Mercurial > mplayer.hg
view libmpdemux/ai_alsa1x.c @ 14217:5b5ebf93ec16
Adds support for LADSPA (Linux Audio Developer's Simple Plugin API) plugins.
Compilation is optional and can be controled by configure. You need to
have the LADSPA SDK installed in order to have it autodetected by configure.
Manual page is updated.
author | ivo |
---|---|
date | Thu, 23 Dec 2004 02:09:52 +0000 |
parents | 66e491c35dc8 |
children | dfbe8cd0e081 |
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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_ALSA1X) #include <alsa/asoundlib.h> #include "audio_in.h" #include "mp_msg.h" int ai_alsa_setup(audio_in_t *ai) { snd_pcm_hw_params_t *params; snd_pcm_sw_params_t *swparams; snd_pcm_uframes_t buffer_size, period_size; int err; int dir; 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, "Broken configuration for this PCM: no configurations available\n"); 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, "Access type not available\n"); 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, "Sample format not available\n"); return -1; } err = snd_pcm_hw_params_set_channels(ai->alsa.handle, params, ai->req_channels); if (err < 0) { snd_pcm_hw_params_get_channels(params, &ai->channels); mp_msg(MSGT_TV, MSGL_ERR, "Channel count not available - reverting to default: %d\n", ai->channels); } else { ai->channels = ai->req_channels; } dir = 0; rate = ai->req_samplerate; err = snd_pcm_hw_params_set_rate_near(ai->alsa.handle, params, &rate, &dir); if (err < 0) { mp_msg(MSGT_TV, MSGL_ERR, "Cannot set samplerate\n"); } ai->samplerate = rate; dir = 0; ai->alsa.buffer_time = 1000000; err = snd_pcm_hw_params_set_buffer_time_near(ai->alsa.handle, params, &ai->alsa.buffer_time, &dir); if (err < 0) { mp_msg(MSGT_TV, MSGL_ERR, "Cannot set buffer time\n"); } dir = 0; ai->alsa.period_time = ai->alsa.buffer_time / 4; err = snd_pcm_hw_params_set_period_time_near(ai->alsa.handle, params, &ai->alsa.period_time, &dir); if (err < 0) { mp_msg(MSGT_TV, MSGL_ERR, "Cannot set period time\n"); } err = snd_pcm_hw_params(ai->alsa.handle, params); if (err < 0) { mp_msg(MSGT_TV, MSGL_ERR, "Unable to install hw params: %s\n", snd_strerror(err)); snd_pcm_hw_params_dump(params, ai->alsa.log); return -1; } dir = -1; snd_pcm_hw_params_get_period_size(params, &period_size, &dir); snd_pcm_hw_params_get_buffer_size(params, &buffer_size); ai->alsa.chunk_size = period_size; if (period_size == buffer_size) { mp_msg(MSGT_TV, MSGL_ERR, "Can't use period equal to buffer size (%u == %lu)\n", 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, "unable to install sw params:\n"); 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, "Error opening audio: %s\n", 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, "ALSA status error: %s", 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, "ALSA xrun!!! (at least %.3f ms long)\n", diff.tv_sec * 1000 + diff.tv_usec / 1000.0); if (mp_msg_test(MSGT_TV, MSGL_V)) { mp_msg(MSGT_TV, MSGL_ERR, "ALSA Status:\n"); snd_pcm_status_dump(status, ai->alsa.log); } if ((res = snd_pcm_prepare(ai->alsa.handle))<0) { mp_msg(MSGT_TV, MSGL_ERR, "ALSA xrun: prepare error: %s", snd_strerror(res)); return -1; } return 0; /* ok, data should be accepted again */ } mp_msg(MSGT_TV, MSGL_ERR, "ALSA read/write error"); return -1; } #endif /* HAVE_ALSA1X */