D: alsa-util.c: avail_min : 1792 D: alsa-util.c: period_event : 0 D: alsa-util.c: start_threshold : -1 D: alsa-util.c: stop_threshold : 8070450532247928832 D: alsa-util.c: silence_threshold: 0 D: alsa-util.c: silence_size : 0 D: alsa-util.c: boundary : 8070450532247928832 D: alsa-util.c: Slave: Hardware PCM card 0 'Audigy 2 [SB0240]' device 0 subdevice 0 D: alsa-util.c: Its setup is: D: alsa-util.c: stream : CAPTURE D: alsa-util.c: access : MMAP_INTERLEAVED D: alsa-util.c: format : S16_LE D: alsa-util.c: subformat : STD D: alsa-util.c: channels : 2 D: alsa-util.c: rate : 44100 D: alsa-util.c: exact rate : 44100 (44100/1) D: alsa-util.c: msbits : 16 D: alsa-util.c: buffer_size : 3584 D: alsa-util.c: period_size : 1792 D: alsa-util.c: period_time : 40634 D: alsa-util.c: tstamp_mode : ENABLE D: alsa-util.c: period_step : 1 D: alsa-util.c: avail_min : 1792 D: alsa-util.c: period_event : 0 D: alsa-util.c: start_threshold : -1 D: alsa-util.c: stop_threshold : 807045053224 D: module-alsa-source.c: Thread starting up D: module-alsa-source.c: Requested volume: 0: 100% 1: 100% D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+28 D: module-alsa-source.c: Got hardware volume: 0: 100% 1: 100% D: module-alsa-source.c: Calculated software volume: 0: 100% 1: 100% D: module-suspend-on-idle.c: Source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 becomes idle. I: module.c: Loaded "module-alsa-source" (index: #5; argument: "device_id=0 source_name=alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 tsched=0"). D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_capture_4 D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_playback_3 D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_capture_2 D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_midi_2 D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_capture_1 D: module-hal-detect.c: Loading module-alsa-sink with arguments 'device_id=0 sink_name=alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 tsched=0' D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying front:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:front:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:front:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem I: alsa-util.c: PCM device plug:front:0 refused our hw parameters: Device or resource busy D: alsa-util.c: Trying surround40:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying surround40:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:surround40:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:surround40:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem I: alsa-util.c: PCM device plug:surround40:0 refused our hw parameters: Device or resource busy D: alsa-util.c: Trying surround41:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying surround41:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:surround41:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:surround41:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem I: alsa-util.c: PCM device plug:surround41:0 refused our hw parameters: Device or resource busy D: alsa-util.c: Trying surround50:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying surround50:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:surround50:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:surround50:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem I: alsa-util.c: PCM device plug:surround50:0 refused our hw parameters: Device or resource busy D: alsa-util.c: Trying surround51:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying surround51:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:surround51:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:surround51:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem I: alsa-util.c: PCM device plug:surround51:0 refused our hw parameters: Device or resource busy D: alsa-util.c: Trying surround71:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying surround71:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:surround71:0 with SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem D: alsa-util.c: Trying plug:surround71:0 without SND_PCM_NO_AUTO_FORMAT ... I: (alsa-lib)setup.c: Cannot lock ctl elem I: alsa-util.c: PCM device plug:surround71:0 refused our hw parameters: Device or resource busy D: alsa-util.c: Trying hw:0 as last resort... D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ... I: module-alsa-sink.c: Successfully opened device hw:0. I: module-alsa-sink.c: Successfully enabled mmap() mode. I: alsa-util.c: Successfully attached to mixer 'hw:0' I: alsa-util.c: Cannot find mixer control "Master" or mixer control is no combination of switch/volume. W: alsa-util.c: Cannot find fallback mixer control "PCM" or mixer control is no combination of switch/volume. I: alsa-util.c: Using mixer control "PCM". I: module-device-restore.c: Restoring volume for sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_0. I: module-device-restore.c: Restoring mute state for sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_0. I: sink.c: Created sink 1 "alsa_output.pci_1102_4_sound_card_0_alsa_playback_0" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: module-device-restore.c: Restoring volume for source alsa_output.pci_1102_4_sound_card_0_alsa_playback_0.monitor. I: module-device-restore.c: Restoring mute state for source alsa_output.pci_1102_4_sound_card_0_alsa_playback_0.monitor. I: source.c: Created source 2 "alsa_output.pci_1102_4_sound_card_0_alsa_playback_0.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: module-alsa-sink.c: Using 8 fragments of size 1764 bytes, buffer time is 80.00ms D: module-alsa-sink.c: hwbuf_unused=0 D: module-alsa-sink.c: setting avail_min=1 I: module-alsa-sink.c: Volume ranges from 0 to 100. I: module-alsa-sink.c: Volume ranges from -40.00 dB to 0.00 dB. I: alsa-util.c: ALSA device lacks independant volume controls for each channel. I: module-alsa-sink.c: Using hardware volume control. Hardware dB scale supported. I: module-alsa-sink.c: Using software mute control. D: alsa-util.c: snd_pcm_dump(): D: alsa-util.c: Hardware PCM card 0 'Audigy 2 [SB0240]' device 0 subdevice 0 D: alsa-util.c: Its setup is: D: alsa-util.c: stream : PLAYBACK D: alsa-util.c: access : MMAP_INTERLEAVED D: alsa-util.c: format : S16_LE D: alsa-util.c: subformat : STD D: alsa-util.c: channels : 2 D: alsa-util.c: rate : 44100 D: alsa-util.c: exact rate : 44100 (44100/1) D: alsa-util.c: msbits : 16 D: alsa-util.c: buffer_size : 3528 D: alsa-util.c: period_size : 441 D: alsa-util.c: period_time : 10000 D: alsa-util.c: tstamp_mode : ENABLE D: alsa-util.c: period_step : 1 D: alsa-util.c: avail_min : 441 D: alsa-util.c: period_event : 0 D: alsa-util.c: start_threshold : -1 D: alsa-util.c: stop_threshold : 7944349742681554944 D: alsa-util.c: silence_threshold: 0 D: alsa-util.c: silence_size : 0 D: alsa-util.c: boundary : 7944349742681554944 D: module-alsa-sink.c: Thread starting up D: module-alsa-sink.c: Requested volume: 0: 77% 1: 77% D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+27 D: module-alsa-sink.c: Got hardware volume: 0: 77% 1: 77% D: module-alsa-sink.c: Calculated software volume: 0: 100% 1: 100% I: module-alsa-sink.c: Starting playback. D: module-suspend-on-idle.c: Source alsa_output.pci_1102_4_sound_card_0_alsa_playback_0.monitor becomes idle. D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 becomes idle. I: module-combine.c: Configuring new sink: alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 D: memblockq.c: memblockq requested: maxlength=16777216, tlength=16777216, base=4, prebuf=1, minreq=0 maxrewind=0 D: memblockq.c: memblockq sanitized: maxlength=16777216, tlength=16777216, base=4, prebuf=4, minreq=4 maxrewind=0 I: module-stream-restore.c: Not restore device for stream sink-input-by-media-role:filter, because already set. I: module-stream-restore.c: Restoring volume for sink input sink-input-by-media-role:filter. I: module-stream-restore.c: Restoring mute state for sink input sink-input-by-media-role:filter. D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 becomes busy. I: resampler.c: Using resampler 'trivial' I: resampler.c: Using s16le as working format. D: memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0 D: memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0 I: sink-input.c: Created input 0 "Simultaneous output on Audigy 2 [SB0240] - ADC Capture/Standard PCM Playback" on alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 with sample spec s16le 2ch 44100Hz and channel map front-left,front-right D: module-alsa-sink.c: hwbuf_unused=0 D: module-alsa-sink.c: setting avail_min=1 I: module.c: Loaded "module-alsa-sink" (index: #6; argument: "device_id=0 sink_name=alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 tsched=0"). D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_midi_1 D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_midi_0 D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_hw_specific_2 D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_hw_specific_0 D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_1102_4_sound_card_0_alsa_control__1 I: module-hal-detect.c: Loaded 2 modules. I: module.c: Loaded "module-hal-detect" (index: #7; argument: "tsched=0"). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9/modules/module-esound-protocol-unix.so': success I: module.c: Loaded "module-esound-protocol-unix" (index: #8; argument: ""). I: module.c: Loaded "module-native-protocol-unix" (index: #9; argument: ""). I: module-default-device-restore.c: Restored default sink 'alsa_output.pci_1102_4_sound_card_0_alsa_playback_0'. D: core-subscribe.c: Dropped redundant event due to change event. I: module-default-device-restore.c: Restored default source 'alsa_input.pci_1102_4_sound_card_0_alsa_capture_0'. I: module.c: Loaded "module-default-device-restore" (index: #10; argument: ""). I: module.c: Loaded "module-rescue-streams" (index: #11; argument: ""). I: module.c: Loaded "module-always-sink" (index: #12; argument: ""). I: client.c: Created 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session3" D: module-console-kit.c: Added new session /org/freedesktop/ConsoleKit/Session3 I: module.c: Loaded "module-console-kit" (index: #13; argument: ""). I: module.c: Loaded "module-position-event-sounds" (index: #14; argument: ""). I: main.c: Daemon startup complete. D: module-hal-detect.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired D: module-console-kit.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired I: module-suspend-on-idle.c: Source combined.monitor idle for too long, suspending ... I: module-suspend-on-idle.c: Sink combined idle for too long, suspending ... D: module-alsa-sink.c: hwbuf_unused=0 D: module-alsa-sink.c: setting avail_min=1 D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 becomes idle. D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 becomes idle. I: sink-input.c: Freeing input 0 "Simultaneous output on Audigy 2 [SB0240] - ADC Capture/Standard PCM Playback" I: module-combine.c: Device suspended... I: module-suspend-on-idle.c: Source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 idle for too long, suspending ... I: module-alsa-source.c: Device suspended... I: module-suspend-on-idle.c: Source alsa_output.pci_1102_4_sound_card_0_alsa_playback_0.monitor idle for too long, suspending ... I: client.c: Created 1 "Native client (UNIX socket client)" I: client.c: Freed 1 "Native client (UNIX socket client)" I: protocol-native.c: Connection died. I: client.c: Created 2 "Native client (UNIX socket client)" D: protocol-native.c: Protocol version: remote 14, local 14 I: protocol-native.c: Got credentials: uid=1000 gid=1000 success=1 D: protocol-native.c: SHM possible: yes D: protocol-native.c: Negotiated SHM: yes I: module-stream-restore.c: Restoring device for stream source-output-by-application-name:ALSA plug-in [duplex]. I: module-alsa-source.c: Trying resume... D: module-alsa-source.c: hwbuf_unused=0 D: module-alsa-source.c: setting avail_min=1 I: module-alsa-source.c: Resumed successfully... D: module-suspend-on-idle.c: Source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 becomes idle. D: module-suspend-on-idle.c: Source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 becomes busy. D: resampler.c: Channel matrix: D: resampler.c: I00 I01 D: resampler.c: +------------ D: resampler.c: O00 | 1.000 1.000 I: resampler.c: Using resampler 'src-linear' I: resampler.c: Using float32le as working format. D: memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0 D: memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0 I: source-output.c: Created output 0 "ALSA Capture" on alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 with sample spec s16le 1ch 8000Hz and channel map mono D: memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=2, prebuf=1, minreq=0 maxrewind=0 D: memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=2, prebuf=2, minreq=2 maxrewind=0 I: protocol-native.c: Final latency 40.00 ms = 20.00 ms + 20.00 ms D: module-alsa-source.c: hwbuf_unused=0 D: module-alsa-source.c: setting avail_min=1 D: module-alsa-source.c: hwbuf_unused=0 D: module-alsa-source.c: setting avail_min=1 I: client.c: Created 3 "Native client (UNIX socket client)" D: protocol-native.c: Protocol version: remote 14, local 14 I: protocol-native.c: Got credentials: uid=1000 gid=1000 success=1 D: protocol-native.c: SHM possible: yes D: protocol-native.c: Negotiated SHM: yes I: module-stream-restore.c: Restoring device for stream sink-input-by-application-name:ALSA plug-in [duplex]. I: module-stream-restore.c: Restoring volume for sink input sink-input-by-application-name:ALSA plug-in [duplex]. D: module-stream-restore.c: Not restoring mute state for sink input sink-input-by-application-name:ALSA plug-in [duplex], because already set. D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 becomes busy. D: resampler.c: Channel matrix: D: resampler.c: I00 D: resampler.c: +------ D: resampler.c: O00 | 1.000 D: resampler.c: O01 | 1.000 I: resampler.c: Using resampler 'src-linear' I: resampler.c: Using float32le as working format. D: memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0 D: memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0 I: sink-input.c: Created input 1 "ALSA Playback" on alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 with sample spec s16le 1ch 8000Hz and channel map mono I: protocol-native.c: Requested tlength=80.00 ms, minreq=20.00 ms D: protocol-native.c: Early requests mode enabled, configuring sink latency to minreq. D: memblockq.c: memblockq requested: maxlength=4194304, tlength=1280, base=2, prebuf=960, minreq=320 maxrewind=0 D: memblockq.c: memblockq sanitized: maxlength=4194304, tlength=1280, base=2, prebuf=960, minreq=320 maxrewind=0 I: protocol-native.c: Final latency 100.00 ms = 40.00 ms + 2*20.00 ms + 20.00 ms D: module-alsa-sink.c: hwbuf_unused=0 D: module-alsa-sink.c: setting avail_min=1 D: module-alsa-source.c: hwbuf_unused=0 D: module-alsa-source.c: setting avail_min=1 D: module-suspend-on-idle.c: Source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 becomes idle. D: module-suspend-on-idle.c: Source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 becomes idle. I: source-output.c: Freeing output 0 "ALSA Capture" I: module-stream-restore.c: Restoring device for stream source-output-by-application-name:ALSA plug-in [duplex]. D: module-suspend-on-idle.c: Source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 becomes busy. D: resampler.c: Channel matrix: D: resampler.c: I00 I01 D: resampler.c: +------------ D: resampler.c: O00 | 1.000 1.000 I: resampler.c: Using resampler 'src-linear' I: resampler.c: Using float32le as working format. D: memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0 D: memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0 I: source-output.c: Created output 1 "ALSA Capture" on alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 with sample spec s16le 1ch 8000Hz and channel map mono D: memblockq.c: memblockq requested: maxlength=4194304, tlength=0, base=2, prebuf=1, minreq=0 maxrewind=0 D: memblockq.c: memblockq sanitized: maxlength=4194304, tlength=4194304, base=2, prebuf=2, minreq=2 maxrewind=0 I: protocol-native.c: Final latency 40.00 ms = 20.00 ms + 20.00 ms D: module-alsa-source.c: hwbuf_unused=0 D: module-alsa-source.c: setting avail_min=1 D: module-alsa-source.c: hwbuf_unused=0 D: module-alsa-source.c: setting avail_min=1 D: module-alsa-sink.c: hwbuf_unused=0 D: module-alsa-sink.c: setting avail_min=1 D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 becomes idle. D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 becomes idle. I: sink-input.c: Freeing input 1 "ALSA Playback" I: module-stream-restore.c: Restoring device for stream sink-input-by-application-name:ALSA plug-in [duplex]. I: module-stream-restore.c: Restoring volume for sink input sink-input-by-application-name:ALSA plug-in [duplex]. D: module-stream-restore.c: Not restoring mute state for sink input sink-input-by-application-name:ALSA plug-in [duplex], because already set. D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 becomes busy. D: resampler.c: Channel matrix: D: resampler.c: I00 D: resampler.c: +------ D: resampler.c: O00 | 1.000 D: resampler.c: O01 | 1.000 I: resampler.c: Using resampler 'src-linear' I: resampler.c: Using float32le as working format. D: memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0 D: memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0 I: sink-input.c: Created input 2 "ALSA Playback" on alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 with sample spec s16le 1ch 8000Hz and channel map mono I: protocol-native.c: Requested tlength=80.00 ms, minreq=20.00 ms D: protocol-native.c: Early requests mode enabled, configuring sink latency to minreq. D: memblockq.c: memblockq requested: maxlength=4194304, tlength=1280, base=2, prebuf=960, minreq=320 maxrewind=0 D: memblockq.c: memblockq sanitized: maxlength=4194304, tlength=1280, base=2, prebuf=960, minreq=320 maxrewind=0 I: protocol-native.c: Final latency 100.00 ms = 40.00 ms + 2*20.00 ms + 20.00 ms D: module-alsa-sink.c: hwbuf_unused=0 D: module-alsa-sink.c: setting avail_min=1 D: protocol-native.c: Requesting rewind due to end of underrun. D: module-alsa-sink.c: hwbuf_unused=0 D: module-alsa-sink.c: setting avail_min=1 D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 becomes idle. D: module-suspend-on-idle.c: Sink alsa_output.pci_1102_4_sound_card_0_alsa_playback_0 becomes idle. I: sink-input.c: Freeing input 2 "ALSA Playback" I: client.c: Freed 3 "ALSA plug-in [duplex]" I: protocol-native.c: Connection died. D: module-alsa-source.c: hwbuf_unused=0 D: module-alsa-source.c: setting avail_min=1 D: module-suspend-on-idle.c: Source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 becomes idle. D: module-suspend-on-idle.c: Source alsa_input.pci_1102_4_sound_card_0_alsa_capture_0 becomes idle. I: source-output.c: Freeing output 1 "ALSA Capture" I: client.c: Freed 2 "ALSA plug-in [duplex]" I: protocol-native.c: Connection died. ------------------------------------------------------------------------------
2010/4/21 Stefan Schoenleitner dev.c0debabe@gmail.com
Hi,
I wrote a small application that opens both streams (i.e. capture and playback) of a PCM in non blocking (i.e. SND_PCM_NONBLOCK) mode.
During device configuration I also set up the minimum number of frames to consider a PCM ready:
// when to consider the device to be ready for the next data transfer operation if ((ret=snd_pcm_sw_params_set_avail_min(handle, swparams, pcm_period_size))<0) { fprintf(stderr, "could not set avail_min: %s\n", snd_strerror(ret)); return NULL; }
As you can see it is set to the number of frames one period contains.
In other words, each time there are at least that many frames available for reading on the capture device, we should get a POLLIN event when using poll(). On the other side we should get a POLLOUT event if that many frames can be written to the playback device.
Then, later in my application, I use poll() on the fds of the PCM devices. As, according to the ALSA reference, poll events can be "mangeled", I use snd_pcm_poll_descriptors_revents() to "demangle" the events:
if ((ret=snd_pcm_poll_descriptors_revents(device_handle, &poll_fds[i], 1, &revents))<0) { fprintf(stderr, "could not snd_pcm_poll_descriptors_revents: %s\n", snd_strerror(ret)); exit(EXIT_FAILURE); }
After that, each time there was a POLLIN event, I am reading exactly one period from the capture PCM. On the other side, if there was a POLLOUT event, I am writing exactly one period to the playback PCM.
While this works for a small number of periods (which is not always the same), I always end up with a "Resource temporarily unavailable" error when trying to read one period from the capture PCM:
demangled poll: POLLIN on capture device could not read from capture device: Resource temporarily unavailable
Since a POLLIN event only occurs after at least a full period is available for reading (as set up by snd_pcm_sw_params_set_avail_min() above) and I only read after a POLLIN event occured on the capture device fd, I really do not understand why I get the above error.
So why does the above happen ? What am I doing wrong ?
If it helps, I pasted the source code here so that you can view it nicely: http://pastebin.com/fCicqctq
cheers, stefan
which sound card are you using ?
The program did not run on my two sound card since you are using 160 frames and mono
are you using "pulse" device for testing since only pulse device return the error messages "could not read from capture device: Resource temporarily unavailable" ?
Raymond Yau wrote:
2010/4/21 Stefan Schoenleitner dev.c0debabe@gmail.com
If it helps, I pasted the source code here so that you can view it nicely: http://pastebin.com/fCicqctq
which sound card are you using ?
The program did not run on my two sound card since you are using 160 frames and mono
are you using "pulse" device for testing since only pulse device return the error messages "could not read from capture device: Resource temporarily unavailable" ?
Yes, I am using pulseaudio since my hardware sound card (an audigy 2) does not support the period size I'm using.
The error message above is is from my own program (line 444):
----------------------------------------------------------------- if ((ret=snd_pcm_readi(capture_device, buffer, 160))<0) { if (ret == -EPIPE) { fprintf(stderr, "XRUN while reading from capture device: %s\n", snd_strerror(ret)); snd_pcm_prepare(capture_device); snd_pcm_start(capture_device); } else { fprintf(stderr, "could not read from capture device: %s\n", snd_strerror(ret)); exit(EXIT_FAILURE); } } -----------------------------------------------------------------
cheers, stefan
participants (3)
-
Jaroslav Kysela -
Raymond Yau -
Stefan Schoenleitner