Mercurial > mplayer.hg
view libmpdemux/ai_alsa.c @ 18150:710d4bc5f8c9
Using channel count, samplerate and input bps values from the container
instead of the decoder breaks some DTS samples where the container says
the audio has 6 channels but the decoder gives 2. In this case take the
number of channels from the decoder instead, the output will almost
certainly be badly garbled anyway if the number of channels is wrong.
patch by Uoti Urpala, uoti <<.>> urpala <<@>> pp1 <<.>> inet <<.>> fi
author | diego |
---|---|
date | Wed, 19 Apr 2006 20:12:01 +0000 |
parents | dfbe8cd0e081 |
children | d2d9d011203f |
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#include <stdio.h> #include <stdlib.h> #include <sys/time.h> #include "config.h" #if defined(USE_TV) && (defined(HAVE_TV_V4L) || defined(HAVE_TV_V4L2)) && defined(HAVE_ALSA9) #include <alsa/asoundlib.h> #include "audio_in.h" #include "mp_msg.h" #include "help_mp.h" int ai_alsa_setup(audio_in_t *ai) { snd_pcm_hw_params_t *params; snd_pcm_sw_params_t *swparams; int buffer_size; int err; unsigned int rate; snd_pcm_hw_params_alloca(¶ms); snd_pcm_sw_params_alloca(&swparams); err = snd_pcm_hw_params_any(ai->alsa.handle, params); if (err < 0) { mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_PcmBrokenConfig); return -1; } err = snd_pcm_hw_params_set_access(ai->alsa.handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); if (err < 0) { mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_UnavailableAccessType); return -1; } err = snd_pcm_hw_params_set_format(ai->alsa.handle, params, SND_PCM_FORMAT_S16_LE); if (err < 0) { mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_UnavailableSampleFmt); return -1; } err = snd_pcm_hw_params_set_channels(ai->alsa.handle, params, ai->req_channels); if (err < 0) { ai->channels = snd_pcm_hw_params_get_channels(params); mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_UnavailableChanCount, ai->channels); } else { ai->channels = ai->req_channels; } err = snd_pcm_hw_params_set_rate_near(ai->alsa.handle, params, ai->req_samplerate, 0); assert(err >= 0); rate = err; ai->samplerate = rate; ai->alsa.buffer_time = 1000000; ai->alsa.buffer_time = snd_pcm_hw_params_set_buffer_time_near(ai->alsa.handle, params, ai->alsa.buffer_time, 0); assert(ai->alsa.buffer_time >= 0); ai->alsa.period_time = ai->alsa.buffer_time / 4; ai->alsa.period_time = snd_pcm_hw_params_set_period_time_near(ai->alsa.handle, params, ai->alsa.period_time, 0); assert(ai->alsa.period_time >= 0); err = snd_pcm_hw_params(ai->alsa.handle, params); if (err < 0) { mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_CannotInstallHWParams); snd_pcm_hw_params_dump(params, ai->alsa.log); return -1; } ai->alsa.chunk_size = snd_pcm_hw_params_get_period_size(params, 0); buffer_size = snd_pcm_hw_params_get_buffer_size(params); if (ai->alsa.chunk_size == buffer_size) { mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_PeriodEqualsBufferSize, ai->alsa.chunk_size, (long)buffer_size); return -1; } snd_pcm_sw_params_current(ai->alsa.handle, swparams); err = snd_pcm_sw_params_set_sleep_min(ai->alsa.handle, swparams,0); assert(err >= 0); err = snd_pcm_sw_params_set_avail_min(ai->alsa.handle, swparams, ai->alsa.chunk_size); assert(err >= 0); err = snd_pcm_sw_params_set_start_threshold(ai->alsa.handle, swparams, 0); assert(err >= 0); err = snd_pcm_sw_params_set_stop_threshold(ai->alsa.handle, swparams, buffer_size); assert(err >= 0); assert(err >= 0); if (snd_pcm_sw_params(ai->alsa.handle, swparams) < 0) { mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_CannotInstallSWParams); snd_pcm_sw_params_dump(swparams, ai->alsa.log); return -1; } if (mp_msg_test(MSGT_TV, MSGL_V)) { snd_pcm_dump(ai->alsa.handle, ai->alsa.log); } ai->alsa.bits_per_sample = snd_pcm_format_physical_width(SND_PCM_FORMAT_S16_LE); ai->alsa.bits_per_frame = ai->alsa.bits_per_sample * ai->channels; ai->blocksize = ai->alsa.chunk_size * ai->alsa.bits_per_frame / 8; ai->samplesize = ai->alsa.bits_per_sample; ai->bytes_per_sample = ai->alsa.bits_per_sample/8; return 0; } int ai_alsa_init(audio_in_t *ai) { int err; err = snd_pcm_open(&ai->alsa.handle, ai->alsa.device, SND_PCM_STREAM_CAPTURE, 0); if (err < 0) { mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_ErrorOpeningAudio, snd_strerror(err)); return -1; } err = snd_output_stdio_attach(&ai->alsa.log, stderr, 0); if (err < 0) { return -1; } err = ai_alsa_setup(ai); return err; } #ifndef timersub #define timersub(a, b, result) \ do { \ (result)->tv_sec = (a)->tv_sec - (b)->tv_sec; \ (result)->tv_usec = (a)->tv_usec - (b)->tv_usec; \ if ((result)->tv_usec < 0) { \ --(result)->tv_sec; \ (result)->tv_usec += 1000000; \ } \ } while (0) #endif int ai_alsa_xrun(audio_in_t *ai) { snd_pcm_status_t *status; int res; snd_pcm_status_alloca(&status); if ((res = snd_pcm_status(ai->alsa.handle, status))<0) { mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_AlsaStatusError, snd_strerror(res)); return -1; } if (snd_pcm_status_get_state(status) == SND_PCM_STATE_XRUN) { struct timeval now, diff, tstamp; gettimeofday(&now, 0); snd_pcm_status_get_trigger_tstamp(status, &tstamp); timersub(&now, &tstamp, &diff); mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_AlsaXRUN, diff.tv_sec * 1000 + diff.tv_usec / 1000.0); if (mp_msg_test(MSGT_TV, MSGL_V)) { mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_AlsaStatus); snd_pcm_status_dump(status, ai->alsa.log); } if ((res = snd_pcm_prepare(ai->alsa.handle))<0) { mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_AlsaXRUNPrepareError, snd_strerror(res)); return -1; } return 0; /* ok, data should be accepted again */ } mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AIALSA_AlsaReadWriteError); return -1; } #endif /* HAVE_ALSA9 */