/* * This file is part of MPlayer. * * MPlayer is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * MPlayer is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License along * with MPlayer; if not, write to the Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ #include #include #include #include "config.h" #include #include "audio_in.h" #include "common/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_ERR(ai, "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_ERR(ai, "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_ERR(ai, "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_ERR(ai, "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_ERR(ai, "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_ERR(ai, "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_ERR(ai, "Cannot set period time.\n"); } err = snd_pcm_hw_params(ai->alsa.handle, params); if (err < 0) { MP_ERR(ai, "Unable to install hardware parameters: %s", 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_ERR(ai, "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_avail_min(ai->alsa.handle, swparams, ai->alsa.chunk_size); err = snd_pcm_sw_params_set_start_threshold(ai->alsa.handle, swparams, 0); err = snd_pcm_sw_params_set_stop_threshold(ai->alsa.handle, swparams, buffer_size); if (snd_pcm_sw_params(ai->alsa.handle, swparams) < 0) { MP_ERR(ai, "Unable to install software parameters:\n"); snd_pcm_sw_params_dump(swparams, ai->alsa.log); return -1; } if (mp_msg_test(ai->log, 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_ERR(ai, "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_ERR(ai, "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_ERR(ai, "ALSA xrun!!! (at least %.3f ms long)\n", diff.tv_sec * 1000 + diff.tv_usec / 1000.0); if (mp_msg_test(ai->log, MSGL_V)) { MP_ERR(ai, "ALSA Status:\n"); snd_pcm_status_dump(status, ai->alsa.log); } if ((res = snd_pcm_prepare(ai->alsa.handle))<0) { MP_ERR(ai, "ALSA xrun: prepare error: %s", snd_strerror(res)); return -1; } return 0; /* ok, data should be accepted again */ } MP_ERR(ai, "ALSA read/write error"); return -1; }