Mercurial > mplayer.hg
view libao2/ao_alsa5.c @ 24590:2c238fa777ff
ao_alsa: Fix get_space() return values larger than buffersize
After a buffer underrun the ALSA get_space() function sometimes returned
values larger than the ao had set in ao_data.buffersize. Fix this by
replacing the old check against MAX_OUTBURST by one against
ao_data.buffersize. There should be no need for the MAX_OUTBURST check;
the current MPlayer side should no longer have any constant limit on the
amount of data an ao can buffer or request at once.
The get_space() values larger than ao_data.buffersize triggered errors
in audio decoding causing the current attempt to fill audio buffers to
be aborted. I'm not sure how often that caused behavior noticeably worse
then an underrun already is.
author | uau |
---|---|
date | Mon, 24 Sep 2007 21:49:58 +0000 |
parents | 99e20a22d5d0 |
children | 0fdf04b07ecb |
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/* 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 "libaf/af_format.h" #include "mp_msg.h" #include "help_mp.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, MSGTR_AO_ALSA5_InitInfo, rate_hz, channels, af_fmt2str_short(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, MSGTR_AO_ALSA5_SoundCardNotFound); 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 AF_FORMAT_S8: alsa_format.format = SND_PCM_SFMT_S8; break; case AF_FORMAT_U8: alsa_format.format = SND_PCM_SFMT_U8; break; case AF_FORMAT_U16_LE: alsa_format.format = SND_PCM_SFMT_U16_LE; break; case AF_FORMAT_U16_BE: alsa_format.format = SND_PCM_SFMT_U16_BE; break; #ifndef WORDS_BIGENDIAN case AF_FORMAT_AC3: #endif case AF_FORMAT_S16_LE: alsa_format.format = SND_PCM_SFMT_S16_LE; break; #ifdef WORDS_BIGENDIAN case AF_FORMAT_AC3: #endif case AF_FORMAT_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, MSGTR_AO_ALSA5_InvalidFormatReq,af_fmt2str_short(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, MSGTR_AO_ALSA5_PlayBackError, snd_strerror(err)); return(0); } if ((err = snd_pcm_info(alsa_handler, &info)) < 0) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PcmInfoError, snd_strerror(err)); return(0); } mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ALSA5_SoundcardsFound, 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, MSGTR_AO_ALSA5_PcmChanInfoError, 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, MSGTR_AO_ALSA5_CantSetParms, 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, MSGTR_AO_ALSA5_CantSetChan, snd_strerror(err)); return(0); } if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ChanPrepareError, 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, MSGTR_AO_ALSA5_DrainError, snd_strerror(err)); return; } if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_FlushError, snd_strerror(err)); return; } if ((err = snd_pcm_close(alsa_handler)) < 0) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PcmCloseError, snd_strerror(err)); return; } } /* stop playing and empty buffers (for seeking/pause) */ static void reset(void) { int err; if ((err = snd_pcm_playback_drain(alsa_handler)) < 0) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ResetDrainError, snd_strerror(err)); return; } if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ResetFlushError, snd_strerror(err)); return; } if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ResetChanPrepareError, snd_strerror(err)); return; } } /* stop playing, keep buffers (for pause) */ static void audio_pause(void) { int err; if ((err = snd_pcm_playback_drain(alsa_handler)) < 0) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PauseDrainError, snd_strerror(err)); return; } if ((err = snd_pcm_channel_flush(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PauseFlushError, snd_strerror(err)); return; } } /* resume playing, after audio_pause() */ static void audio_resume(void) { int err; if ((err = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_ResumePrepareError, 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, MSGTR_AO_ALSA5_Underrun); if ((got_len = snd_pcm_channel_prepare(alsa_handler, SND_PCM_CHANNEL_PLAYBACK)) < 0) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_PlaybackPrepareError, snd_strerror(got_len)); return(0); } if ((got_len = snd_pcm_write(alsa_handler, data, len)) < 0) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_WriteErrorAfterReset, snd_strerror(got_len)); return(0); } return(got_len); /* 2nd write was ok */ } mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ALSA5_OutPutError, snd_strerror(got_len)); return(0); } return(got_len); } /* how many byes are free in the buffer */ static int get_space(void) { 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(void) { 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); }