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;
}
static bool scene_audio_render(void *data, uint64_t *ts_out,
struct obs_source_audio_mix *audio_output, uint32_t mixers,
size_t channels, size_t sample_rate)
{
uint64_t timestamp = 0;
float *buf = NULL;
struct obs_source_audio_mix child_audio;
struct obs_scene *scene = data;
struct obs_scene_item *item;
audio_lock(scene);
item = scene->first_item;
while (item) {
if (!obs_source_audio_pending(item->source)) {
uint64_t source_ts =
obs_source_get_audio_timestamp(item->source);
if (!timestamp || source_ts < timestamp)
timestamp = source_ts;
}
item = item->next;
}
if (!timestamp) {
/* just process all pending audio actions if no audio playing,
* otherwise audio actions will just never be processed */
item = scene->first_item;
while (item) {
process_all_audio_actions(item, sample_rate);
item = item->next;
}
audio_unlock(scene);
return false;
}
item = scene->first_item;
while (item) {
uint64_t source_ts;
size_t pos, count;
bool apply_buf;
apply_buf = apply_scene_item_volume(item, &buf, timestamp,
sample_rate);
if (obs_source_audio_pending(item->source)) {
item = item->next;
continue;
}
source_ts = obs_source_get_audio_timestamp(item->source);
pos = (size_t)ns_to_audio_frames(sample_rate,
source_ts - timestamp);
count = AUDIO_OUTPUT_FRAMES - pos;
if (!apply_buf && !item->visible) {
item = item->next;
continue;
}
obs_source_get_audio_mix(item->source, &child_audio);
for (size_t mix = 0; mix < MAX_AUDIO_MIXES; mix++) {
if ((mixers & (1 << mix)) == 0)
continue;
for (size_t ch = 0; ch < channels; ch++) {
float *out = audio_output->output[mix].data[ch];
float *in = child_audio.output[mix].data[ch];
if (apply_buf)
mix_audio_with_buf(out, in, buf, pos,
count);
else
mix_audio(out, in, pos, count);
}
}
item = item->next;
}
*ts_out = timestamp;
audio_unlock(scene);
free(buf);
return true;
}
bool audio_callback(void *param,
uint64_t start_ts_in, uint64_t end_ts_in, uint64_t *out_ts,
uint32_t mixers, struct audio_output_data *mixes)
{
struct obs_core_data *data = &obs->data;
struct obs_core_audio *audio = &obs->audio;
struct obs_source *source;
size_t sample_rate = audio_output_get_sample_rate(audio->audio);
size_t channels = audio_output_get_channels(audio->audio);
struct ts_info ts = {start_ts_in, end_ts_in};
size_t audio_size;
uint64_t min_ts;
da_resize(audio->render_order, 0);
da_resize(audio->root_nodes, 0);
circlebuf_push_back(&audio->buffered_timestamps, &ts, sizeof(ts));
circlebuf_peek_front(&audio->buffered_timestamps, &ts, sizeof(ts));
min_ts = ts.start;
audio_size = AUDIO_OUTPUT_FRAMES * sizeof(float);
#if DEBUG_AUDIO == 1
blog(LOG_DEBUG, "ts %llu-%llu", ts.start, ts.end);
#endif
/* ------------------------------------------------ */
/* build audio render order
* NOTE: these are source channels, not audio channels */
for (uint32_t i = 0; i < MAX_CHANNELS; i++) {
obs_source_t *source = obs_get_output_source(i);
if (source) {
obs_source_enum_active_tree(source, push_audio_tree,
audio);
push_audio_tree(NULL, source, audio);
da_push_back(audio->root_nodes, &source);
obs_source_release(source);
}
}
pthread_mutex_lock(&data->audio_sources_mutex);
source = data->first_audio_source;
while (source) {
push_audio_tree(NULL, source, audio);
source = (struct obs_source*)source->next_audio_source;
}
pthread_mutex_unlock(&data->audio_sources_mutex);
/* ------------------------------------------------ */
/* render audio data */
for (size_t i = 0; i < audio->render_order.num; i++) {
obs_source_t *source = audio->render_order.array[i];
obs_source_audio_render(source, mixers, channels, sample_rate,
audio_size);
}
/* ------------------------------------------------ */
/* get minimum audio timestamp */
pthread_mutex_lock(&data->audio_sources_mutex);
calc_min_ts(data, sample_rate, &min_ts);
pthread_mutex_unlock(&data->audio_sources_mutex);
/* ------------------------------------------------ */
/* if a source has gone backward in time, buffer */
if (min_ts < ts.start)
add_audio_buffering(audio, sample_rate, &ts, min_ts);
/* ------------------------------------------------ */
/* mix audio */
if (!audio->buffering_wait_ticks) {
for (size_t i = 0; i < audio->root_nodes.num; i++) {
obs_source_t *source = audio->root_nodes.array[i];
if (source->audio_pending)
continue;
pthread_mutex_lock(&source->audio_buf_mutex);
if (source->audio_output_buf[0][0] && source->audio_ts)
mix_audio(mixes, source, channels, sample_rate,
&ts);
pthread_mutex_unlock(&source->audio_buf_mutex);
}
}
/* ------------------------------------------------ */
/* discard audio */
pthread_mutex_lock(&data->audio_sources_mutex);
source = data->first_audio_source;
while (source) {
pthread_mutex_lock(&source->audio_buf_mutex);
discard_audio(audio, source, channels, sample_rate, &ts);
pthread_mutex_unlock(&source->audio_buf_mutex);
source = (struct obs_source*)source->next_audio_source;
}
pthread_mutex_unlock(&data->audio_sources_mutex);
/* ------------------------------------------------ */
/* release audio sources */
release_audio_sources(audio);
circlebuf_pop_front(&audio->buffered_timestamps, NULL, sizeof(ts));
*out_ts = ts.start;
if (audio->buffering_wait_ticks) {
audio->buffering_wait_ticks--;
return false;
}
UNUSED_PARAMETER(param);
return true;
}
