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RTC: Refine audio to RTP packet base.

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This commit is contained in:
winlin 2020-05-13 20:13:25 +08:00
parent 588d17c09d
commit 54d8c36905
7 changed files with 129 additions and 190 deletions
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73
trunk/src/app/srs_app_rtc_source.cpp
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@ -80,30 +80,27 @@ srs_error_t aac_raw_append_adts_header(SrsSharedPtrMessage* shared_audio, SrsFor
return err;
}
SrsRtcConsumer::SrsRtcConsumer(SrsRtcSource* s, SrsConnection* c)
SrsRtcConsumer::SrsRtcConsumer(SrsRtcSource* s)
{
source = s;
conn = c;
should_update_source_id = false;
queue = new SrsMessageQueue();
#ifdef SRS_PERF_QUEUE_COND_WAIT
mw_wait = srs_cond_new();
mw_min_msgs = 0;
mw_duration = 0;
mw_waiting = false;
#endif
}
SrsRtcConsumer::~SrsRtcConsumer()
{
source->on_consumer_destroy(this);
srs_freep(queue);
vector<SrsRtpPacket2*>::iterator it;
for (it = queue.begin(); it != queue.end(); ++it) {
SrsRtpPacket2* pkt = *it;
srs_freep(pkt);
}
#ifdef SRS_PERF_QUEUE_COND_WAIT
srs_cond_destroy(mw_wait);
#endif
}
void SrsRtcConsumer::update_source_id()
@ -114,66 +111,46 @@ void SrsRtcConsumer::update_source_id()
srs_error_t SrsRtcConsumer::enqueue(SrsSharedPtrMessage* shared_msg, bool atc, SrsRtmpJitterAlgorithm ag)
{
srs_error_t err = srs_success;
return err;
}
SrsSharedPtrMessage* msg = shared_msg->copy();
srs_error_t SrsRtcConsumer::enqueue2(SrsRtpPacket2* pkt)
{
srs_error_t err = srs_success;
if ((err = queue->enqueue(msg, NULL)) != srs_success) {
return srs_error_wrap(err, "enqueue message");
}
queue.push_back(pkt);
#ifdef SRS_PERF_QUEUE_COND_WAIT
// fire the mw when msgs is enough.
if (mw_waiting) {
if (queue->size() > mw_min_msgs) {
if ((int)queue.size() > mw_min_msgs) {
srs_cond_signal(mw_wait);
mw_waiting = false;
return err;
}
return err;
}
#endif
return err;
}
srs_error_t SrsRtcConsumer::dump_packets(SrsMessageArray* msgs, int& count)
srs_error_t SrsRtcConsumer::dump_packets(std::vector<SrsRtpPacket2*>& pkts)
{
srs_error_t err = srs_success;
srs_assert(count >= 0);
srs_assert(msgs->max > 0);
// the count used as input to reset the max if positive.
int max = count? srs_min(count, msgs->max) : msgs->max;
// the count specifies the max acceptable count,
// here maybe 1+, and we must set to 0 when got nothing.
count = 0;
if (should_update_source_id) {
srs_trace("update source_id=%d[%d]", source->source_id(), source->source_id());
should_update_source_id = false;
}
// pump msgs from queue.
if ((err = queue->dump_packets(max, msgs->msgs, count)) != srs_success) {
return srs_error_wrap(err, "dump packets");
}
queue.swap(pkts);
return err;
}
#ifdef SRS_PERF_QUEUE_COND_WAIT
void SrsRtcConsumer::wait(int nb_msgs, srs_utime_t msgs_duration)
void SrsRtcConsumer::wait(int nb_msgs)
{
mw_min_msgs = nb_msgs;
mw_duration = msgs_duration;
srs_utime_t duration = queue->duration();
bool match_min_msgs = queue->size() > mw_min_msgs;
// when duration ok, signal to flush.
if (match_min_msgs && duration > mw_duration) {
if ((int)queue.size() > mw_min_msgs) {
return;
}
@ -183,7 +160,6 @@ void SrsRtcConsumer::wait(int nb_msgs, srs_utime_t msgs_duration)
// use cond block wait for high performance mode.
srs_cond_wait(mw_wait);
}
#endif
SrsRtcSourceManager::SrsRtcSourceManager()
{
@ -347,11 +323,11 @@ SrsMetaCache* SrsRtcSource::cached_meta()
return meta;
}
srs_error_t SrsRtcSource::create_consumer(SrsConnection* conn, SrsRtcConsumer*& consumer)
srs_error_t SrsRtcSource::create_consumer(SrsRtcConsumer*& consumer)
{
srs_error_t err = srs_success;
consumer = new SrsRtcConsumer(this, conn);
consumer = new SrsRtcConsumer(this);
consumers.push_back(consumer);
// TODO: FIXME: Implements edge cluster.
@ -441,7 +417,6 @@ srs_error_t SrsRtcSource::on_audio_imp(SrsSharedPtrMessage* msg)
{
srs_error_t err = srs_success;
// copy to all consumer
for (int i = 0; i < (int)consumers.size(); i++) {
SrsRtcConsumer* consumer = consumers.at(i);
if ((err = consumer->enqueue(msg, true, SrsRtmpJitterAlgorithmOFF)) != srs_success) {
@ -455,6 +430,14 @@ srs_error_t SrsRtcSource::on_audio_imp(SrsSharedPtrMessage* msg)
srs_error_t SrsRtcSource::on_audio2(SrsRtpPacket2* pkt)
{
srs_error_t err = srs_success;
for (int i = 0; i < (int)consumers.size(); i++) {
SrsRtcConsumer* consumer = consumers.at(i);
if ((err = consumer->enqueue2(pkt)) != srs_success) {
return srs_error_wrap(err, "consume message");
}
}
return err;
}
@ -686,7 +669,7 @@ srs_error_t SrsRtcFromRtmpBridger::transcode(char* adts_audio, int nn_adts_audio
nn_max_extra_payload = srs_max(nn_max_extra_payload, p->size);
SrsRtpPacket2* packet = new SrsRtpPacket2();
packet->rtp_header.set_marker(true);
packet->frame_type = SrsFrameTypeAudio;
SrsRtpRawPayload* raw = packet->reuse_raw();
raw->payload = new char[p->size];