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00048 #include <alsa/asoundlib.h>
00049 #include "libavformat/internal.h"
00050 #include "libavutil/opt.h"
00051 #include "libavutil/mathematics.h"
00052
00053 #include "avdevice.h"
00054 #include "alsa-audio.h"
00055
00056 static av_cold int audio_read_header(AVFormatContext *s1)
00057 {
00058 AlsaData *s = s1->priv_data;
00059 AVStream *st;
00060 int ret;
00061 enum AVCodecID codec_id;
00062
00063 st = avformat_new_stream(s1, NULL);
00064 if (!st) {
00065 av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
00066
00067 return AVERROR(ENOMEM);
00068 }
00069 codec_id = s1->audio_codec_id;
00070
00071 ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
00072 &codec_id);
00073 if (ret < 0) {
00074 return AVERROR(EIO);
00075 }
00076
00077
00078 st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
00079 st->codec->codec_id = codec_id;
00080 st->codec->sample_rate = s->sample_rate;
00081 st->codec->channels = s->channels;
00082 avpriv_set_pts_info(st, 64, 1, 1000000);
00083
00084 s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate,
00085 s->period_size, 1.5E-6);
00086 if (!s->timefilter)
00087 goto fail;
00088
00089 return 0;
00090
00091 fail:
00092 snd_pcm_close(s->h);
00093 return AVERROR(EIO);
00094 }
00095
00096 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
00097 {
00098 AlsaData *s = s1->priv_data;
00099 int res;
00100 int64_t dts;
00101 snd_pcm_sframes_t delay = 0;
00102
00103 if (av_new_packet(pkt, s->period_size * s->frame_size) < 0) {
00104 return AVERROR(EIO);
00105 }
00106
00107 while ((res = snd_pcm_readi(s->h, pkt->data, s->period_size)) < 0) {
00108 if (res == -EAGAIN) {
00109 av_free_packet(pkt);
00110
00111 return AVERROR(EAGAIN);
00112 }
00113 if (ff_alsa_xrun_recover(s1, res) < 0) {
00114 av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
00115 snd_strerror(res));
00116 av_free_packet(pkt);
00117
00118 return AVERROR(EIO);
00119 }
00120 ff_timefilter_reset(s->timefilter);
00121 }
00122
00123 dts = av_gettime();
00124 snd_pcm_delay(s->h, &delay);
00125 dts -= av_rescale(delay + res, 1000000, s->sample_rate);
00126 pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period);
00127 s->last_period = res;
00128
00129 pkt->size = res * s->frame_size;
00130
00131 return 0;
00132 }
00133
00134 static const AVOption options[] = {
00135 { "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
00136 { "channels", "", offsetof(AlsaData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
00137 { NULL },
00138 };
00139
00140 static const AVClass alsa_demuxer_class = {
00141 .class_name = "ALSA demuxer",
00142 .item_name = av_default_item_name,
00143 .option = options,
00144 .version = LIBAVUTIL_VERSION_INT,
00145 };
00146
00147 AVInputFormat ff_alsa_demuxer = {
00148 .name = "alsa",
00149 .long_name = NULL_IF_CONFIG_SMALL("ALSA audio input"),
00150 .priv_data_size = sizeof(AlsaData),
00151 .read_header = audio_read_header,
00152 .read_packet = audio_read_packet,
00153 .read_close = ff_alsa_close,
00154 .flags = AVFMT_NOFILE,
00155 .priv_class = &alsa_demuxer_class,
00156 };