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RTC: Move source to play stream

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This commit is contained in:
winlin 2020-08-06 13:51:12 +08:00
parent 994c53b6c1
commit b070e5aa05
2 changed files with 32 additions and 24 deletions
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53
trunk/src/app/srs_app_rtc_conn.cpp
View file

@ -200,6 +200,9 @@ SrsRtcPlayStream::SrsRtcPlayStream(SrsRtcConnection* s, SrsContextId parent_cid)
_parent_cid = parent_cid;
trd = new SrsDummyCoroutine();
req_ = NULL;
source_ = NULL;
session_ = s;
mw_msgs = 0;
@ -215,6 +218,8 @@ SrsRtcPlayStream::~SrsRtcPlayStream()
{
_srs_config->unsubscribe(this);
srs_freep(req_);
srs_freep(trd);
srs_freep(timer_);
@ -237,6 +242,12 @@ srs_error_t SrsRtcPlayStream::initialize(SrsRequest* req, std::map<uint32_t, Srs
{
srs_error_t err = srs_success;
req_ = req->copy();
if ((err = _srs_rtc_sources->fetch_or_create(req_, &source_)) != srs_success) {
return srs_error_wrap(err, "rtc fetch source failed");
}
if (true) {
std::map<uint32_t, SrsRtcTrackDescription*>::iterator it = sub_relations.begin();
while (it != sub_relations.end()) {
@ -262,14 +273,12 @@ srs_error_t SrsRtcPlayStream::initialize(SrsRequest* req, std::map<uint32_t, Srs
srs_error_t SrsRtcPlayStream::on_reload_vhost_play(string vhost)
{
SrsRequest* req = session_->req;
if (req->vhost != vhost) {
if (req_->vhost != vhost) {
return srs_success;
}
realtime = _srs_config->get_realtime_enabled(req->vhost, true);
mw_msgs = _srs_config->get_mw_msgs(req->vhost, realtime, true);
realtime = _srs_config->get_realtime_enabled(req_->vhost, true);
mw_msgs = _srs_config->get_mw_msgs(req_->vhost, realtime, true);
srs_trace("Reload play realtime=%d, mw_msgs=%d", realtime, mw_msgs);
@ -309,7 +318,7 @@ srs_error_t SrsRtcPlayStream::start()
}
if (_srs_rtc_hijacker) {
if ((err = _srs_rtc_hijacker->on_start_play(session_, this, session_->req)) != srs_success) {
if ((err = _srs_rtc_hijacker->on_start_play(session_, this, req_)) != srs_success) {
return srs_error_wrap(err, "on start play");
}
}
@ -328,30 +337,26 @@ srs_error_t SrsRtcPlayStream::cycle()
{
srs_error_t err = srs_success;
SrsRtcStream* source = NULL;
SrsRequest* req = session_->req;
if ((err = _srs_rtc_sources->fetch_or_create(req, &source)) != srs_success) {
return srs_error_wrap(err, "fetch source");
}
SrsRtcStream* source = source_;
SrsRtcConsumer* consumer = NULL;
SrsAutoFree(SrsRtcConsumer, consumer);
if ((err = source->create_consumer(consumer)) != srs_success) {
return srs_error_wrap(err, "create consumer, source=%s", req->get_stream_url().c_str());
return srs_error_wrap(err, "create consumer, source=%s", req_->get_stream_url().c_str());
}
// TODO: FIXME: Dumps the SPS/PPS from gop cache, without other frames.
if ((err = source->consumer_dumps(consumer)) != srs_success) {
return srs_error_wrap(err, "dumps consumer, url=%s", req->get_stream_url().c_str());
return srs_error_wrap(err, "dumps consumer, url=%s", req_->get_stream_url().c_str());
}
realtime = _srs_config->get_realtime_enabled(req->vhost, true);
mw_msgs = _srs_config->get_mw_msgs(req->vhost, realtime, true);
realtime = _srs_config->get_realtime_enabled(req_->vhost, true);
mw_msgs = _srs_config->get_mw_msgs(req_->vhost, realtime, true);
// TODO: FIXME: Add cost in ms.
srs_trace("RTC: start play, url=%s, source_id=[%d][%s], encrypt=%d, realtime=%d, mw_msgs=%d", req->get_stream_url().c_str(),
::getpid(), source->source_id().c_str(), session_->encrypt, realtime, mw_msgs);
SrsContextId cid = source->source_id();
srs_trace("RTC: start play url=%s, source_id=[%d][%s], encrypt=%d, realtime=%d, mw_msgs=%d", req_->get_stream_url().c_str(),
::getpid(), cid.c_str(), session_->encrypt, realtime, mw_msgs);
SrsPithyPrint* pprint = SrsPithyPrint::create_rtc_play();
SrsAutoFree(SrsPithyPrint, pprint);
@ -363,7 +368,7 @@ srs_error_t SrsRtcPlayStream::cycle()
vector<SrsRtpPacket2*> pkts;
if (_srs_rtc_hijacker) {
if ((err = _srs_rtc_hijacker->on_start_consume(session_, this, session_->req, consumer)) != srs_success) {
if ((err = _srs_rtc_hijacker->on_start_consume(session_, this, req_, consumer)) != srs_success) {
return srs_error_wrap(err, "on start consuming");
}
}
@ -738,7 +743,7 @@ srs_error_t SrsRtcPlayStream::on_rtcp_ps_feedback(char* buf, int nb_buf)
switch (fmt) {
case kPLI: {
ISrsRtcPublishStream* publisher = session_->source_->publish_stream();
ISrsRtcPublishStream* publisher = source_->publish_stream();
if (publisher) {
uint32_t ssrc = get_video_publish_ssrc(ssrc_of_media_source);
if (ssrc != 0) {
@ -1653,7 +1658,6 @@ SrsRtcConnection::SrsRtcConnection(SrsRtcServer* s, SrsContextId context_id)
stat_ = new SrsRtcConnectionStatistic();
timer_ = new SrsHourGlass(this, 1000 * SRS_UTIME_MILLISECONDS);
source_ = NULL;
publisher_ = NULL;
player_ = NULL;
sendonly_skt = NULL;
@ -1862,14 +1866,17 @@ srs_error_t SrsRtcConnection::add_player2(SrsRequest* req, SrsSdp& local_sdp)
return err;
}
// TODO: FIXME: Remove unused source.
srs_error_t SrsRtcConnection::initialize(SrsRtcStream* source, SrsRequest* r, bool is_publisher, string username)
{
srs_error_t err = srs_success;
username_ = username;
req = r->copy();
is_publisher_ = is_publisher;
source_ = source;
if (r) { //TODO: FIXME: SrsRequest owner by Stream, not connection
req = r->copy();
}
SrsSessionConfig* cfg = &local_sdp.session_config_;
if ((err = transport_->initialize(cfg)) != srs_success) {

3
trunk/src/app/srs_app_rtc_conn.hpp
View file

@ -168,6 +168,8 @@ private:
SrsCoroutine* trd;
SrsRtcConnection* session_;
private:
SrsRequest* req_;
SrsRtcStream* source_;
SrsHourGlass* timer_;
// key: publish_ssrc, value: send track to process rtp/rtcp
std::map<uint32_t, SrsRtcAudioSendTrack*> audio_tracks_;
@ -346,7 +348,6 @@ private:
// TODO: FIXME: Support reload.
bool encrypt;
SrsRequest* req;
SrsRtcStream* source_;
SrsSdp remote_sdp;
SrsSdp local_sdp;
private: