static mlt_properties find_default_streams( mlt_properties meta_media, AVFormatContext *context, int *audio_index, int *video_index )
{
int i;
char key[200];
mlt_properties_set_int( meta_media, "meta.media.nb_streams", context->nb_streams );
// Allow for multiple audio and video streams in the file and select first of each (if available)
for( i = 0; i < context->nb_streams; i++ )
{
// Get the codec context
AVStream *stream = context->streams[ i ];
if ( ! stream ) continue;
AVCodecContext *codec_context = stream->codec;
if ( ! codec_context ) continue;
AVCodec *codec = avcodec_find_decoder( codec_context->codec_id );
if ( ! codec ) continue;
snprintf( key, sizeof(key), "meta.media.%d.stream.type", i );
// Determine the type and obtain the first index of each type
switch( codec_context->codec_type )
{
case CODEC_TYPE_VIDEO:
if ( *video_index < 0 )
*video_index = i;
mlt_properties_set( meta_media, key, "video" );
snprintf( key, sizeof(key), "meta.media.%d.stream.frame_rate", i );
mlt_properties_set_double( meta_media, key, av_q2d( context->streams[ i ]->r_frame_rate ) );
#if LIBAVFORMAT_VERSION_INT >= ((52<<16)+(21<<8)+0)
snprintf( key, sizeof(key), "meta.media.%d.stream.sample_aspect_ratio", i );
mlt_properties_set_double( meta_media, key, av_q2d( context->streams[ i ]->sample_aspect_ratio ) );
#endif
snprintf( key, sizeof(key), "meta.media.%d.codec.pix_fmt", i );
mlt_properties_set( meta_media, key, avcodec_get_pix_fmt_name( codec_context->pix_fmt ) );
snprintf( key, sizeof(key), "meta.media.%d.codec.sample_aspect_ratio", i );
mlt_properties_set_double( meta_media, key, av_q2d( codec_context->sample_aspect_ratio ) );
break;
case CODEC_TYPE_AUDIO:
if ( *audio_index < 0 )
*audio_index = i;
mlt_properties_set( meta_media, key, "audio" );
#if (LIBAVCODEC_VERSION_INT >= ((51<<16)+(71<<8)+0))
snprintf( key, sizeof(key), "meta.media.%d.codec.sample_fmt", i );
mlt_properties_set( meta_media, key, avcodec_get_sample_fmt_name( codec_context->sample_fmt ) );
#endif
snprintf( key, sizeof(key), "meta.media.%d.codec.sample_rate", i );
mlt_properties_set_int( meta_media, key, codec_context->sample_rate );
snprintf( key, sizeof(key), "meta.media.%d.codec.channels", i );
mlt_properties_set_int( meta_media, key, codec_context->channels );
break;
default:
break;
}
// snprintf( key, sizeof(key), "meta.media.%d.stream.time_base", i );
// mlt_properties_set_double( meta_media, key, av_q2d( context->streams[ i ]->time_base ) );
snprintf( key, sizeof(key), "meta.media.%d.codec.name", i );
mlt_properties_set( meta_media, key, codec->name );
#if (LIBAVCODEC_VERSION_INT >= ((51<<16)+(55<<8)+0))
snprintf( key, sizeof(key), "meta.media.%d.codec.long_name", i );
mlt_properties_set( meta_media, key, codec->long_name );
#endif
snprintf( key, sizeof(key), "meta.media.%d.codec.bit_rate", i );
mlt_properties_set_int( meta_media, key, codec_context->bit_rate );
// snprintf( key, sizeof(key), "meta.media.%d.codec.time_base", i );
// mlt_properties_set_double( meta_media, key, av_q2d( codec_context->time_base ) );
snprintf( key, sizeof(key), "meta.media.%d.codec.profile", i );
mlt_properties_set_int( meta_media, key, codec_context->profile );
snprintf( key, sizeof(key), "meta.media.%d.codec.level", i );
mlt_properties_set_int( meta_media, key, codec_context->level );
}
return meta_media;
}
/**
* Gets the AVPacket stored in _packet, and decodes all the samples it can from it,
* putting them in _buffer, the total number of bytes written begin stored in _dataSize.
*/
int AudioLoader::decodePacket() {
/*
E_DEBUG(EAlgorithm, "-----------------------------------------------------");
E_DEBUG(EAlgorithm, "decoding packet of " << _packet.size << " bytes");
E_DEBUG(EAlgorithm, "pts: " << _packet.pts << " - dts: " << _packet.dts); //" - pos: " << pkt->pos);
E_DEBUG(EAlgorithm, "flags: " << _packet.flags);
E_DEBUG(EAlgorithm, "duration: " << _packet.duration);
*/
int len = 0;
// buff is an offset in our output buffer, it points to where we should start
// writing the next decoded samples
int16_t* buff = _buffer;
#if !HAVE_SWRESAMPLE
if (_audioConvert) { buff = _buff1; }
#endif
// _dataSize gets the size of the buffer, in bytes
_dataSize = FFMPEG_BUFFER_SIZE*sizeof(int16_t);
// _dataSize input = number of bytes available for write in buff
// output = number of bytes actually written (actual: S16 data)
//E_DEBUG(EAlgorithm, "decode_audio_frame, available bytes in buffer = " << _dataSize);
len = decode_audio_frame(_audioCtx, buff, &_dataSize, &_packet);
if (len < 0) {
// only print error msg when file is not an mp3, because mp3 streams can have tag
// frames (id3v2?) which libavcodec tries to read as audio anyway, and we don't want
// to print an error message for that...
if (_audioCtx->codec_id == CODEC_ID_MP3) {
E_DEBUG(EAlgorithm, "AudioLoader: invalid frame, probably an mp3 tag frame, skipping it");
}
else {
E_WARNING("AudioLoader: error while decoding, skipping frame");
}
return 0;
}
if (_dataSize <= 0) {
// No data yet, get more frames
//cout << "no data yet, get more frames" << endl;
_dataSize = 0;
return 0;
}
#if !HAVE_SWRESAMPLE
if (_audioConvert) {
// this assumes that all audio is interleaved in the first channel
// it works as we're only doing sample format conversion, but we
// should be very careful
const void* ibuf[6] = { buff };
void* obuf[6] = { _buff2 };
int istride[6] = { av_get_bytes_per_sample(_audioCtx->sample_fmt) };
int ostride[6] = { av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) };
int totalsamples = _dataSize / istride[0]; // == num_samp_per_channel * num_channels
if (av_audio_convert(_audioConvert, obuf, ostride, ibuf, istride, totalsamples) < 0) {
ostringstream msg;
msg << "AudioLoader: Error converting "
<< " from " << avcodec_get_sample_fmt_name(_audioCtx->sample_fmt)
<< " to " << avcodec_get_sample_fmt_name(SAMPLE_FMT_S16);
throw EssentiaException(msg);
}
// when entering the current block, dataSize contained the size in bytes
// that the audio was taking in its native format. Now it needs to be set
// to the size of the audio we're returning, after conversion
_dataSize = totalsamples * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16);
memcpy(_buffer, _buff2, _dataSize);
}
#endif
if (len != _packet.size) {
// FIXME: investigate why this happens and whether it is a big issue
// (looks like it only loses silent samples at the end of files)
// more than 1 frame in a packet, happens a lot with flac for instance...
E_WARNING("AudioLoader: more than 1 frame in packet, dropping remaining bytes...");
E_WARNING("at sample index: " << output("audio").totalProduced());
E_WARNING("decoded samples: " << len);
E_WARNING("packet size: " << _packet.size);
}
return len;
}
