/* * ALSA input and output * Copyright (c) 2007 Luca Abeni ( lucabe72 email it ) * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr ) * * This file is part of Libav. * * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * Libav 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 * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * ALSA input and output: output * @author Luca Abeni ( lucabe72 email it ) * @author Benoit Fouet ( benoit fouet free fr ) * * This avdevice encoder allows to play audio to an ALSA (Advanced Linux * Sound Architecture) device. * * The filename parameter is the name of an ALSA PCM device capable of * capture, for example "default" or "plughw:1"; see the ALSA documentation * for naming conventions. The empty string is equivalent to "default". * * The playback period is set to the lower value available for the device, * which gives a low latency suitable for real-time playback. */ #include <alsa/asoundlib.h> #include "libavformat/avformat.h" #include "alsa-audio.h" static av_cold int audio_write_header(AVFormatContext *s1) { AlsaData *s = s1->priv_data; AVStream *st; unsigned int sample_rate; enum CodecID codec_id; int res; st = s1->streams[0]; sample_rate = st->codec->sample_rate; codec_id = st->codec->codec_id; res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate, st->codec->channels, &codec_id); if (sample_rate != st->codec->sample_rate) { av_log(s1, AV_LOG_ERROR, "sample rate %d not available, nearest is %d\n", st->codec->sample_rate, sample_rate); goto fail; } return res; fail: snd_pcm_close(s->h); return AVERROR(EIO); } static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt) { AlsaData *s = s1->priv_data; int res; int size = pkt->size; uint8_t *buf = pkt->data; while((res = snd_pcm_writei(s->h, buf, size / s->frame_size)) < 0) { if (res == -EAGAIN) { return AVERROR(EAGAIN); } if (ff_alsa_xrun_recover(s1, res) < 0) { av_log(s1, AV_LOG_ERROR, "ALSA write error: %s\n", snd_strerror(res)); return AVERROR(EIO); } } return 0; } AVOutputFormat ff_alsa_muxer = { "alsa", NULL_IF_CONFIG_SMALL("ALSA audio output"), "", "", sizeof(AlsaData), DEFAULT_CODEC_ID, CODEC_ID_NONE, audio_write_header, audio_write_packet, ff_alsa_close, .flags = AVFMT_NOFILE, };