static void rtmp_stream_data(void *data, struct encoder_packet *packet)
{
struct rtmp_stream *stream = data;
struct encoder_packet new_packet;
bool added_packet = false;
if (disconnected(stream))
return;
if (packet->type == OBS_ENCODER_VIDEO)
obs_parse_avc_packet(&new_packet, packet);
else
obs_duplicate_encoder_packet(&new_packet, packet);
pthread_mutex_lock(&stream->packets_mutex);
if (!disconnected(stream)) {
added_packet = (packet->type == OBS_ENCODER_VIDEO) ?
add_video_packet(stream, &new_packet) :
add_packet(stream, &new_packet);
}
pthread_mutex_unlock(&stream->packets_mutex);
if (added_packet)
os_sem_post(stream->send_sem);
else
obs_free_encoder_packet(&new_packet);
}
static void interleave_packets(void *data, struct encoder_packet *packet)
{
struct obs_output *output = data;
struct encoder_packet out;
bool was_started;
pthread_mutex_lock(&output->interleaved_mutex);
was_started = output->received_audio && output->received_video;
obs_duplicate_encoder_packet(&out, packet);
apply_interleaved_packet_offset(output, &out);
insert_interleaved_packet(output, &out);
set_higher_ts(output, &out);
/* when both video and audio have been received, we're ready
* to start sending out packets (one at a time) */
if (output->received_audio && output->received_video) {
if (!was_started) {
prune_interleaved_packets(output);
initialize_interleaved_packets(output);
resort_interleaved_packets(output);
}
send_interleaved(output);
}
pthread_mutex_unlock(&output->interleaved_mutex);
}
static void interleave_packets(void *data, struct encoder_packet *packet)
{
struct obs_output *output = data;
struct encoder_packet out;
bool was_started;
if (!active(output))
return;
if (packet->type == OBS_ENCODER_AUDIO)
packet->track_idx = get_track_index(output, packet);
pthread_mutex_lock(&output->interleaved_mutex);
/* if first video frame is not a keyframe, discard until received */
if (!output->received_video &&
packet->type == OBS_ENCODER_VIDEO &&
!packet->keyframe) {
discard_unused_audio_packets(output, packet->dts_usec);
pthread_mutex_unlock(&output->interleaved_mutex);
if (output->active_delay_ns)
obs_free_encoder_packet(packet);
return;
}
was_started = output->received_audio && output->received_video;
if (output->active_delay_ns)
out = *packet;
else
obs_duplicate_encoder_packet(&out, packet);
if (was_started)
apply_interleaved_packet_offset(output, &out);
else
check_received(output, packet);
insert_interleaved_packet(output, &out);
set_higher_ts(output, &out);
/* when both video and audio have been received, we're ready
* to start sending out packets (one at a time) */
if (output->received_audio && output->received_video) {
if (!was_started) {
if (prune_interleaved_packets(output)) {
if (initialize_interleaved_packets(output)) {
resort_interleaved_packets(output);
send_interleaved(output);
}
}
} else {
send_interleaved(output);
}
}
pthread_mutex_unlock(&output->interleaved_mutex);
}
static bool prepare_interleaved_packet(struct obs_output *output,
struct encoder_packet *out, struct encoder_packet *in)
{
int64_t offset;
/* audio and video need to start at timestamp 0, and the encoders
* may not currently be at 0 when we get data. so, we store the
* current dts as offset and subtract that value from the dts/pts
* of the output packet. */
if (in->type == OBS_ENCODER_VIDEO) {
if (!output->received_video) {
output->first_video_ts = in->dts_usec;
output->video_offset = in->dts;
output->received_video = true;
}
offset = output->video_offset;
} else{
/* don't accept audio that's before the first video timestamp */
if (!output->received_video ||
in->dts_usec < output->first_video_ts)
return false;
if (!output->received_audio) {
output->audio_offset = in->dts;
output->received_audio = true;
}
offset = output->audio_offset;
}
obs_duplicate_encoder_packet(out, in);
out->dts -= offset;
out->pts -= offset;
/* convert the newly adjusted dts to relative dts time to ensure proper
* interleaving. if we're using an audio encoder that's already been
* started on another output, then the first audio packet may not be
* quite perfectly synced up in terms of system time (and there's
* nothing we can really do about that), but it will always at least be
* within a 23ish millisecond threshold (at least for AAC) */
out->dts_usec = packet_dts_usec(out);
return true;
}
static void interleave_packets(void *data, struct encoder_packet *packet)
{
struct obs_output *output = data;
struct encoder_packet out;
bool was_started;
if (packet->type == OBS_ENCODER_AUDIO)
packet->track_idx = get_track_index(output, packet);
pthread_mutex_lock(&output->interleaved_mutex);
was_started = output->received_audio && output->received_video;
if (output->active_delay_ns)
out = *packet;
else
obs_duplicate_encoder_packet(&out, packet);
if (was_started)
apply_interleaved_packet_offset(output, &out);
else
check_received(output, packet);
insert_interleaved_packet(output, &out);
set_higher_ts(output, &out);
/* when both video and audio have been received, we're ready
* to start sending out packets (one at a time) */
if (output->received_audio && output->received_video) {
if (!was_started) {
prune_interleaved_packets(output);
if (initialize_interleaved_packets(output)) {
resort_interleaved_packets(output);
send_interleaved(output);
}
} else {
send_interleaved(output);
}
}
pthread_mutex_unlock(&output->interleaved_mutex);
}
