X-Git-Url: http://research.m1stereo.tv/gitweb?a=blobdiff_plain;f=src%2Fmodules%2Favformat%2Fproducer_avformat.c;h=0f3a5597e5ca65c0c48beebb637ce94fe4b060b0;hb=1d3ab5568258841b1741b8c2fc88ce4e958e6fc6;hp=773632ab94523b9bff351d702c463b8fa5cd2d6a;hpb=b49b8059af440b18c427842272f57808fc465c4f;p=melted

diff --git a/src/modules/avformat/producer_avformat.c b/src/modules/avformat/producer_avformat.c
index 773632a..0f3a559 100644
--- a/src/modules/avformat/producer_avformat.c
+++ b/src/modules/avformat/producer_avformat.c
@@ -41,6 +41,7 @@ static int producer_get_frame( mlt_producer this, mlt_frame_ptr frame, int index
 static int avformat_initialised = 0;
 static pthread_mutex_t avformat_mutex;
 
+#if 0
 void *av_malloc( unsigned int size )
 {
 	return mlt_pool_alloc( size );
@@ -55,6 +56,7 @@ void av_free( void *ptr )
 {
 	return mlt_pool_release( ptr );
 }
+#endif
 
 /** Constructor for libavformat.
 */
@@ -230,7 +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;
@@ -297,7 +299,7 @@ static int producer_open( mlt_producer this, char *file )
 			// Store selected audio and video indexes on properties
 			mlt_properties_set_int( properties, "audio_index", audio_index );
 			mlt_properties_set_int( properties, "video_index", video_index );
-
+			
 			// We're going to cheat here - for a/v files, we will have two contexts (reasoning will be clear later)
 			if ( audio_index != -1 && video_index != -1 )
 			{
@@ -377,7 +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 ];
@@ -490,14 +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;
 
@@ -763,7 +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 );
@@ -771,6 +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 )
@@ -832,7 +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
@@ -849,9 +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 )