mirror of
https://github.com/mpv-player/mpv
synced 2024-12-15 19:35:49 +00:00
504270e549
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@12146 b3059339-0415-0410-9bf9-f77b7e298cf2
375 lines
9.3 KiB
C
375 lines
9.3 KiB
C
/*
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ao_alsa5 - ALSA-0.5.x output plugin for MPlayer
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(C) Alex Beregszaszi
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Thanks to Arpi for helping me ;)
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*/
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#include <errno.h>
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#include <sys/asoundlib.h>
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#include "../config.h"
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#include "audio_out.h"
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#include "audio_out_internal.h"
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#include "afmt.h"
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#include "../mp_msg.h"
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static ao_info_t info =
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{
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"ALSA-0.5.x audio output",
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"alsa5",
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"Alex Beregszaszi",
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""
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};
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LIBAO_EXTERN(alsa5)
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static snd_pcm_t *alsa_handler;
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static snd_pcm_format_t alsa_format;
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static int alsa_rate = SND_PCM_RATE_CONTINUOUS;
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/* to set/get/query special features/parameters */
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static int control(int cmd, void *arg)
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{
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return(CONTROL_UNKNOWN);
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}
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/*
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open & setup audio device
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return: 1=success 0=fail
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*/
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static int init(int rate_hz, int channels, int format, int flags)
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{
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int err;
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int cards = -1;
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snd_pcm_channel_params_t params;
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snd_pcm_channel_setup_t setup;
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snd_pcm_info_t info;
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snd_pcm_channel_info_t chninfo;
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mp_msg(MSGT_AO, MSGL_INFO, "alsa-init: requested format: %d Hz, %d channels, %s\n", rate_hz,
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channels, audio_out_format_name(format));
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alsa_handler = NULL;
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mp_msg(MSGT_AO, MSGL_V, "alsa-init: compiled for ALSA-%s (%d)\n", SND_LIB_VERSION_STR,
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SND_LIB_VERSION);
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if ((cards = snd_cards()) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: no soundcards found\n");
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return(0);
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}
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ao_data.format = format;
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ao_data.channels = channels;
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ao_data.samplerate = rate_hz;
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ao_data.bps = ao_data.samplerate*ao_data.channels;
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ao_data.outburst = OUTBURST;
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ao_data.buffersize = 16384;
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memset(&alsa_format, 0, sizeof(alsa_format));
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switch (format)
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{
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case AFMT_S8:
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alsa_format.format = SND_PCM_SFMT_S8;
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break;
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case AFMT_U8:
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alsa_format.format = SND_PCM_SFMT_U8;
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break;
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case AFMT_U16_LE:
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alsa_format.format = SND_PCM_SFMT_U16_LE;
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break;
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case AFMT_U16_BE:
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alsa_format.format = SND_PCM_SFMT_U16_BE;
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break;
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#ifndef WORDS_BIGENDIAN
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case AFMT_AC3:
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#endif
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case AFMT_S16_LE:
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alsa_format.format = SND_PCM_SFMT_S16_LE;
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break;
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#ifdef WORDS_BIGENDIAN
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case AFMT_AC3:
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#endif
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case AFMT_S16_BE:
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alsa_format.format = SND_PCM_SFMT_S16_BE;
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break;
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default:
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alsa_format.format = SND_PCM_SFMT_MPEG;
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break;
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}
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switch(alsa_format.format)
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{
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case SND_PCM_SFMT_S16_LE:
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case SND_PCM_SFMT_U16_LE:
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ao_data.bps *= 2;
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break;
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case -1:
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: invalid format (%s) requested - output disabled\n",
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audio_out_format_name(format));
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return(0);
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default:
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break;
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}
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switch(rate_hz)
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{
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case 8000:
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alsa_rate = SND_PCM_RATE_8000;
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break;
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case 11025:
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alsa_rate = SND_PCM_RATE_11025;
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break;
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case 16000:
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alsa_rate = SND_PCM_RATE_16000;
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break;
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case 22050:
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alsa_rate = SND_PCM_RATE_22050;
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break;
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case 32000:
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alsa_rate = SND_PCM_RATE_32000;
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break;
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case 44100:
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alsa_rate = SND_PCM_RATE_44100;
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break;
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case 48000:
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alsa_rate = SND_PCM_RATE_48000;
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break;
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case 88200:
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alsa_rate = SND_PCM_RATE_88200;
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break;
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case 96000:
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alsa_rate = SND_PCM_RATE_96000;
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break;
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case 176400:
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alsa_rate = SND_PCM_RATE_176400;
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break;
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case 192000:
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alsa_rate = SND_PCM_RATE_192000;
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break;
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default:
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alsa_rate = SND_PCM_RATE_CONTINUOUS;
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break;
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}
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alsa_format.rate = ao_data.samplerate;
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alsa_format.voices = ao_data.channels;
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alsa_format.interleave = 1;
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if ((err = snd_pcm_open(&alsa_handler, 0, 0, SND_PCM_OPEN_PLAYBACK)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: playback open error: %s\n", snd_strerror(err));
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return(0);
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}
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if ((err = snd_pcm_info(alsa_handler, &info)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: pcm info error: %s\n", snd_strerror(err));
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return(0);
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}
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mp_msg(MSGT_AO, MSGL_INFO, "alsa-init: %d soundcard(s) found, using: %s\n",
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cards, info.name);
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if (info.flags & SND_PCM_INFO_PLAYBACK)
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{
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memset(&chninfo, 0, sizeof(chninfo));
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chninfo.channel = SND_PCM_CHANNEL_PLAYBACK;
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if ((err = snd_pcm_channel_info(alsa_handler, &chninfo)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: pcm channel info error: %s\n", snd_strerror(err));
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return(0);
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}
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#ifndef __QNX__
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if (chninfo.buffer_size)
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ao_data.buffersize = chninfo.buffer_size;
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#endif
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mp_msg(MSGT_AO, MSGL_V, "alsa-init: setting preferred buffer size from driver: %d bytes\n",
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ao_data.buffersize);
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}
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memset(¶ms, 0, sizeof(params));
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params.channel = SND_PCM_CHANNEL_PLAYBACK;
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params.mode = SND_PCM_MODE_STREAM;
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params.format = alsa_format;
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params.start_mode = SND_PCM_START_DATA;
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params.stop_mode = SND_PCM_STOP_ROLLOVER;
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params.buf.stream.queue_size = ao_data.buffersize;
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params.buf.stream.fill = SND_PCM_FILL_NONE;
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if ((err = snd_pcm_channel_params(alsa_handler, ¶ms)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: error setting parameters: %s\n", snd_strerror(err));
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return(0);
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}
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memset(&setup, 0, sizeof(setup));
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setup.channel = SND_PCM_CHANNEL_PLAYBACK;
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setup.mode = SND_PCM_MODE_STREAM;
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setup.format = alsa_format;
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setup.buf.stream.queue_size = ao_data.buffersize;
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setup.msbits_per_sample = ao_data.bps;
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if ((err = snd_pcm_channel_setup(alsa_handler, &setup)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: error setting up channel: %s\n", snd_strerror(err));
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return(0);
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}
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if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-init: channel prepare error: %s\n", snd_strerror(err));
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return(0);
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}
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mp_msg(MSGT_AO, MSGL_INFO, "AUDIO: %d Hz/%d channels/%d bps/%d bytes buffer/%s\n",
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ao_data.samplerate, ao_data.channels, ao_data.bps, ao_data.buffersize,
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snd_pcm_get_format_name(alsa_format.format));
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return(1);
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}
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/* close audio device */
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static void uninit(int immed)
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{
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int err;
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if ((err = snd_pcm_playback_drain(alsa_handler)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-uninit: playback drain error: %s\n", snd_strerror(err));
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return;
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}
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if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-uninit: playback flush error: %s\n", snd_strerror(err));
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return;
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}
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if ((err = snd_pcm_close(alsa_handler)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-uninit: pcm close error: %s\n", snd_strerror(err));
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return;
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}
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}
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/* stop playing and empty buffers (for seeking/pause) */
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static void reset()
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{
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int err;
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if ((err = snd_pcm_playback_drain(alsa_handler)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-reset: playback drain error: %s\n", snd_strerror(err));
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return;
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}
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if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-reset: playback flush error: %s\n", snd_strerror(err));
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return;
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}
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if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-reset: channel prepare error: %s\n", snd_strerror(err));
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return;
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}
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}
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/* stop playing, keep buffers (for pause) */
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static void audio_pause()
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{
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int err;
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if ((err = snd_pcm_playback_drain(alsa_handler)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-pause: playback drain error: %s\n", snd_strerror(err));
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return;
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}
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if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-pause: playback flush error: %s\n", snd_strerror(err));
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return;
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}
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}
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/* resume playing, after audio_pause() */
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static void audio_resume()
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{
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int err;
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if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-resume: channel prepare error: %s\n", snd_strerror(err));
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return;
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}
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}
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/*
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plays 'len' bytes of 'data'
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returns: number of bytes played
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*/
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static int play(void* data, int len, int flags)
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{
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int got_len;
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if (!len)
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return(0);
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if ((got_len = snd_pcm_write(alsa_handler, data, len)) < 0)
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{
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if (got_len == -EPIPE) /* underrun? */
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-play: alsa underrun, resetting stream\n");
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if ((got_len = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-play: playback prepare error: %s\n", snd_strerror(got_len));
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return(0);
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}
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if ((got_len = snd_pcm_write(alsa_handler, data, len)) < 0)
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{
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-play: write error after reset: %s - giving up\n",
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snd_strerror(got_len));
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return(0);
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}
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return(got_len); /* 2nd write was ok */
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}
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mp_msg(MSGT_AO, MSGL_ERR, "alsa-play: output error: %s\n", snd_strerror(got_len));
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return(0);
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}
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return(got_len);
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}
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/* how many byes are free in the buffer */
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static int get_space()
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{
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snd_pcm_channel_status_t ch_stat;
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ch_stat.channel = SND_PCM_CHANNEL_PLAYBACK;
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if (snd_pcm_channel_status(alsa_handler, &ch_stat) < 0)
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return(0); /* error occurred */
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else
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return(ch_stat.free);
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}
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/* delay in seconds between first and last sample in buffer */
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static float get_delay()
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{
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snd_pcm_channel_status_t ch_stat;
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ch_stat.channel = SND_PCM_CHANNEL_PLAYBACK;
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if (snd_pcm_channel_status(alsa_handler, &ch_stat) < 0)
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return((float)ao_data.buffersize/(float)ao_data.bps); /* error occurred */
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else
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return((float)ch_stat.count/(float)ao_data.bps);
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}
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