#include <framework/mlt_filter.h>
#include <framework/mlt_frame.h>
+#include <framework/mlt_log.h>
#include <stdio.h>
#include <stdlib.h>
// ffmpeg Header files
#include <libavformat/avformat.h>
+#if LIBAVUTIL_VERSION_INT >= ((50<<16)+(38<<8)+0)
+# include <libavutil/samplefmt.h>
+#else
+# define AV_SAMPLE_FMT_S16 SAMPLE_FMT_S16
+#endif
/** Get the audio.
*/
static int resample_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
{
- // Get the properties of the frame
- mlt_properties properties = MLT_FRAME_PROPERTIES( frame );
-
// Get the filter service
mlt_filter filter = mlt_frame_pop_audio( frame );
// Obtain the resample context if it exists
ReSampleContext *resample = mlt_properties_get_data( filter_properties, "audio_resample", NULL );
- // Used to return number of channels in the source
- int channels_avail = *channels;
-
- // Loop variable
- int i;
-
// If no resample frequency is specified, default to requested value
if ( output_rate == 0 )
output_rate = *frequency;
// Get the producer's audio
- *format = mlt_audio_s16;
- mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
-
- // Duplicate channels as necessary
- if ( channels_avail < *channels )
- {
- int size = *channels * *samples * sizeof( int16_t );
- int16_t *new_buffer = mlt_pool_alloc( size );
- int j, k = 0;
-
- // Duplicate the existing channels
- for ( i = 0; i < *samples; i++ )
- {
- for ( j = 0; j < *channels; j++ )
- {
- new_buffer[ ( i * *channels ) + j ] = ((int16_t*)(*buffer))[ ( i * channels_avail ) + k ];
- k = ( k + 1 ) % channels_avail;
- }
- }
-
- // Update the audio buffer now - destroys the old
- mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
-
- *buffer = new_buffer;
- }
- else if ( channels_avail == 6 && *channels == 2 )
- {
- // Nasty hack for ac3 5.1 audio - may be a cause of failure?
- int size = *channels * *samples * sizeof( int16_t );
- int16_t *new_buffer = mlt_pool_alloc( size );
-
- // Drop all but the first *channels
- for ( i = 0; i < *samples; i++ )
- {
- new_buffer[ ( i * *channels ) + 0 ] = ((int16_t*)(*buffer))[ ( i * channels_avail ) + 2 ];
- new_buffer[ ( i * *channels ) + 1 ] = ((int16_t*)(*buffer))[ ( i * channels_avail ) + 3 ];
- }
-
- // Update the audio buffer now - destroys the old
- mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
-
- *buffer = new_buffer;
- }
+ int error = mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
+ if ( error ) return error;
// Return now if no work to do
if ( output_rate != *frequency )
// Will store number of samples created
int used = 0;
+ mlt_log_debug( MLT_FILTER_SERVICE(filter), "channels %d samples %d frequency %d -> %d\n",
+ *channels, *samples, *frequency, output_rate );
+
+ // Do not convert to s16 unless we need to change the rate
+ if ( *format != mlt_audio_s16 )
+ {
+ *format = mlt_audio_s16;
+ mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
+ }
+
// Create a resampler if nececessary
if ( resample == NULL || *frequency != mlt_properties_get_int( filter_properties, "last_frequency" ) )
{
// Create the resampler
#if (LIBAVCODEC_VERSION_INT >= ((52<<16)+(15<<8)+0))
resample = av_audio_resample_init( *channels, *channels, output_rate, *frequency,
- SAMPLE_FMT_S16, SAMPLE_FMT_S16, 16, 10, 0, 0.8 );
+ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16, 16, 10, 0, 0.8 );
#else
resample = audio_resample_init( *channels, *channels, output_rate, *frequency );
#endif
mlt_service_unlock( MLT_FILTER_SERVICE( filter ) );
}
- return 0;
+ return error;
}
/** Filter processing.
*/
-static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
+static mlt_frame filter_process( mlt_filter filter, mlt_frame frame )
{
// Only call this if we have a means to get audio
if ( mlt_frame_is_test_audio( frame ) == 0 )
{
// Push the filter on to the stack
- mlt_frame_push_audio( frame, this );
+ mlt_frame_push_audio( frame, filter );
// Assign our get_audio method
mlt_frame_push_audio( frame, resample_get_audio );
mlt_filter filter_avresample_init( char *arg )
{
// Create a filter
- mlt_filter this = mlt_filter_new( );
+ mlt_filter filter = mlt_filter_new( );
// Initialise if successful
- if ( this != NULL )
+ if ( filter != NULL )
{
// Calculate size of the buffer
int size = AVCODEC_MAX_AUDIO_FRAME_SIZE * sizeof( int16_t );
int16_t *buffer = mlt_pool_alloc( size );
// Assign the process method
- this->process = filter_process;
+ filter->process = filter_process;
// Deal with argument
if ( arg != NULL )
- mlt_properties_set( MLT_FILTER_PROPERTIES( this ), "frequency", arg );
+ mlt_properties_set( MLT_FILTER_PROPERTIES( filter ), "frequency", arg );
// Default to 2 channel output
- mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "channels", 2 );
+ mlt_properties_set_int( MLT_FILTER_PROPERTIES( filter ), "channels", 2 );
// Store the buffer
- mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "buffer", buffer, size, mlt_pool_release, NULL );
+ mlt_properties_set_data( MLT_FILTER_PROPERTIES( filter ), "buffer", buffer, size, mlt_pool_release, NULL );
}
- return this;
+ return filter;
}
projects / mlt / blobdiff