mirror of https://github.com/mpv-player/mpv
377 lines
9.3 KiB
C
377 lines
9.3 KiB
C
/*
|
|
ao_alsa5 - ALSA-0.5.x output plugin for MPlayer
|
|
|
|
(C) Alex Beregszaszi <alex@naxine.org>
|
|
|
|
Thanks to Arpi for helping me ;)
|
|
*/
|
|
|
|
#include <errno.h>
|
|
#include <sys/asoundlib.h>
|
|
|
|
#include "../config.h"
|
|
|
|
#include "audio_out.h"
|
|
#include "audio_out_internal.h"
|
|
#include "afmt.h"
|
|
|
|
#include "../mp_msg.h"
|
|
|
|
extern int verbose;
|
|
|
|
static ao_info_t info =
|
|
{
|
|
"ALSA-0.5.x audio output",
|
|
"alsa5",
|
|
"Alex Beregszaszi <alex@naxine.org>",
|
|
""
|
|
};
|
|
|
|
LIBAO_EXTERN(alsa5)
|
|
|
|
static snd_pcm_t *alsa_handler;
|
|
static snd_pcm_format_t alsa_format;
|
|
static int alsa_rate = SND_PCM_RATE_CONTINUOUS;
|
|
|
|
/* to set/get/query special features/parameters */
|
|
static int control(int cmd, int arg)
|
|
{
|
|
return(CONTROL_UNKNOWN);
|
|
}
|
|
|
|
/*
|
|
open & setup audio device
|
|
return: 1=success 0=fail
|
|
*/
|
|
static int init(int rate_hz, int channels, int format, int flags)
|
|
{
|
|
int err;
|
|
int cards = -1;
|
|
snd_pcm_channel_params_t params;
|
|
snd_pcm_channel_setup_t setup;
|
|
snd_pcm_info_t info;
|
|
snd_pcm_channel_info_t chninfo;
|
|
|
|
mp_msg(MSGT_AO, MSGL_INFO, "alsa-init: requested format: %d Hz, %d channels, %s\n", rate_hz,
|
|
channels, audio_out_format_name(format));
|
|
|
|
alsa_handler = NULL;
|
|
|
|
mp_msg(MSGT_AO, MSGL_V, "alsa-init: compiled for ALSA-%s (%d)\n", SND_LIB_VERSION_STR,
|
|
SND_LIB_VERSION);
|
|
|
|
if ((cards = snd_cards()) < 0)
|
|
{
|
|
mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: no soundcards found\n");
|
|
return(0);
|
|
}
|
|
|
|
ao_data.format = format;
|
|
ao_data.channels = channels;
|
|
ao_data.samplerate = rate_hz;
|
|
ao_data.bps = ao_data.samplerate*ao_data.channels;
|
|
ao_data.outburst = OUTBURST;
|
|
ao_data.buffersize = 16384;
|
|
|
|
memset(&alsa_format, 0, sizeof(alsa_format));
|
|
switch (format)
|
|
{
|
|
case AFMT_S8:
|
|
alsa_format.format = SND_PCM_SFMT_S8;
|
|
break;
|
|
case AFMT_U8:
|
|
alsa_format.format = SND_PCM_SFMT_U8;
|
|
break;
|
|
case AFMT_U16_LE:
|
|
alsa_format.format = SND_PCM_SFMT_U16_LE;
|
|
break;
|
|
case AFMT_U16_BE:
|
|
alsa_format.format = SND_PCM_SFMT_U16_BE;
|
|
break;
|
|
#ifndef WORDS_BIGENDIAN
|
|
case AFMT_AC3:
|
|
#endif
|
|
case AFMT_S16_LE:
|
|
alsa_format.format = SND_PCM_SFMT_S16_LE;
|
|
break;
|
|
#ifdef WORDS_BIGENDIAN
|
|
case AFMT_AC3:
|
|
#endif
|
|
case AFMT_S16_BE:
|
|
alsa_format.format = SND_PCM_SFMT_S16_BE;
|
|
break;
|
|
default:
|
|
alsa_format.format = SND_PCM_SFMT_MPEG;
|
|
break;
|
|
}
|
|
|
|
switch(alsa_format.format)
|
|
{
|
|
case SND_PCM_SFMT_S16_LE:
|
|
case SND_PCM_SFMT_U16_LE:
|
|
ao_data.bps *= 2;
|
|
break;
|
|
case -1:
|
|
mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: invalid format (%s) requested - output disabled\n",
|
|
audio_out_format_name(format));
|
|
return(0);
|
|
default:
|
|
break;
|
|
}
|
|
|
|
switch(rate_hz)
|
|
{
|
|
case 8000:
|
|
alsa_rate = SND_PCM_RATE_8000;
|
|
break;
|
|
case 11025:
|
|
alsa_rate = SND_PCM_RATE_11025;
|
|
break;
|
|
case 16000:
|
|
alsa_rate = SND_PCM_RATE_16000;
|
|
break;
|
|
case 22050:
|
|
alsa_rate = SND_PCM_RATE_22050;
|
|
break;
|
|
case 32000:
|
|
alsa_rate = SND_PCM_RATE_32000;
|
|
break;
|
|
case 44100:
|
|
alsa_rate = SND_PCM_RATE_44100;
|
|
break;
|
|
case 48000:
|
|
alsa_rate = SND_PCM_RATE_48000;
|
|
break;
|
|
case 88200:
|
|
alsa_rate = SND_PCM_RATE_88200;
|
|
break;
|
|
case 96000:
|
|
alsa_rate = SND_PCM_RATE_96000;
|
|
break;
|
|
case 176400:
|
|
alsa_rate = SND_PCM_RATE_176400;
|
|
break;
|
|
case 192000:
|
|
alsa_rate = SND_PCM_RATE_192000;
|
|
break;
|
|
default:
|
|
alsa_rate = SND_PCM_RATE_CONTINUOUS;
|
|
break;
|
|
}
|
|
|
|
alsa_format.rate = ao_data.samplerate;
|
|
alsa_format.voices = ao_data.channels*2;
|
|
alsa_format.interleave = 1;
|
|
|
|
if ((err = snd_pcm_open(&alsa_handler, 0, 0, SND_PCM_OPEN_PLAYBACK)) < 0)
|
|
{
|
|
mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: playback open error: %s\n", snd_strerror(err));
|
|
return(0);
|
|
}
|
|
|
|
if ((err = snd_pcm_info(alsa_handler, &info)) < 0)
|
|
{
|
|
mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: pcm info error: %s\n", snd_strerror(err));
|
|
return(0);
|
|
}
|
|
|
|
mp_msg(MSGT_AO, MSGL_INFO, "alsa-init: %d soundcard(s) found, using: %s\n",
|
|
cards, info.name);
|
|
|
|
if (info.flags & SND_PCM_INFO_PLAYBACK)
|
|
{
|
|
memset(&chninfo, 0, sizeof(chninfo));
|
|
chninfo.channel = SND_PCM_CHANNEL_PLAYBACK;
|
|
if ((err = snd_pcm_channel_info(alsa_handler, &chninfo)) < 0)
|
|
{
|
|
mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: pcm channel info error: %s\n", snd_strerror(err));
|
|
return(0);
|
|
}
|
|
|
|
#ifndef __QNX__
|
|
if (chninfo.buffer_size)
|
|
ao_data.buffersize = chninfo.buffer_size;
|
|
#endif
|
|
|
|
mp_msg(MSGT_AO, MSGL_V, "alsa-init: setting preferred buffer size from driver: %d bytes\n",
|
|
ao_data.buffersize);
|
|
}
|
|
|
|
memset(¶ms, 0, sizeof(params));
|
|
params.channel = SND_PCM_CHANNEL_PLAYBACK;
|
|
params.mode = SND_PCM_MODE_STREAM;
|
|
params.format = alsa_format;
|
|
params.start_mode = SND_PCM_START_DATA;
|
|
params.stop_mode = SND_PCM_STOP_ROLLOVER;
|
|
params.buf.stream.queue_size = ao_data.buffersize;
|
|
params.buf.stream.fill = SND_PCM_FILL_NONE;
|
|
|
|
if ((err = snd_pcm_channel_params(alsa_handler, ¶ms)) < 0)
|
|
{
|
|
mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: error setting parameters: %s\n", snd_strerror(err));
|
|
return(0);
|
|
}
|
|
|
|
memset(&setup, 0, sizeof(setup));
|
|
setup.channel = SND_PCM_CHANNEL_PLAYBACK;
|
|
setup.mode = SND_PCM_MODE_STREAM;
|
|
setup.format = alsa_format;
|
|
setup.buf.stream.queue_size = ao_data.buffersize;
|
|
setup.msbits_per_sample = ao_data.bps;
|
|
|
|
if ((err = snd_pcm_channel_setup(alsa_handler, &setup)) < 0)
|
|
{
|
|
mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: error setting up channel: %s\n", snd_strerror(err));
|
|
return(0);
|
|
}
|
|
|
|
if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
|
|
{
|
|
mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: channel prepare error: %s\n", snd_strerror(err));
|
|
return(0);
|
|
}
|
|
|
|
mp_msg(MSGT_AO, MSGL_INFO, "AUDIO: %d Hz/%d channels/%d bps/%d bytes buffer/%s\n",
|
|
ao_data.samplerate, ao_data.channels, ao_data.bps, ao_data.buffersize,
|
|
snd_pcm_get_format_name(alsa_format.format));
|
|
return(1);
|
|
}
|
|
|
|
/* close audio device */
|
|
static void uninit()
|
|
{
|
|
int err;
|
|
|
|
if ((err = snd_pcm_playback_drain(alsa_handler)) < 0)
|
|
{
|
|
mp_msg(MSGT_AO, MSGL_ERR, "alsa-uninit: playback drain error: %s\n", snd_strerror(err));
|
|
return;
|
|
}
|
|
|
|
if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
|
|
{
|
|
mp_msg(MSGT_AO, MSGL_ERR, "alsa-uninit: playback flush error: %s\n", snd_strerror(err));
|
|
return;
|
|
}
|
|
|
|
if ((err = snd_pcm_close(alsa_handler)) < 0)
|
|
{
|
|
mp_msg(MSGT_AO, MSGL_ERR, "alsa-uninit: pcm close error: %s\n", snd_strerror(err));
|
|
return;
|
|
}
|
|
}
|
|
|
|
/* stop playing and empty buffers (for seeking/pause) */
|
|
static void reset()
|
|
{
|
|
int err;
|
|
|
|
if ((err = snd_pcm_playback_drain(alsa_handler)) < 0)
|
|
{
|
|
mp_msg(MSGT_AO, MSGL_ERR, "alsa-reset: playback drain error: %s\n", snd_strerror(err));
|
|
return;
|
|
}
|
|
|
|
if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
|
|
{
|
|
mp_msg(MSGT_AO, MSGL_ERR, "alsa-reset: playback flush error: %s\n", snd_strerror(err));
|
|
return;
|
|
}
|
|
|
|
if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
|
|
{
|
|
mp_msg(MSGT_AO, MSGL_ERR, "alsa-reset: channel prepare error: %s\n", snd_strerror(err));
|
|
return;
|
|
}
|
|
}
|
|
|
|
/* stop playing, keep buffers (for pause) */
|
|
static void audio_pause()
|
|
{
|
|
int err;
|
|
|
|
if ((err = snd_pcm_playback_drain(alsa_handler)) < 0)
|
|
{
|
|
mp_msg(MSGT_AO, MSGL_ERR, "alsa-pause: playback drain error: %s\n", snd_strerror(err));
|
|
return;
|
|
}
|
|
|
|
if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
|
|
{
|
|
mp_msg(MSGT_AO, MSGL_ERR, "alsa-pause: playback flush error: %s\n", snd_strerror(err));
|
|
return;
|
|
}
|
|
}
|
|
|
|
/* resume playing, after audio_pause() */
|
|
static void audio_resume()
|
|
{
|
|
int err;
|
|
if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
|
|
{
|
|
mp_msg(MSGT_AO, MSGL_ERR, "alsa-resume: channel prepare error: %s\n", snd_strerror(err));
|
|
return;
|
|
}
|
|
}
|
|
|
|
/*
|
|
plays 'len' bytes of 'data'
|
|
returns: number of bytes played
|
|
*/
|
|
static int play(void* data, int len, int flags)
|
|
{
|
|
int got_len;
|
|
|
|
if (!len)
|
|
return(0);
|
|
|
|
if ((got_len = snd_pcm_write(alsa_handler, data, len)) < 0)
|
|
{
|
|
if (got_len == -EPIPE) /* underrun? */
|
|
{
|
|
mp_msg(MSGT_AO, MSGL_ERR, "alsa-play: alsa underrun, resetting stream\n");
|
|
if ((got_len = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
|
|
{
|
|
mp_msg(MSGT_AO, MSGL_ERR, "alsa-play: playback prepare error: %s\n", snd_strerror(got_len));
|
|
return(0);
|
|
}
|
|
if ((got_len = snd_pcm_write(alsa_handler, data, len)) < 0)
|
|
{
|
|
mp_msg(MSGT_AO, MSGL_ERR, "alsa-play: write error after reset: %s - giving up\n",
|
|
snd_strerror(got_len));
|
|
return(0);
|
|
}
|
|
return(got_len); /* 2nd write was ok */
|
|
}
|
|
mp_msg(MSGT_AO, MSGL_ERR, "alsa-play: output error: %s\n", snd_strerror(got_len));
|
|
return(0);
|
|
}
|
|
return(got_len);
|
|
}
|
|
|
|
/* how many byes are free in the buffer */
|
|
static int get_space()
|
|
{
|
|
snd_pcm_channel_status_t ch_stat;
|
|
|
|
ch_stat.channel = SND_PCM_CHANNEL_PLAYBACK;
|
|
|
|
if (snd_pcm_channel_status(alsa_handler, &ch_stat) < 0)
|
|
return(0); /* error occured */
|
|
else
|
|
return(ch_stat.free);
|
|
}
|
|
|
|
/* delay in seconds between first and last sample in buffer */
|
|
static float get_delay()
|
|
{
|
|
snd_pcm_channel_status_t ch_stat;
|
|
|
|
ch_stat.channel = SND_PCM_CHANNEL_PLAYBACK;
|
|
|
|
if (snd_pcm_channel_status(alsa_handler, &ch_stat) < 0)
|
|
return((float)ao_data.buffersize/(float)ao_data.bps); /* error occured */
|
|
else
|
|
return((float)ch_stat.count/(float)ao_data.bps);
|
|
}
|