Mercurial > mplayer.hg
view stream/ai_alsa.c @ 29626:2a2bc7ee5b81
Add w64 to list of preferred lavf formats (otherwise demux_audio incorrectly
claims the Wave64 files but can not handle them).
Patch by Daniel Verkamp [daniel drv nu].
author | reimar |
---|---|
date | Thu, 10 Sep 2009 17:30:02 +0000 |
parents | 0f1b5b68af32 |
children | ce0122361a39 |
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#include <stdio.h> #include <stdlib.h> #include <sys/time.h> #include <alloca.h> #include "config.h" #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; }