1/* 2 * ALSA input and output 3 * Copyright (c) 2007 Luca Abeni ( lucabe72 email it ) 4 * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr ) 5 * 6 * This file is part of FFmpeg. 7 * 8 * FFmpeg is free software; you can redistribute it and/or 9 * modify it under the terms of the GNU Lesser General Public 10 * License as published by the Free Software Foundation; either 11 * version 2.1 of the License, or (at your option) any later version. 12 * 13 * FFmpeg is distributed in the hope that it will be useful, 14 * but WITHOUT ANY WARRANTY; without even the implied warranty of 15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 16 * Lesser General Public License for more details. 17 * 18 * You should have received a copy of the GNU Lesser General Public 19 * License along with FFmpeg; if not, write to the Free Software 20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 21 */ 22 23/** 24 * @file 25 * ALSA input and output: output 26 * @author Luca Abeni ( lucabe72 email it ) 27 * @author Benoit Fouet ( benoit fouet free fr ) 28 * 29 * This avdevice encoder allows to play audio to an ALSA (Advanced Linux 30 * Sound Architecture) device. 31 * 32 * The filename parameter is the name of an ALSA PCM device capable of 33 * capture, for example "default" or "plughw:1"; see the ALSA documentation 34 * for naming conventions. The empty string is equivalent to "default". 35 * 36 * The playback period is set to the lower value available for the device, 37 * which gives a low latency suitable for real-time playback. 38 */ 39 40#include <alsa/asoundlib.h> 41 42#include "libavutil/time.h" 43#include "libavformat/internal.h" 44#include "avdevice.h" 45#include "alsa-audio.h" 46 47static av_cold int audio_write_header(AVFormatContext *s1) 48{ 49 AlsaData *s = s1->priv_data; 50 AVStream *st = NULL; 51 unsigned int sample_rate; 52 enum AVCodecID codec_id; 53 int res; 54 55 if (s1->nb_streams != 1 || s1->streams[0]->codec->codec_type != AVMEDIA_TYPE_AUDIO) { 56 av_log(s1, AV_LOG_ERROR, "Only a single audio stream is supported.\n"); 57 return AVERROR(EINVAL); 58 } 59 st = s1->streams[0]; 60 61 sample_rate = st->codec->sample_rate; 62 codec_id = st->codec->codec_id; 63 res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate, 64 st->codec->channels, &codec_id); 65 if (sample_rate != st->codec->sample_rate) { 66 av_log(s1, AV_LOG_ERROR, 67 "sample rate %d not available, nearest is %d\n", 68 st->codec->sample_rate, sample_rate); 69 goto fail; 70 } 71 avpriv_set_pts_info(st, 64, 1, sample_rate); 72 73 return res; 74 75fail: 76 snd_pcm_close(s->h); 77 return AVERROR(EIO); 78} 79 80static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt) 81{ 82 AlsaData *s = s1->priv_data; 83 int res; 84 int size = pkt->size; 85 uint8_t *buf = pkt->data; 86 87 size /= s->frame_size; 88 if (pkt->dts != AV_NOPTS_VALUE) 89 s->timestamp = pkt->dts; 90 s->timestamp += pkt->duration ? pkt->duration : size; 91 92 if (s->reorder_func) { 93 if (size > s->reorder_buf_size) 94 if (ff_alsa_extend_reorder_buf(s, size)) 95 return AVERROR(ENOMEM); 96 s->reorder_func(buf, s->reorder_buf, size); 97 buf = s->reorder_buf; 98 } 99 while ((res = snd_pcm_writei(s->h, buf, size)) < 0) { 100 if (res == -EAGAIN) { 101 102 return AVERROR(EAGAIN); 103 } 104 105 if (ff_alsa_xrun_recover(s1, res) < 0) { 106 av_log(s1, AV_LOG_ERROR, "ALSA write error: %s\n", 107 snd_strerror(res)); 108 109 return AVERROR(EIO); 110 } 111 } 112 113 return 0; 114} 115 116static int audio_write_frame(AVFormatContext *s1, int stream_index, 117 AVFrame **frame, unsigned flags) 118{ 119 AlsaData *s = s1->priv_data; 120 AVPacket pkt; 121 122 /* ff_alsa_open() should have accepted only supported formats */ 123 if ((flags & AV_WRITE_UNCODED_FRAME_QUERY)) 124 return av_sample_fmt_is_planar(s1->streams[stream_index]->codec->sample_fmt) ? 125 AVERROR(EINVAL) : 0; 126 /* set only used fields */ 127 pkt.data = (*frame)->data[0]; 128 pkt.size = (*frame)->nb_samples * s->frame_size; 129 pkt.dts = (*frame)->pkt_dts; 130 pkt.duration = av_frame_get_pkt_duration(*frame); 131 return audio_write_packet(s1, &pkt); 132} 133 134static void 135audio_get_output_timestamp(AVFormatContext *s1, int stream, 136 int64_t *dts, int64_t *wall) 137{ 138 AlsaData *s = s1->priv_data; 139 snd_pcm_sframes_t delay = 0; 140 *wall = av_gettime(); 141 snd_pcm_delay(s->h, &delay); 142 *dts = s->timestamp - delay; 143} 144 145static const AVClass alsa_muxer_class = { 146 .class_name = "ALSA muxer", 147 .item_name = av_default_item_name, 148 .version = LIBAVUTIL_VERSION_INT, 149 .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT, 150}; 151 152AVOutputFormat ff_alsa_muxer = { 153 .name = "alsa", 154 .long_name = NULL_IF_CONFIG_SMALL("ALSA audio output"), 155 .priv_data_size = sizeof(AlsaData), 156 .audio_codec = DEFAULT_CODEC_ID, 157 .video_codec = AV_CODEC_ID_NONE, 158 .write_header = audio_write_header, 159 .write_packet = audio_write_packet, 160 .write_trailer = ff_alsa_close, 161 .write_uncoded_frame = audio_write_frame, 162 .get_output_timestamp = audio_get_output_timestamp, 163 .flags = AVFMT_NOFILE, 164 .priv_class = &alsa_muxer_class, 165}; 166