| /* |
| * 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 FFmpeg. |
| * |
| * FFmpeg 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. |
| * |
| * FFmpeg 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 FFmpeg; 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 "avdevice.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; |
| |
| size /= s->frame_size; |
| if (s->reorder_func) { |
| if (size > s->reorder_buf_size) |
| if (ff_alsa_extend_reorder_buf(s, size)) |
| return AVERROR(ENOMEM); |
| s->reorder_func(buf, s->reorder_buf, size); |
| buf = s->reorder_buf; |
| } |
| while ((res = snd_pcm_writei(s->h, buf, 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, |
| }; |