* HDA-based STAC9200 soundcard loses interrupts
@ 2007-02-20 9:42 Jean-Marc Valin
0 siblings, 0 replies; only message in thread
From: Jean-Marc Valin @ 2007-02-20 9:42 UTC (permalink / raw)
To: alsa-devel
[-- Attachment #1: Type: text/plain, Size: 1297 bytes --]
Hi,
I've got a laptop that has skipping problems, mainly full-duplex
programs. This happens for both native ALSA applications and those using
OSS. Digging a bit into that, I realised that it mainly happens when
using two periods. I wrote a test program (attached) to dump the time I
get the "interrupts" (from user space). The output is attached in the
"timings" file (gettimeofday() microseconds). Basically, the time
increases with near perfect regularity, except for a few cases, where
there's a skip of exactly one period. Looks a lot like lost interrupts
or something like that (at least that was James Courtier-Dutton's
interpretation when he looked at it during LCA2007).
This data was obtained on a Dell D820 laptop with a STAC9200-based
soundcard running kernel 2.6.20 and ALSA 1.0.14rc2, but I've observed
the same problem with all other versions of kernel/ALSA I've tested.
I've also seen that problem on all other HDA-based desktops I have, but
not on any of the i810-based machines I have.
The problem usually does not show up with playback-only applications
(probably because they use >2 periods or something), but it makes VoIP
apps like Ekiga (native ALSA) and OpenWengo (OSS) totally unusable as I
get several xruns per second.
Does anyone know of a fix for that?
Jean-Marc
[-- Attachment #2: timings --]
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[-- Attachment #3: duplex.c --]
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/*
Copyright (C) 2004-2006 Jean-Marc Valin
Copyright (C) 2006 Commonwealth Scientific and Industrial Research
Organisation (CSIRO) Australia
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions are
met:
1. Redistributions of source code must retain the above copyright notice,
this list of conditions and the following disclaimer.
2. Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
3. The name of the author may not be used to endorse or promote products
derived from this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
DISCLAIMED. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT,
INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
(INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN
ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
POSSIBILITY OF SUCH DAMAGE.
*/
#include "alsa_device.h"
#include <stdlib.h>
#include <alsa/asoundlib.h>
#include <sched.h>
#include <sys/time.h>
#include <time.h>
struct AlsaDevice_ {
char *device_name;
int channels;
int period;
snd_pcm_t *capture_handle;
snd_pcm_t *playback_handle;
int readN, writeN;
struct pollfd *read_fd, *write_fd;
};
AlsaDevice *alsa_device_open(char *device_name, unsigned int rate, int channels, int period)
{
int dir;
int err;
snd_pcm_hw_params_t *hw_params;
snd_pcm_sw_params_t *sw_params;
snd_pcm_uframes_t period_size = period;
snd_pcm_uframes_t buffer_size = 4*period;
static snd_output_t *jcd_out;
AlsaDevice *dev = malloc(sizeof(*dev));
if (!dev)
return NULL;
dev->device_name = malloc(1+strlen(device_name));
if (!dev->device_name)
{
free(dev);
return NULL;
}
strcpy(dev->device_name, device_name);
dev->channels = channels;
dev->period = period;
err = snd_output_stdio_attach(&jcd_out, stdout, 0);
if ((err = snd_pcm_open (&dev->capture_handle, dev->device_name, SND_PCM_STREAM_CAPTURE, 0)) < 0) {
fprintf (stderr, "cannot open audio device %s (%s)\n",
dev->device_name,
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) {
fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n",
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_hw_params_any (dev->capture_handle, hw_params)) < 0) {
fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n",
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_hw_params_set_access (dev->capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
fprintf (stderr, "cannot set access type (%s)\n",
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_hw_params_set_format (dev->capture_handle, hw_params, SND_PCM_FORMAT_S16_LE)) < 0) {
fprintf (stderr, "cannot set sample format (%s)\n",
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_hw_params_set_rate_near (dev->capture_handle, hw_params, &rate, 0)) < 0) {
fprintf (stderr, "cannot set sample rate (%s)\n",
snd_strerror (err));
assert(0);
}
/*fprintf (stderr, "rate = %d\n", rate);*/
if ((err = snd_pcm_hw_params_set_channels (dev->capture_handle, hw_params, channels)) < 0) {
fprintf (stderr, "cannot set channel count (%s)\n",
snd_strerror (err));
assert(0);
}
period_size = period;
dir = 0;
if ((err = snd_pcm_hw_params_set_period_size_near (dev->capture_handle, hw_params, &period_size, &dir)) < 0) {
fprintf (stderr, "cannot set period size (%s)\n",
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_hw_params_set_periods (dev->capture_handle, hw_params, 4, 0)) < 0) {
fprintf (stderr, "cannot set number of periods (%s)\n",
snd_strerror (err));
assert(0);
}
buffer_size = period_size * 4;
dir=0;
if ((err = snd_pcm_hw_params_set_buffer_size_near (dev->capture_handle, hw_params, &buffer_size)) < 0) {
fprintf (stderr, "cannot set buffer time (%s)\n",
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_hw_params (dev->capture_handle, hw_params)) < 0) {
fprintf (stderr, "cannot set capture parameters (%s)\n",
snd_strerror (err));
assert(0);
}
snd_pcm_dump_setup(dev->capture_handle, jcd_out);
snd_pcm_hw_params_free (hw_params);
if ((err = snd_pcm_sw_params_malloc (&sw_params)) < 0) {
fprintf (stderr, "cannot allocate software parameters structure (%s)\n",
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_sw_params_current (dev->capture_handle, sw_params)) < 0) {
fprintf (stderr, "cannot initialize software parameters structure (%s)\n",
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_sw_params_set_avail_min (dev->capture_handle, sw_params, period)) < 0) {
fprintf (stderr, "cannot set minimum available count (%s)\n",
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_sw_params (dev->capture_handle, sw_params)) < 0) {
fprintf (stderr, "cannot set software parameters (%s)\n",
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_open (&dev->playback_handle, dev->device_name, SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
fprintf (stderr, "cannot open audio device %s (%s)\n",
dev->device_name,
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) {
fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n",
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_hw_params_any (dev->playback_handle, hw_params)) < 0) {
fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n",
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_hw_params_set_access (dev->playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
fprintf (stderr, "cannot set access type (%s)\n",
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_hw_params_set_format (dev->playback_handle, hw_params, SND_PCM_FORMAT_S16_LE)) < 0) {
fprintf (stderr, "cannot set sample format (%s)\n",
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_hw_params_set_rate_near (dev->playback_handle, hw_params, &rate, 0)) < 0) {
fprintf (stderr, "cannot set sample rate (%s)\n",
snd_strerror (err));
assert(0);
}
/*fprintf (stderr, "rate = %d\n", rate);*/
if ((err = snd_pcm_hw_params_set_channels (dev->playback_handle, hw_params, channels)) < 0) {
fprintf (stderr, "cannot set channel count (%s)\n",
snd_strerror (err));
assert(0);
}
period_size = period;
dir = 0;
if ((err = snd_pcm_hw_params_set_period_size_near (dev->playback_handle, hw_params, &period_size, &dir)) < 0) {
fprintf (stderr, "cannot set period size (%s)\n",
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_hw_params_set_periods (dev->playback_handle, hw_params, 4, 0)) < 0) {
fprintf (stderr, "cannot set number of periods (%s)\n",
snd_strerror (err));
assert(0);
}
buffer_size = period_size * 4;
dir=0;
if ((err = snd_pcm_hw_params_set_buffer_size_near (dev->playback_handle, hw_params, &buffer_size)) < 0) {
fprintf (stderr, "cannot set buffer time (%s)\n",
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_hw_params (dev->playback_handle, hw_params)) < 0) {
fprintf (stderr, "cannot set playback parameters (%s)\n",
snd_strerror (err));
assert(0);
}
snd_pcm_dump_setup(dev->playback_handle, jcd_out);
snd_pcm_hw_params_free (hw_params);
if ((err = snd_pcm_sw_params_malloc (&sw_params)) < 0) {
fprintf (stderr, "cannot allocate software parameters structure (%s)\n",
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_sw_params_current (dev->playback_handle, sw_params)) < 0) {
fprintf (stderr, "cannot initialize software parameters structure (%s)\n",
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_sw_params_set_avail_min (dev->playback_handle, sw_params, period)) < 0) {
fprintf (stderr, "cannot set minimum available count (%s)\n",
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_sw_params_set_start_threshold (dev->playback_handle, sw_params, period)) < 0) {
fprintf (stderr, "cannot set start mode (%s)\n",
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_sw_params (dev->playback_handle, sw_params)) < 0) {
fprintf (stderr, "cannot set software parameters (%s)\n",
snd_strerror (err));
assert(0);
}
snd_pcm_link(dev->capture_handle, dev->playback_handle);
if ((err = snd_pcm_prepare (dev->capture_handle)) < 0) {
fprintf (stderr, "cannot prepare audio interface for use (%s)\n",
snd_strerror (err));
assert(0);
}
if ((err = snd_pcm_prepare (dev->playback_handle)) < 0) {
fprintf (stderr, "cannot prepare audio interface for use (%s)\n",
snd_strerror (err));
assert(0);
}
dev->readN = snd_pcm_poll_descriptors_count(dev->capture_handle);
dev->writeN = snd_pcm_poll_descriptors_count(dev->playback_handle);
dev->read_fd = malloc(dev->readN*sizeof(*dev->read_fd));
/*printf ("descriptors: %d %d\n", dev->readN, dev->writeN);*/
if (snd_pcm_poll_descriptors(dev->capture_handle, dev->read_fd, dev->readN) != dev->readN)
{
fprintf (stderr, "cannot obtain capture file descriptors (%s)\n",
snd_strerror (err));
assert(0);
}
dev->write_fd = malloc(dev->writeN*sizeof(*dev->read_fd));
if (snd_pcm_poll_descriptors(dev->playback_handle, dev->write_fd, dev->writeN) != dev->writeN)
{
fprintf (stderr, "cannot obtain playback file descriptors (%s)\n",
snd_strerror (err));
assert(0);
}
return dev;
}
void alsa_device_close(AlsaDevice *dev)
{
snd_pcm_close(dev->capture_handle);
snd_pcm_close(dev->playback_handle);
free(dev->device_name);
free(dev);
}
int alsa_device_read(AlsaDevice *dev, short *pcm, int len)
{
int err;
/*fprintf (stderr, "-");*/
if ((err = snd_pcm_readi (dev->capture_handle, pcm, len)) != len)
{
if (err<0)
{
//fprintf(stderr, "error %d, EPIPE = %d\n", err, EPIPE);
if (err == -EPIPE)
{
fprintf (stderr, "An overrun has occured, reseting capture\n");
} else
{
fprintf (stderr, "read from audio interface failed (%s)\n",
snd_strerror (err));
}
if ((err = snd_pcm_prepare (dev->capture_handle)) < 0)
{
fprintf (stderr, "cannot prepare audio interface for use (%s)\n",
snd_strerror (err));
}
if ((err = snd_pcm_start (dev->capture_handle)) < 0)
{
fprintf (stderr, "cannot prepare audio interface for use (%s)\n",
snd_strerror (err));
}
/*alsa_device_read(dev,pcm,len);*/
} else {
fprintf (stderr, "Couldn't read as many samples as I wanted (%d instead of %d)\n", err, len);
}
return 1;
}
//fprintf (stderr, "-");
return 0;
}
int alsa_device_write(AlsaDevice *dev, const short *pcm, int len)
{
int err;
if ((err = snd_pcm_writei (dev->playback_handle, pcm, len)) != len)
{
if (err<0)
{
if (err == -EPIPE)
{
fprintf (stderr, "An underrun has occured, reseting playback, len=%d\n", len);
} else
{
fprintf (stderr, "write to audio interface failed (%s)\n",
snd_strerror (err));
}
if ((err = snd_pcm_prepare (dev->playback_handle)) < 0)
{
fprintf (stderr, "cannot prepare audio interface for use (%s)\n",
snd_strerror (err));
}
} else {
fprintf (stderr, "Couldn't write as many samples as I wanted (%d instead of %d)\n", err, len);
}
/*alsa_device_write(dev,pcm,len);*/
return 1;
}
//fprintf (stderr, "+%d ", err);
return 0;
}
int alsa_device_capture_ready(AlsaDevice *dev, struct pollfd *pfds, unsigned int nfds)
{
unsigned short revents=0;
int err;
if ((err = snd_pcm_poll_descriptors_revents(dev->capture_handle, pfds, dev->readN, &revents)) < 0)
{
//cerr << "error in snd_pcm_poll_descriptors_revents for capture: " << snd_strerror (err) << endl;
//FIXME: This is a kludge
fprintf (stderr, "error in alsa_device_capture_ready: %s\n", snd_strerror (err));
return pfds[0].revents & POLLIN;
}
//cerr << (revents & POLLERR) << endl;
return revents & POLLIN;
}
int alsa_device_playback_ready(AlsaDevice *dev, struct pollfd *pfds, unsigned int nfds)
{
unsigned short revents=0;
int err;
if ((err = snd_pcm_poll_descriptors_revents(dev->playback_handle, pfds+dev->readN, dev->writeN, &revents)) < 0)
{
//cerr << "error in snd_pcm_poll_descriptors_revents for playback: " << snd_strerror (err) << endl;
//FIXME: This is a kludge
fprintf (stderr, "error in alsa_device_playback_ready: %s\n", snd_strerror (err));
return pfds[1].revents & POLLOUT;
}
//cerr << (revents & POLLERR) << endl;
return revents & POLLOUT;
}
void alsa_device_start(AlsaDevice *dev)
{
int i;
short pcm[dev->period*dev->channels];
for (i=0;i<dev->period*dev->channels;i++)
pcm[i] = 0;
alsa_device_write(dev, pcm, dev->period);
alsa_device_write(dev, pcm, dev->period);
snd_pcm_start(dev->capture_handle);
snd_pcm_start(dev->playback_handle);
}
int alsa_device_nfds(AlsaDevice *dev)
{
return dev->writeN+dev->readN;
}
void alsa_device_getfds(AlsaDevice *dev, struct pollfd *pfds, unsigned int nfds)
{
int i;
assert (nfds >= dev->writeN+dev->readN);
for (i=0;i<dev->readN;i++)
pfds[i] = dev->read_fd[i];
for (i=0;i<dev->writeN;i++)
pfds[i+dev->readN] = dev->write_fd[i];
}
#define SAMPLING_RATE 48000
#define FRAME_SIZE 1024
int main(int argc, char **argv)
{
int i;
int nfds;
struct pollfd *pfds;
AlsaDevice *audio_dev;
struct timeval tv;
struct sched_param param;
param.sched_priority = sched_get_priority_max(SCHED_FIFO);
sched_setscheduler(0,SCHED_FIFO,¶m);
perror("sched_setscheduler");
audio_dev = alsa_device_open(argv[1], SAMPLING_RATE, 1, FRAME_SIZE);
/* Setup all file descriptors for poll()ing */
nfds = alsa_device_nfds(audio_dev);
pfds = malloc(sizeof(*pfds)*(nfds));
alsa_device_getfds(audio_dev, pfds, nfds);
alsa_device_start(audio_dev);
while (1)
{
/* Wait for either 1) capture 2) playback 3) socket data */
poll(pfds, nfds, -1);
if (alsa_device_playback_ready(audio_dev, pfds, nfds))
{
short pcm[FRAME_SIZE];
for (i=0;i<FRAME_SIZE;i++)
pcm[i] = 0;
/* Playback the audio and reset the echo canceller if we got an underru
n */
gettimeofday(&tv, NULL);
fprintf (stderr, "play %d\n", tv.tv_usec);
alsa_device_write(audio_dev, pcm, FRAME_SIZE);
}
/* Audio available from the soundcard (capture) */
if (alsa_device_capture_ready(audio_dev, pfds, nfds))
{
short pcm[FRAME_SIZE];
/* Get audio from the soundcard */
gettimeofday(&tv, NULL);
fprintf (stderr, "rec %d\n", tv.tv_usec);
alsa_device_read(audio_dev, pcm, FRAME_SIZE);
}
}
}
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2007-02-20 9:42 HDA-based STAC9200 soundcard loses interrupts Jean-Marc Valin
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