Mercurial > mplayer.hg
view libao2/ao_alsa5.c @ 13316:0d17bef9894d
* Changed malloc and strncpy to strdup. Less code.
* More error checking. If malloc or strdup fails, print message and exit.
* Free malloc'd memory when uninit is called.
* Moved default of jpeg_outdir to preinit, so it is always malloc'd and can
easily be freed at uninit.
author | ivo |
---|---|
date | Sat, 11 Sep 2004 19:59:31 +0000 |
parents | 99798c3cdb93 |
children | c1955840883d |
line wrap: on
line source
/* ao_alsa5 - ALSA-0.5.x output plugin for MPlayer (C) Alex Beregszaszi 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" static ao_info_t info = { "ALSA-0.5.x audio output", "alsa5", "Alex Beregszaszi", "" }; 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, void *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; 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 immed) { 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 occurred */ 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 occurred */ else return((float)ch_stat.count/(float)ao_data.bps); }