Remaining audio handling switched to stacks; Minor corrections to compositing and...
[melted] / src / modules / resample / filter_resample.c
index c8fbf5f..72df011 100644 (file)
 
 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
-       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 );
+       // 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 );
+
+       // 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;
-       float *input_buffer = mlt_properties_get_data( properties, "resample.input_buffer", NULL );
-       float *output_buffer = mlt_properties_get_data( properties, "resample.output_buffer", NULL );
        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 );
-
        // Get the producer's audio
-       mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
-
-       // 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 )
+       mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );
+
+       // Duplicate channels as necessary
+       if ( channels_avail < *channels )
        {
-               if ( data.output_frames_gen > *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++ )
                {
-                       *buffer = (int16_t*) malloc( data.output_frames_gen * *channels * 2 );
-                       mlt_properties_set_data( properties, "audio", *buffer, *channels * data.output_frames_gen * 2, free, NULL );
+                       for ( j = 0; j < *channels; j++ )
+                       {
+                               new_buffer[ ( i * *channels ) + j ] = (*buffer)[ ( i * channels_avail ) + k ];
+                               k = ( k + 1 ) % channels_avail;
+                       }
                }
-               *samples = data.output_frames_gen;
-               *frequency = output_rate;
                
-               // Convert from floating back to signed 16bit
-               for ( i = 0; i < *samples * *channels; ++i )
+               // 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++ )
                {
-                       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 );
+                       new_buffer[ ( i * *channels ) + 0 ] = (*buffer)[ ( i * channels_avail ) + 2 ];
+                       new_buffer[ ( i * *channels ) + 1 ] = (*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;
        }
-       else
-               fprintf( stderr, "resample_get_audio: %s %d,%d,%d\n", src_strerror( i ), *frequency, *samples, output_rate );
-       
+
+       // Return now if no work to do
+       if ( output_rate != *frequency )
+       {
+               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 )
+               {
+                       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 );
+       }
+
        return 0;
 }
 
@@ -101,24 +161,11 @@ static int resample_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_form
 
 static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
 {
-       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
-       frame->get_audio = resample_get_audio;
+       if ( mlt_frame_is_test_audio( frame ) != 0 )
+       {
+               mlt_frame_push_audio( frame, this );
+               mlt_frame_push_audio( frame, resample_get_audio );
+       }
 
        return frame;
 }
@@ -128,22 +175,22 @@ static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
 
 mlt_filter filter_resample_init( char *arg )
 {
-       mlt_filter this = calloc( sizeof( struct mlt_filter_s ), 1 );
-       if ( this != NULL && mlt_filter_init( this, NULL ) == 0 )
+       mlt_filter this = mlt_filter_new( );
+       if ( this != NULL )
        {
                int error;
                SRC_STATE *state = src_new( RESAMPLE_TYPE, 2 /* channels */, &error );
                if ( error == 0 )
                {
+                       void *input_buffer = mlt_pool_alloc( BUFFER_LEN );
+                       void *output_buffer = mlt_pool_alloc( BUFFER_LEN );
                        this->process = filter_process;
                        if ( arg != NULL )
-                               mlt_properties_set_int( mlt_filter_properties( this ), "frequency", atoi( arg ) );
-                       mlt_properties_set_int( mlt_filter_properties( this ), "channels", 2 );
-                       mlt_properties_set_data( mlt_filter_properties( this ), "state", state, 0, (mlt_destructor)src_delete, NULL );
-                       mlt_properties_set_data( mlt_filter_properties( this ), "input_buffer",
-                               malloc( BUFFER_LEN ), BUFFER_LEN, free, NULL );
-                       mlt_properties_set_data( mlt_filter_properties( this ), "output_buffer",
-                               malloc( BUFFER_LEN ), BUFFER_LEN, free, NULL );
+                               mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "frequency", atoi( arg ) );
+                       mlt_properties_set_int( MLT_FILTER_PROPERTIES( this ), "channels", 2 );
+                       mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "state", state, 0, (mlt_destructor)src_delete, NULL );
+                       mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "input_buffer", input_buffer, BUFFER_LEN, mlt_pool_release, NULL );
+                       mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), "output_buffer", output_buffer, BUFFER_LEN, mlt_pool_release, NULL );
                }
                else
                {
@@ -152,4 +199,3 @@ mlt_filter filter_resample_init( char *arg )
        }
        return this;
 }
-