static int resample_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
{
- // Get the properties of the a frame
+ // Get the properties of the frame
mlt_properties properties = mlt_frame_properties( frame );
- int output_rate = mlt_properties_get_int( properties, "resample.frequency" );
- SRC_STATE *state = mlt_properties_get_data( properties, "resample.state", NULL );
- SRC_DATA data;
- float *input_buffer = mlt_properties_get_data( properties, "resample.input_buffer", NULL );
- float *output_buffer = mlt_properties_get_data( properties, "resample.output_buffer", NULL );
+
+ // Get the filter service
+ mlt_filter filter = mlt_frame_pop_audio( frame );
+
+ // Get the filter properties
+ mlt_properties filter_properties = mlt_filter_properties( filter );
+
+ // Get the resample information
+ int output_rate = mlt_properties_get_int( filter_properties, "frequency" );
+ SRC_STATE *state = mlt_properties_get_data( filter_properties, "state", NULL );
+ float *input_buffer = mlt_properties_get_data( filter_properties, "input_buffer", NULL );
+ float *output_buffer = mlt_properties_get_data( filter_properties, "output_buffer", NULL );
int channels_avail = *channels;
+ SRC_DATA data;
int i;
+ // If no resample frequency is specified, default to requested value
if ( output_rate == 0 )
output_rate = *frequency;
// Restore the original get_audio
- frame->get_audio = mlt_properties_get_data( properties, "resample.get_audio", NULL );
+ frame->get_audio = mlt_frame_pop_audio( frame );
// Get the producer's audio
mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
{
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++ )
{
- int j, k = 0;
for ( j = 0; j < *channels; j++ )
{
new_buffer[ ( i * *channels ) + j ] = (*buffer)[ ( i * channels_avail ) + k ];
*buffer = new_buffer;
}
- else if ( channels_avail > *channels )
+ 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 );
}
// Return now if no work to do
- if ( output_rate == *frequency )
- return 0;
-
- //fprintf( stderr, "resample_get_audio: input_rate %d output_rate %d\n", *frequency, output_rate );
-
- // Convert to floating point
- for ( i = 0; i < *samples * *channels; ++i )
- input_buffer[ i ] = ( float )( (*buffer)[ i ] ) / 32768;
-
- // Resample
- data.data_in = input_buffer;
- data.data_out = output_buffer;
- data.src_ratio = ( float ) output_rate / ( float ) *frequency;
- data.input_frames = *samples;
- data.output_frames = BUFFER_LEN / *channels;
- data.end_of_input = 0;
- i = src_process( state, &data );
- if ( i == 0 )
+ if ( output_rate != *frequency )
{
- if ( data.output_frames_gen > *samples )
- {
- *buffer = mlt_pool_alloc( data.output_frames_gen * *channels * sizeof( int16_t ) );
- mlt_properties_set_data( properties, "audio", *buffer, *channels * data.output_frames_gen * 2, mlt_pool_release, NULL );
- }
- *samples = data.output_frames_gen;
- *frequency = output_rate;
-
- // Convert from floating back to signed 16bit
- for ( i = 0; i < *samples * *channels; ++i )
+ float *p = input_buffer;
+ float *end = p + *samples * *channels;
+ int16_t *q = *buffer;
+
+ // Convert to floating point
+ while( p != end )
+ *p ++ = ( float )( *q ++ ) / 32768.0;
+
+ // Resample
+ data.data_in = input_buffer;
+ data.data_out = output_buffer;
+ data.src_ratio = ( float ) output_rate / ( float ) *frequency;
+ data.input_frames = *samples;
+ data.output_frames = BUFFER_LEN / *channels;
+ data.end_of_input = 0;
+ i = src_process( state, &data );
+ if ( i == 0 )
{
- float sample = output_buffer[ i ];
- if ( sample > 1.0 )
- sample = 1.0;
- if ( sample < -1.0 )
- sample = -1.0;
- if ( sample >= 0 )
- (*buffer)[ i ] = lrint( 32767.0 * sample );
- else
- (*buffer)[ i ] = lrint( 32768.0 * sample );
+ if ( data.output_frames_gen > *samples )
+ {
+ *buffer = mlt_pool_realloc( *buffer, data.output_frames_gen * *channels * sizeof( int16_t ) );
+ mlt_properties_set_data( properties, "audio", *buffer, *channels * data.output_frames_gen * 2, mlt_pool_release, NULL );
+ }
+
+ *samples = data.output_frames_gen;
+ *frequency = output_rate;
+
+ p = output_buffer;
+ q = *buffer;
+ end = p + *samples * *channels;
+
+ // Convert from floating back to signed 16bit
+ while( p != end )
+ {
+ if ( *p > 1.0 )
+ *p = 1.0;
+ if ( *p < -1.0 )
+ *p = -1.0;
+ if ( *p > 0 )
+ *q ++ = 32767 * *p ++;
+ else
+ *q ++ = 32768 * *p ++;
+ }
}
+ else
+ fprintf( stderr, "resample_get_audio: %s %d,%d,%d\n", src_strerror( i ), *frequency, *samples, output_rate );
}
- else
- fprintf( stderr, "resample_get_audio: %s %d,%d,%d\n", src_strerror( i ), *frequency, *samples, output_rate );
-
+
return 0;
}
{
if ( frame->get_audio != NULL )
{
- mlt_properties properties = mlt_filter_properties( this );
- mlt_properties frame_props = mlt_frame_properties( frame );
-
- // Propogate the frequency property if supplied
- if ( mlt_properties_get( properties, "frequency" ) != NULL )
- mlt_properties_set_int( frame_props, "resample.frequency", mlt_properties_get_int( properties, "frequency" ) );
-
- // Propogate the other properties
- mlt_properties_set_int( frame_props, "resample.channels", mlt_properties_get_int( properties, "channels" ) );
- mlt_properties_set_data( frame_props, "resample.state", mlt_properties_get_data( properties, "state", NULL ), 0, NULL, NULL );
- mlt_properties_set_data( frame_props, "resample.input_buffer", mlt_properties_get_data( properties, "input_buffer", NULL ), 0, NULL, NULL );
- mlt_properties_set_data( frame_props, "resample.output_buffer", mlt_properties_get_data( properties, "output_buffer", NULL ), 0, NULL, NULL );
-
- // Backup the original get_audio (it's still needed)
- mlt_properties_set_data( frame_props, "resample.get_audio", frame->get_audio, 0, NULL, NULL );
-
- // Override the get_audio method
+ mlt_frame_push_audio( frame, frame->get_audio );
+ mlt_frame_push_audio( frame, this );
frame->get_audio = resample_get_audio;
}
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