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Initial support for multiple MediaStreams in a single WebRtcConnection (
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#1162)
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@jcague
jcague committed Mar 6, 2018
1 parent d694a99 commit 4a517f1
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Showing 2 changed files with 76 additions and 51 deletions.
120 changes: 74 additions & 46 deletions erizo/src/erizo/WebRtcConnection.cpp
View file
Expand Up @@ -100,14 +100,6 @@ bool WebRtcConnection::init() {
return true;
}

bool WebRtcConnection::isSourceSSRC(uint32_t ssrc) {
return media_stream_->isVideoSourceSSRC(ssrc) || media_stream_->isAudioSourceSSRC(ssrc);
}

bool WebRtcConnection::isSinkSSRC(uint32_t ssrc) {
return media_stream_->isVideoSinkSSRC(ssrc) || media_stream_->isAudioSinkSSRC(ssrc);
}

bool WebRtcConnection::createOffer(bool video_enabled, bool audioEnabled, bool bundle) {
bundle_ = bundle;
video_enabled_ = video_enabled;
Expand All @@ -117,10 +109,17 @@ bool WebRtcConnection::createOffer(bool video_enabled, bool audioEnabled, bool b

ELOG_DEBUG("%s message: Creating sdp offer, isBundle: %d", toLog(), bundle_);

if (video_enabled_)
local_sdp_->video_ssrc_list.push_back(media_stream_->getVideoSinkSSRC());
if (audio_enabled_)
local_sdp_->audio_ssrc = media_stream_->getAudioSinkSSRC();
if (video_enabled_) {
forEachMediaStream([this] (const std::shared_ptr<MediaStream> &media_stream) {
local_sdp_->video_ssrc_list.push_back(media_stream->getVideoSinkSSRC());
});
}
if (audio_enabled_) {
forEachMediaStream([this] (const std::shared_ptr<MediaStream> &media_stream) {
local_sdp_->audio_ssrc = media_stream->getAudioSinkSSRC();
});
}


auto listener = std::dynamic_pointer_cast<TransportListener>(shared_from_this());

Expand Down Expand Up @@ -148,18 +147,27 @@ bool WebRtcConnection::createOffer(bool video_enabled, bool audioEnabled, bool b
std::string msg = this->getLocalSdp();
conn_event_listener_->notifyEvent(global_state_, msg);
}
media_stream_->setLocalSdp(local_sdp_);
forEachMediaStream([this] (const std::shared_ptr<MediaStream> &media_stream) {
media_stream->setLocalSdp(local_sdp_);
});
return true;
}

void WebRtcConnection::addMediaStream(std::shared_ptr<MediaStream> media_stream) {
ELOG_DEBUG("%s message: Adding mediaStream, id: %s", toLog(), media_stream->getId().c_str());
media_stream_ = media_stream;
media_streams_.push_back(media_stream);
}

void WebRtcConnection::removeMediaStream(const std::string& stream_id) {
ELOG_DEBUG("%s message: removing mediaStream, id: %s", toLog(), stream_id.c_str())
media_stream_.reset();
media_streams_.erase(std::remove_if(media_streams_.begin(), media_streams_.end(),
[stream_id](const std::shared_ptr<MediaStream> &stream) {
return stream->getId() == stream_id;
}), media_streams_.end());
}

void WebRtcConnection::forEachMediaStream(std::function<void(const std::shared_ptr<MediaStream>&)> func) {
std::for_each(media_streams_.begin(), media_streams_.end(), func);
}

bool WebRtcConnection::setRemoteSdpInfo(std::shared_ptr<SdpInfo> sdp) {
Expand Down Expand Up @@ -192,7 +200,9 @@ bool WebRtcConnection::setRemoteSdp(const std::string &sdp) {
bool WebRtcConnection::processRemoteSdp() {
ELOG_DEBUG("%s message: processing remote SDP", toLog());
if (remote_sdp_processed_) {
media_stream_->setRemoteSdp(remote_sdp_);
forEachMediaStream([this] (const std::shared_ptr<MediaStream> &media_stream) {
media_stream->setRemoteSdp(remote_sdp_);
});
return true;
}

Expand All @@ -201,8 +211,10 @@ bool WebRtcConnection::processRemoteSdp() {
extension_processor_.setSdpInfo(local_sdp_);
local_sdp_->updateSupportedExtensionMap(extension_processor_.getSupportedExtensionMap());

local_sdp_->video_ssrc_list.push_back(media_stream_->getVideoSinkSSRC());
local_sdp_->audio_ssrc = media_stream_->getAudioSinkSSRC();
forEachMediaStream([this] (const std::shared_ptr<MediaStream> &media_stream) {
local_sdp_->video_ssrc_list.push_back(media_stream->getVideoSinkSSRC());
local_sdp_->audio_ssrc = media_stream->getAudioSinkSSRC();
});

if (remote_sdp_->dtlsRole == ACTPASS) {
local_sdp_->dtlsRole = ACTIVE;
Expand Down Expand Up @@ -268,9 +280,10 @@ bool WebRtcConnection::processRemoteSdp() {
conn_event_listener_->notifyEvent(CONN_SDP, object);
}
}

media_stream_->setRemoteSdp(remote_sdp_);
media_stream_->setLocalSdp(local_sdp_);
forEachMediaStream([this] (const std::shared_ptr<MediaStream> &media_stream) {
media_stream->setRemoteSdp(remote_sdp_);
media_stream->setLocalSdp(local_sdp_);
});
remote_sdp_processed_ = true;
return true;
}
Expand Down Expand Up @@ -385,7 +398,18 @@ void WebRtcConnection::onTransportData(std::shared_ptr<DataPacket> packet, Trans
if (getCurrentState() != CONN_READY) {
return;
}
media_stream_->onTransportData(packet, transport);
char* buf = packet->data;
RtpHeader *head = reinterpret_cast<RtpHeader*> (buf);
RtcpHeader *chead = reinterpret_cast<RtcpHeader*> (buf);
uint32_t ssrc = head->getSSRC();
if (chead->isRtcp() && chead->packettype != RTCP_Sender_PT) { // Sender Report
ssrc = chead->getSourceSSRC();
}
forEachMediaStream([packet, transport, ssrc] (const std::shared_ptr<MediaStream> &media_stream) {
if (media_stream->isSourceSSRC(ssrc) || media_stream->isSinkSSRC(ssrc)) {
media_stream->onTransportData(packet, transport);
}
});
}

void WebRtcConnection::asyncTask(std::function<void(std::shared_ptr<WebRtcConnection>)> f) {
Expand Down Expand Up @@ -456,9 +480,9 @@ void WebRtcConnection::updateState(TransportState state, Transport * transport)
if (bundle_) {
temp = CONN_READY;
trackTransportInfo();
if (media_stream_) {
media_stream_->sendPLIToFeedback();
}
forEachMediaStream([] (const std::shared_ptr<MediaStream> &media_stream) {
media_stream->sendPLIToFeedback();
});
} else {
if ((!remote_sdp_->hasAudio || (audio_transport_.get() != nullptr
&& audio_transport_->getTransportState() == TRANSPORT_READY)) &&
Expand All @@ -467,9 +491,9 @@ void WebRtcConnection::updateState(TransportState state, Transport * transport)
// WebRTCConnection will be ready only when all channels are ready.
temp = CONN_READY;
trackTransportInfo();
if (media_stream_) {
media_stream_->sendPLIToFeedback();
}
forEachMediaStream([] (const std::shared_ptr<MediaStream> &media_stream) {
media_stream->sendPLIToFeedback();
});
}
}
break;
Expand Down Expand Up @@ -515,17 +539,19 @@ void WebRtcConnection::trackTransportInfo() {
candidate_pair = video_transport_->getIceConnection()->getSelectedPair();
asyncTask([candidate_pair] (std::shared_ptr<WebRtcConnection> connection) {
std::shared_ptr<Stats> stats = connection->stats_;
uint32_t video_sink_ssrc = connection->getMediaStream()->getVideoSinkSSRC();
uint32_t video_source_ssrc = connection->getMediaStream()->getVideoSourceSSRC();
connection->forEachMediaStream([stats, candidate_pair] (const std::shared_ptr<MediaStream> &media_stream) {
uint32_t video_sink_ssrc = media_stream->getVideoSinkSSRC();
uint32_t video_source_ssrc = media_stream->getVideoSourceSSRC();

if (video_sink_ssrc != kDefaultVideoSinkSSRC) {
stats->getNode()[video_sink_ssrc].insertStat("clientHostType",
StringStat{candidate_pair.clientHostType});
}
if (video_source_ssrc != 0) {
stats->getNode()[video_source_ssrc].insertStat("clientHostType",
if (video_sink_ssrc != kDefaultVideoSinkSSRC) {
stats->getNode()[video_sink_ssrc].insertStat("clientHostType",
StringStat{candidate_pair.clientHostType});
}
}
if (video_source_ssrc != 0) {
stats->getNode()[video_source_ssrc].insertStat("clientHostType",
StringStat{candidate_pair.clientHostType});
}
});
});
}

Expand All @@ -535,17 +561,19 @@ void WebRtcConnection::trackTransportInfo() {
}
asyncTask([candidate_pair] (std::shared_ptr<WebRtcConnection> connection) {
std::shared_ptr<Stats> stats = connection->stats_;
uint32_t audio_sink_ssrc = connection->getMediaStream()->getAudioSinkSSRC();
uint32_t audio_source_ssrc = connection->getMediaStream()->getAudioSourceSSRC();
connection->forEachMediaStream([stats, candidate_pair] (const std::shared_ptr<MediaStream> &media_stream) {
uint32_t audio_sink_ssrc = media_stream->getAudioSinkSSRC();
uint32_t audio_source_ssrc = media_stream->getAudioSourceSSRC();

if (audio_sink_ssrc != kDefaultAudioSinkSSRC) {
stats->getNode()[audio_sink_ssrc].insertStat("clientHostType",
StringStat{candidate_pair.clientHostType});
}
if (audio_source_ssrc != 0) {
stats->getNode()[audio_source_ssrc].insertStat("clientHostType",
if (audio_sink_ssrc != kDefaultAudioSinkSSRC) {
stats->getNode()[audio_sink_ssrc].insertStat("clientHostType",
StringStat{candidate_pair.clientHostType});
}
}
if (audio_source_ssrc != 0) {
stats->getNode()[audio_source_ssrc].insertStat("clientHostType",
StringStat{candidate_pair.clientHostType});
}
});
});
}
}
Expand Down
7 changes: 2 additions & 5 deletions erizo/src/erizo/WebRtcConnection.h
View file
Expand Up @@ -137,19 +137,16 @@ class WebRtcConnection: public TransportListener, public LogContext,
bool isAudioMuted() { return audio_muted_; }
bool isVideoMuted() { return video_muted_; }

std::shared_ptr<MediaStream> getMediaStream() { return media_stream_; }
void addMediaStream(std::shared_ptr<MediaStream> media_stream);
void removeMediaStream(const std::string& stream_id);
void forEachMediaStream(std::function<void(const std::shared_ptr<MediaStream>&)> func);

std::shared_ptr<Stats> getStatsService() { return stats_; }

RtpExtensionProcessor& getRtpExtensionProcessor() { return extension_processor_; }

std::shared_ptr<Worker> getWorker() { return worker_; }

bool isSourceSSRC(uint32_t ssrc);
bool isSinkSSRC(uint32_t ssrc);

inline std::string toLog() {
return "id: " + connection_id_ + ", " + printLogContext();
}
Expand Down Expand Up @@ -182,7 +179,7 @@ class WebRtcConnection: public TransportListener, public LogContext,

std::shared_ptr<Worker> worker_;
std::shared_ptr<IOWorker> io_worker_;
std::shared_ptr<MediaStream> media_stream_;
std::vector<std::shared_ptr<MediaStream>> media_streams_;
std::shared_ptr<SdpInfo> remote_sdp_;
std::shared_ptr<SdpInfo> local_sdp_;
bool audio_muted_;
Expand Down

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