ffmpeg/libavdevice/alsa-audio-enc.c

   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
  47 static 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
  75 fail:
  76     snd_pcm_close(s->h);
  77     return AVERROR(EIO);
  78 }
  79
  80 static 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
 116 static 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
 134 static void
 135 audio_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
 145 static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
 146 {
 147     return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_PLAYBACK);
 148 }
 149
 150 static const AVClass alsa_muxer_class = {
 151     .class_name     = "ALSA muxer",
 152     .item_name      = av_default_item_name,
 153     .version        = LIBAVUTIL_VERSION_INT,
 154     .category       = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT,
 155 };
 156
 157 AVOutputFormat ff_alsa_muxer = {
 158     .name           = "alsa",
 159     .long_name      = NULL_IF_CONFIG_SMALL("ALSA audio output"),
 160     .priv_data_size = sizeof(AlsaData),
 161     .audio_codec    = DEFAULT_CODEC_ID,
 162     .video_codec    = AV_CODEC_ID_NONE,
 163     .write_header   = audio_write_header,
 164     .write_packet   = audio_write_packet,
 165     .write_trailer  = ff_alsa_close,
 166     .write_uncoded_frame = audio_write_frame,
 167     .get_device_list = audio_get_device_list,
 168     .get_output_timestamp = audio_get_output_timestamp,
 169     .flags          = AVFMT_NOFILE,
 170     .priv_class     = &alsa_muxer_class,
 171 };