static int transition_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
{
// Get the b frame from the stack
mlt_frame b_frame = mlt_frame_pop_audio( frame );
// Get the effect
mlt_transition effect = mlt_frame_pop_audio( frame );
// Get the properties of the b frame
mlt_properties b_props = MLT_FRAME_PROPERTIES( b_frame );
// We can only mix s16
*format = mlt_audio_s16;
if ( mlt_properties_get_int( MLT_TRANSITION_PROPERTIES( effect ), "combine" ) == 0 )
{
double mix_start = 0.5, mix_end = 0.5;
if ( mlt_properties_get( b_props, "audio.previous_mix" ) != NULL )
mix_start = mlt_properties_get_double( b_props, "audio.previous_mix" );
if ( mlt_properties_get( b_props, "audio.mix" ) != NULL )
mix_end = mlt_properties_get_double( b_props, "audio.mix" );
if ( mlt_properties_get_int( b_props, "audio.reverse" ) )
{
mix_start = 1 - mix_start;
mix_end = 1 - mix_end;
}
mix_audio( frame, b_frame, mix_start, mix_end, buffer, format, frequency, channels, samples );
}
else
{
combine_audio( frame, b_frame, buffer, format, frequency, channels, samples );
}
return 0;
}
static int transition_get_audio( mlt_frame frame_a, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
{
int error = 0;
// Get the b frame from the stack
mlt_frame frame_b = mlt_frame_pop_audio( frame_a );
// Get the effect
mlt_transition transition = mlt_frame_pop_audio( frame_a );
// Get the properties of the b frame
mlt_properties b_props = MLT_FRAME_PROPERTIES( frame_b );
transition_mix self = transition->child;
int16_t *buffer_b, *buffer_a;
int frequency_b = *frequency, frequency_a = *frequency;
int channels_b = *channels, channels_a = *channels;
int samples_b = *samples, samples_a = *samples;
// We can only mix s16
*format = mlt_audio_s16;
mlt_frame_get_audio( frame_b, (void**) &buffer_b, format, &frequency_b, &channels_b, &samples_b );
mlt_frame_get_audio( frame_a, (void**) &buffer_a, format, &frequency_a, &channels_a, &samples_a );
if ( buffer_b == buffer_a )
{
*samples = samples_b;
*channels = channels_b;
*buffer = buffer_b;
*frequency = frequency_b;
return error;
}
int silent = mlt_properties_get_int( MLT_FRAME_PROPERTIES( frame_a ), "silent_audio" );
mlt_properties_set_int( MLT_FRAME_PROPERTIES( frame_a ), "silent_audio", 0 );
if ( silent )
memset( buffer_a, 0, samples_a * channels_a * sizeof( int16_t ) );
silent = mlt_properties_get_int( b_props, "silent_audio" );
mlt_properties_set_int( b_props, "silent_audio", 0 );
if ( silent )
memset( buffer_b, 0, samples_b * channels_b * sizeof( int16_t ) );
// determine number of samples to process
*samples = MIN( self->src_buffer_count + samples_b, self->dest_buffer_count + samples_a );
*channels = MIN( MIN( channels_b, channels_a ), MAX_CHANNELS );
*frequency = frequency_a;
// Prevent src buffer overflow by discarding oldest samples.
samples_b = MIN( samples_b, MAX_SAMPLES * MAX_CHANNELS / channels_b );
size_t bytes = PCM16_BYTES( samples_b, channels_b );
if ( PCM16_BYTES( self->src_buffer_count + samples_b, channels_b ) > MAX_BYTES ) {
mlt_log_verbose( MLT_TRANSITION_SERVICE(transition), "buffer overflow: src_buffer_count %d\n",
self->src_buffer_count );
self->src_buffer_count = MAX_SAMPLES * MAX_CHANNELS / channels_b - samples_b;
memmove( self->src_buffer, &self->src_buffer[MAX_SAMPLES * MAX_CHANNELS - samples_b * channels_b],
PCM16_BYTES( samples_b, channels_b ) );
}
// Buffer new src samples.
memcpy( &self->src_buffer[self->src_buffer_count * channels_b], buffer_b, bytes );
self->src_buffer_count += samples_b;
buffer_b = self->src_buffer;
// Prevent dest buffer overflow by discarding oldest samples.
samples_a = MIN( samples_a, MAX_SAMPLES * MAX_CHANNELS / channels_a );
bytes = PCM16_BYTES( samples_a, channels_a );
if ( PCM16_BYTES( self->dest_buffer_count + samples_a, channels_a ) > MAX_BYTES ) {
mlt_log_verbose( MLT_TRANSITION_SERVICE(transition), "buffer overflow: dest_buffer_count %d\n",
self->dest_buffer_count );
self->dest_buffer_count = MAX_SAMPLES * MAX_CHANNELS / channels_a - samples_a;
memmove( self->dest_buffer, &self->dest_buffer[MAX_SAMPLES * MAX_CHANNELS - samples_a * channels_a],
PCM16_BYTES( samples_a, channels_a ) );
}
// Buffer the new dest samples.
memcpy( &self->dest_buffer[self->dest_buffer_count * channels_a], buffer_a, bytes );
self->dest_buffer_count += samples_a;
buffer_a = self->dest_buffer;
// Do the mixing.
if ( mlt_properties_get_int( MLT_TRANSITION_PROPERTIES(transition), "combine" ) )
{
double weight = 1.0;
if ( mlt_properties_get_int( MLT_FRAME_PROPERTIES( frame_a ), "meta.mixdown" ) )
weight = 1.0 - mlt_properties_get_double( MLT_FRAME_PROPERTIES( frame_a ), "meta.volume" );
combine_audio( weight, buffer_a, buffer_b, channels_a, channels_b, *channels, *samples );
}
else
{
double mix_start = 0.5, mix_end = 0.5;
if ( mlt_properties_get( b_props, "audio.previous_mix" ) )
mix_start = mlt_properties_get_double( b_props, "audio.previous_mix" );
if ( mlt_properties_get( b_props, "audio.mix" ) )
mix_end = mlt_properties_get_double( b_props, "audio.mix" );
if ( mlt_properties_get_int( b_props, "audio.reverse" ) )
{
mix_start = 1.0 - mix_start;
mix_end = 1.0 - mix_end;
}
mix_audio( mix_start, mix_end, buffer_a, buffer_b, channels_a, channels_b, *channels, *samples );
}
// Copy the audio into the frame.
bytes = PCM16_BYTES( *samples, *channels );
*buffer = mlt_pool_alloc( bytes );
memcpy( *buffer, buffer_a, bytes );
mlt_frame_set_audio( frame_a, *buffer, *format, bytes, mlt_pool_release );
if ( mlt_properties_get_int( b_props, "_speed" ) == 0 )
{
// Flush the buffer when paused and scrubbing.
samples_b = self->src_buffer_count;
samples_a = self->dest_buffer_count;
}
else
{
// Determine the maximum amount of latency permitted in the buffer.
int max_latency = CLAMP( *frequency / 1000, 0, MAX_SAMPLES ); // samples in 1ms
// samples_b becomes the new target src buffer count.
samples_b = CLAMP( self->src_buffer_count - *samples, 0, max_latency );
// samples_b becomes the number of samples to consume: difference between actual and the target.
samples_b = self->src_buffer_count - samples_b;
// samples_a becomes the new target dest buffer count.
samples_a = CLAMP( self->dest_buffer_count - *samples, 0, max_latency );
// samples_a becomes the number of samples to consume: difference between actual and the target.
samples_a = self->dest_buffer_count - samples_a;
}
// Consume the src buffer.
self->src_buffer_count -= samples_b;
if ( self->src_buffer_count ) {
memmove( self->src_buffer, &self->src_buffer[samples_b * channels_b],
PCM16_BYTES( self->src_buffer_count, channels_b ));
}
// Consume the dest buffer.
self->dest_buffer_count -= samples_a;
if ( self->dest_buffer_count ) {
memmove( self->dest_buffer, &self->dest_buffer[samples_a * channels_a],
PCM16_BYTES( self->dest_buffer_count, channels_a ));
}
return error;
}
