X-Git-Url: http://research.m1stereo.tv/gitweb?a=blobdiff_plain;f=src%2Fmodules%2Favformat%2Fproducer_avformat.c;h=0f3a5597e5ca65c0c48beebb637ce94fe4b060b0;hb=3c425566058901f3ab8dbb858afc957e2609d935;hp=6ebd5683a63de1cc609d07329631fb72b12926b8;hpb=0d368e962076c3b96040b86cd5d87a1bbfe0a594;p=melted diff --git a/src/modules/avformat/producer_avformat.c b/src/modules/avformat/producer_avformat.c index 6ebd568..0f3a559 100644 --- a/src/modules/avformat/producer_avformat.c +++ b/src/modules/avformat/producer_avformat.c @@ -232,7 +232,7 @@ static int producer_open( mlt_producer this, char *file ) { mrl[0] = 0; char *name = strdup( ++mrl ); - char *value = strchr( name, '=' ); + char *value = strchr( name, ':' ); if ( value ) { value[0] = 0; @@ -379,7 +379,7 @@ static int producer_get_image( mlt_frame frame, uint8_t **buffer, mlt_image_form mlt_position expected = mlt_properties_get_position( properties, "video_expected" ); // Calculate the real time code - double real_timecode = producer_time_of_frame( this, position ) + mlt_properties_get_double( properties, "_v_pts_offset" ); + double real_timecode = producer_time_of_frame( this, position ); // Get the video stream AVStream *stream = context->streams[ index ]; @@ -492,14 +492,7 @@ static int producer_get_image( mlt_frame frame, uint8_t **buffer, mlt_image_form if ( got_picture ) { if ( pkt.pts != AV_NOPTS_VALUE && pkt.pts != 0 ) - { - if ( current_time == 0 ) - { - mlt_properties_set_double( properties, "_v_pts_offset", ( double )( pkt.pts / 1000000 ) ); - real_timecode += pkt.pts / 1000000; - } current_time = ( double )pkt.pts / 1000000.0; - } else current_time = real_timecode; @@ -765,7 +758,7 @@ static int producer_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_form pthread_mutex_lock( &avformat_mutex ); // Check for resample and create if necessary - if ( resample == NULL ) + if ( resample == NULL && codec_context->channels <= 2 ) { // Create the resampler resample = audio_resample_init( *channels, codec_context->channels, *frequency, codec_context->sample_rate ); @@ -773,6 +766,11 @@ static int producer_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_form // And store it on properties mlt_properties_set_data( properties, "audio_resample", resample, 0, ( mlt_destructor )audio_resample_close, NULL ); } + else if ( resample == NULL ) + { + *channels = codec_context->channels; + *frequency = codec_context->sample_rate; + } // Check for audio buffer and create if necessary if ( audio_buffer == NULL ) @@ -834,7 +832,7 @@ static int producer_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_form uint8_t *ptr = pkt.data; int data_size; - // We only deal with video from the selected video_index + // We only deal with audio from the selected audio_index while ( ptr != NULL && ret >= 0 && pkt.stream_index == index && len > 0 ) { // Decode the audio @@ -851,9 +849,15 @@ static int producer_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_form if ( data_size > 0 ) { - int size_out = audio_resample( resample, &audio_buffer[ audio_used * *channels ], temp, data_size / ( codec_context->channels * sizeof( int16_t ) ) ); - - audio_used += size_out; + if ( resample != NULL ) + { + audio_used += audio_resample( resample, &audio_buffer[ audio_used * *channels ], temp, data_size / ( codec_context->channels * sizeof( int16_t ) ) ); + } + else + { + memcpy( &audio_buffer[ audio_used * *channels ], temp, data_size ); + audio_used += data_size / ( codec_context->channels * sizeof( int16_t ) ); + } // Handle ignore while ( ignore && audio_used > *samples )