/* * 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 can 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 "libavutil/internal.h" #include "libavutil/time.h" #include "libavformat/internal.h" #include "avdevice.h" #include "alsa.h" static av_cold int audio_write_header(AVFormatContext *s1) { AlsaData *s = s1->priv_data; AVStream *st = NULL; unsigned int sample_rate; enum AVCodecID codec_id; int res; if (s1->nb_streams != 1 || s1->streams[0]->codecpar->codec_type != AVMEDIA_TYPE_AUDIO) { av_log(s1, AV_LOG_ERROR, "Only a single audio stream is supported.\n"); return AVERROR(EINVAL); } st = s1->streams[0]; sample_rate = st->codecpar->sample_rate; codec_id = st->codecpar->codec_id; res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate, st->codecpar->channels, &codec_id); if (sample_rate != st->codecpar->sample_rate) { av_log(s1, AV_LOG_ERROR, "sample rate %d not available, nearest is %d\n", st->codecpar->sample_rate, sample_rate); goto fail; } avpriv_set_pts_info(st, 64, 1, sample_rate); 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 (pkt->dts != AV_NOPTS_VALUE) s->timestamp = pkt->dts; s->timestamp += pkt->duration ? pkt->duration : 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; } static int audio_write_frame(AVFormatContext *s1, int stream_index, AVFrame **frame, unsigned flags) { AlsaData *s = s1->priv_data; AVPacket pkt; /* ff_alsa_open() should have accepted only supported formats */ if ((flags & AV_WRITE_UNCODED_FRAME_QUERY)) return av_sample_fmt_is_planar(s1->streams[stream_index]->codecpar->format) ? AVERROR(EINVAL) : 0; /* set only used fields */ pkt.data = (*frame)->data[0]; pkt.size = (*frame)->nb_samples * s->frame_size; pkt.dts = (*frame)->pkt_dts; pkt.duration = (*frame)->pkt_duration; return audio_write_packet(s1, &pkt); } static void audio_get_output_timestamp(AVFormatContext *s1, int stream, int64_t *dts, int64_t *wall) { AlsaData *s = s1->priv_data; snd_pcm_sframes_t delay = 0; *wall = av_gettime(); snd_pcm_delay(s->h, &delay); *dts = s->timestamp - delay; } static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list) { return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_PLAYBACK); } static const AVClass alsa_muxer_class = { .class_name = "ALSA outdev", .item_name = av_default_item_name, .version = LIBAVUTIL_VERSION_INT, .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT, }; AVOutputFormat ff_alsa_muxer = { .name = "alsa", .long_name = NULL_IF_CONFIG_SMALL("ALSA audio output"), .priv_data_size = sizeof(AlsaData), .audio_codec = DEFAULT_CODEC_ID, .video_codec = AV_CODEC_ID_NONE, .write_header = audio_write_header, .write_packet = audio_write_packet, .write_trailer = ff_alsa_close, .write_uncoded_frame = audio_write_frame, .get_device_list = audio_get_device_list, .get_output_timestamp = audio_get_output_timestamp, .flags = AVFMT_NOFILE, .priv_class = &alsa_muxer_class, };