mpv/libao2/ao_alsa5.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(&params, 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, &params)) < 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);
}