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/*
* This file is part of mpv.
*
* mpv is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* mpv is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with mpv. If not, see <http://www.gnu.org/licenses/>.
*/
#include <stdio.h>
#include <stdlib.h>
#include <sys/time.h>
#include "config.h"
#include <alsa/asoundlib.h>
#include "audio_in.h"
#include "common/msg.h"
int ai_alsa_setup(audio_in_t *ai)
{
snd_pcm_hw_params_t *params;
snd_pcm_sw_params_t *swparams;
snd_pcm_uframes_t buffer_size, period_size;
int err;
int dir;
unsigned int rate;
snd_pcm_hw_params_alloca(¶ms);
snd_pcm_sw_params_alloca(&swparams);
err = snd_pcm_hw_params_any(ai->alsa.handle, params);
if (err < 0) {
MP_ERR(ai, "Broken configuration for this PCM: no configurations available.\n");
return -1;
}
err = snd_pcm_hw_params_set_access(ai->alsa.handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0) {
MP_ERR(ai, "Access type not available.\n");
return -1;
}
err = snd_pcm_hw_params_set_format(ai->alsa.handle, params, SND_PCM_FORMAT_S16);
if (err < 0) {
MP_ERR(ai, "Sample format not available.\n");
return -1;
}
err = snd_pcm_hw_params_set_channels(ai->alsa.handle, params, ai->req_channels);
if (err < 0) {
snd_pcm_hw_params_get_channels(params, &ai->channels);
MP_ERR(ai, "Channel count not available - reverting to default: %d\n",
ai->channels);
} else {
ai->channels = ai->req_channels;
}
dir = 0;
rate = ai->req_samplerate;
err = snd_pcm_hw_params_set_rate_near(ai->alsa.handle, params, &rate, &dir);
if (err < 0) {
MP_ERR(ai, "Cannot set samplerate.\n");
}
ai->samplerate = rate;
dir = 0;
ai->alsa.buffer_time = 1000000;
err = snd_pcm_hw_params_set_buffer_time_near(ai->alsa.handle, params,
&ai->alsa.buffer_time, &dir);
if (err < 0) {
MP_ERR(ai, "Cannot set buffer time.\n");
}
dir = 0;
ai->alsa.period_time = ai->alsa.buffer_time / 4;
err = snd_pcm_hw_params_set_period_time_near(ai->alsa.handle, params,
&ai->alsa.period_time, &dir);
if (err < 0) {
MP_ERR(ai, "Cannot set period time.\n");
}
err = snd_pcm_hw_params(ai->alsa.handle, params);
if (err < 0) {
MP_ERR(ai, "Unable to install hardware parameters: %s", snd_strerror(err));
snd_pcm_hw_params_dump(params, ai->alsa.log);
return -1;
}
dir = -1;
snd_pcm_hw_params_get_period_size(params, &period_size, &dir);
snd_pcm_hw_params_get_buffer_size(params, &buffer_size);
ai->alsa.chunk_size = period_size;
if (period_size == buffer_size) {
MP_ERR(ai, "Can't use period equal to buffer size (%u == %lu)\n", ai->alsa.chunk_size, (long)buffer_size);
return -1;
}
snd_pcm_sw_params_current(ai->alsa.handle, swparams);
err = snd_pcm_sw_params_set_avail_min(ai->alsa.handle, swparams, ai->alsa.chunk_size);
err = snd_pcm_sw_params_set_start_threshold(ai->alsa.handle, swparams, 0);
err = snd_pcm_sw_params_set_stop_threshold(ai->alsa.handle, swparams, buffer_size);
if (snd_pcm_sw_params(ai->alsa.handle, swparams) < 0) {
MP_ERR(ai, "Unable to install software parameters:\n");
snd_pcm_sw_params_dump(swparams, ai->alsa.log);
return -1;
}
if (mp_msg_test(ai->log, MSGL_V)) {
snd_pcm_dump(ai->alsa.handle, ai->alsa.log);
}
ai->alsa.bits_per_sample = snd_pcm_format_physical_width(SND_PCM_FORMAT_S16);
ai->alsa.bits_per_frame = ai->alsa.bits_per_sample * ai->channels;
ai->blocksize = ai->alsa.chunk_size * ai->alsa.bits_per_frame / 8;
ai->samplesize = ai->alsa.bits_per_sample;
ai->bytes_per_sample = ai->alsa.bits_per_sample/8;
return 0;
}
int ai_alsa_init(audio_in_t *ai)
{
int err;
const char *device = ai->alsa.device;
if (!device)
device = "default";
err = snd_pcm_open(&ai->alsa.handle, device, SND_PCM_STREAM_CAPTURE, 0);
if (err < 0) {
MP_ERR(ai, "Error opening audio: %s\n", snd_strerror(err));
return -1;
}
err = snd_output_stdio_attach(&ai->alsa.log, stderr, 0);
if (err < 0) {
return -1;
}
err = ai_alsa_setup(ai);
return err;
}
#ifndef timersub
#define timersub(a, b, result) \
do { \
(result)->tv_sec = (a)->tv_sec - (b)->tv_sec; \
(result)->tv_usec = (a)->tv_usec - (b)->tv_usec; \
if ((result)->tv_usec < 0) { \
--(result)->tv_sec; \
(result)->tv_usec += 1000000; \
} \
} while (0)
#endif
int ai_alsa_xrun(audio_in_t *ai)
{
snd_pcm_status_t *status;
int res;
snd_pcm_status_alloca(&status);
if ((res = snd_pcm_status(ai->alsa.handle, status))<0) {
MP_ERR(ai, "ALSA status error: %s", snd_strerror(res));
return -1;
}
if (snd_pcm_status_get_state(status) == SND_PCM_STATE_XRUN) {
struct timeval now, diff, tstamp;
gettimeofday(&now, 0);
snd_pcm_status_get_trigger_tstamp(status, &tstamp);
timersub(&now, &tstamp, &diff);
MP_ERR(ai, "ALSA xrun!!! (at least %.3f ms long)\n",
diff.tv_sec * 1000 + diff.tv_usec / 1000.0);
if (mp_msg_test(ai->log, MSGL_V)) {
MP_ERR(ai, "ALSA Status:\n");
snd_pcm_status_dump(status, ai->alsa.log);
}
if ((res = snd_pcm_prepare(ai->alsa.handle))<0) {
MP_ERR(ai, "ALSA xrun: prepare error: %s", snd_strerror(res));
return -1;
}
return 0; /* ok, data should be accepted again */
}
MP_ERR(ai, "ALSA read/write error");
return -1;
}
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