| /* |
| * 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, |
| }; |