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Merged
cloudwebrtc merged 1 commit into from
Sep 12, 2022
Merged

Add GetStats for RtpSender and RtpReceiver. #44

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10 changes: 6 additions & 4 deletions include/rtc_peerconnection.h
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Original file line number Diff line number Diff line change
Expand Up @@ -221,11 +221,13 @@ class RTCPeerConnection : public RefCountInterface {

virtual vector<scoped_refptr<RTCMediaStream>> remote_streams() = 0;

virtual bool GetStats(const RTCAudioTrack* track,
scoped_refptr<TrackStatsObserver> observer) = 0;
virtual bool GetStats(scoped_refptr<RTCRtpSender> sender,
OnStatsCollectorSuccess success,
OnStatsCollectorFailure failure) = 0;

virtual bool GetStats(const RTCVideoTrack* track,
scoped_refptr<TrackStatsObserver> observer) = 0;
virtual bool GetStats(scoped_refptr<RTCRtpReceiver> receiver,
OnStatsCollectorSuccess success,
OnStatsCollectorFailure failure) = 0;

virtual void GetStats(OnStatsCollectorSuccess success,
OnStatsCollectorFailure failure) = 0;
Expand Down
26 changes: 0 additions & 26 deletions src/rtc_media_stream_impl.h
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Original file line number Diff line number Diff line change
Expand Up @@ -10,32 +10,6 @@

namespace libwebrtc {

class WebRTCStatsObserver : public webrtc::StatsObserver {
public:
WebRTCStatsObserver(scoped_refptr<TrackStatsObserver> observer,
std::string direction)
: observer_(observer), direction_(direction) {}

virtual ~WebRTCStatsObserver() {}

static rtc::scoped_refptr<WebRTCStatsObserver> Create(
scoped_refptr<TrackStatsObserver> observer,
std::string direction) {
rtc::scoped_refptr<WebRTCStatsObserver> rtc_stats_observer =
rtc::scoped_refptr<WebRTCStatsObserver>(
new rtc::RefCountedObject<WebRTCStatsObserver>(observer,
direction));
rtc_stats_observer->AddRef();
return rtc_stats_observer;
}

virtual void OnComplete(const webrtc::StatsReports& reports) override;

private:
scoped_refptr<TrackStatsObserver> observer_;
std::string direction_;
};

class WebRTCStatsCollectorCallback : public webrtc::RTCStatsCollectorCallback {
public:
WebRTCStatsCollectorCallback(OnStatsCollectorSuccess success,
Expand Down
154 changes: 44 additions & 110 deletions src/rtc_peerconnection_impl.cc
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Original file line number Diff line number Diff line change
Expand Up @@ -27,10 +27,10 @@ static std::map<libwebrtc::RtcpMuxPolicy,
webrtc::PeerConnectionInterface::kRtcpMuxPolicyRequire}};

static std::map<libwebrtc::SdpSemantics, webrtc::SdpSemantics>
sdp_semantics_map = {
{libwebrtc::SdpSemantics::kPlanB, webrtc::SdpSemantics::kPlanB_DEPRECATED},
{libwebrtc::SdpSemantics::kUnifiedPlan,
webrtc::SdpSemantics::kUnifiedPlan}};
sdp_semantics_map = {{libwebrtc::SdpSemantics::kPlanB,
webrtc::SdpSemantics::kPlanB_DEPRECATED},
{libwebrtc::SdpSemantics::kUnifiedPlan,
webrtc::SdpSemantics::kUnifiedPlan}};

static std::map<libwebrtc::CandidateNetworkPolicy,
webrtc::PeerConnectionInterface::CandidateNetworkPolicy>
Expand Down Expand Up @@ -69,8 +69,7 @@ static std::map<webrtc::PeerConnectionInterface::PeerConnectionState,
{webrtc::PeerConnectionInterface::PeerConnectionState::kConnected,
libwebrtc::RTCPeerConnectionState::RTCPeerConnectionStateConnected},
{webrtc::PeerConnectionInterface::PeerConnectionState::kDisconnected,
libwebrtc::RTCPeerConnectionState::
RTCPeerConnectionStateDisconnected},
libwebrtc::RTCPeerConnectionState::RTCPeerConnectionStateDisconnected},
{webrtc::PeerConnectionInterface::PeerConnectionState::kClosed,
libwebrtc::RTCPeerConnectionState::RTCPeerConnectionStateClosed},
{webrtc::PeerConnectionInterface::PeerConnectionState::kFailed,
Expand Down Expand Up @@ -352,8 +351,8 @@ void RTCPeerConnectionImpl::OnIceCandidate(
}

RTC_LOG(LS_INFO) << __FUNCTION__ << ", mid " << candidate->sdp_mid()
<< ", mline " << candidate->sdp_mline_index() << ", sdp"
<< cand_sdp;
<< ", mline " << candidate->sdp_mline_index() << ", sdp"
<< cand_sdp;
}

void RTCPeerConnectionImpl::RegisterRTCPeerConnectionObserver(
Expand Down Expand Up @@ -408,7 +407,8 @@ bool RTCPeerConnectionImpl::Initialize() {

RTCMediaConstraintsImpl* media_constraints =
static_cast<RTCMediaConstraintsImpl*>(constraints_.get());
webrtc::MediaConstraints rtc_constraints(media_constraints->GetMandatory(),media_constraints->GetOptional());
webrtc::MediaConstraints rtc_constraints(media_constraints->GetMandatory(),
media_constraints->GetOptional());
CopyConstraintsIntoRtcConfiguration(&rtc_constraints, &config);

webrtc::PeerConnectionFactoryInterface::Options options;
Expand Down Expand Up @@ -563,7 +563,8 @@ void RTCPeerConnectionImpl::CreateOffer(
RTCMediaConstraintsImpl* media_constraints =
static_cast<RTCMediaConstraintsImpl*>(constraints.get());
webrtc::PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options;
webrtc::MediaConstraints rtc_constraints(media_constraints->GetMandatory(),media_constraints->GetOptional());
webrtc::MediaConstraints rtc_constraints(media_constraints->GetMandatory(),
media_constraints->GetOptional());
if (CopyConstraintsIntoOfferAnswerOptions(&rtc_constraints,
&offer_answer_options) == false) {
offer_answer_options = offer_answer_options_;
Expand All @@ -586,7 +587,8 @@ void RTCPeerConnectionImpl::CreateAnswer(
RTCMediaConstraintsImpl* media_constraints =
static_cast<RTCMediaConstraintsImpl*>(constraints.get());
webrtc::PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options;
webrtc::MediaConstraints rtc_constraints(media_constraints->GetMandatory(),media_constraints->GetOptional());
webrtc::MediaConstraints rtc_constraints(media_constraints->GetMandatory(),
media_constraints->GetOptional());
if (CopyConstraintsIntoOfferAnswerOptions(&rtc_constraints,
&offer_answer_options) == false) {
offer_answer_options = offer_answer_options_;
Expand Down Expand Up @@ -662,37 +664,45 @@ int RTCPeerConnectionImpl::RemoveStream(scoped_refptr<RTCMediaStream> stream) {
return 0;
}

scoped_refptr<RTCMediaStream> RTCPeerConnectionImpl::CreateLocalMediaStream(const string stream_id) {
scoped_refptr<RTCMediaStream> RTCPeerConnectionImpl::CreateLocalMediaStream(
const string stream_id) {
if (!rtc_peerconnection_factory_.get()) {
return nullptr;
}
auto stream = rtc_peerconnection_factory_->CreateLocalMediaStream(stream_id.c_string());
auto stream =
rtc_peerconnection_factory_->CreateLocalMediaStream(stream_id.c_string());
auto rtc_stream = new RefCountedObject<MediaStreamImpl>(stream);
local_streams_.push_back(rtc_stream);
return rtc_stream;
}

bool RTCPeerConnectionImpl::GetStats(
const RTCAudioTrack* track,
scoped_refptr<TrackStatsObserver> observer) {
AudioTrackImpl* impl = static_cast<AudioTrackImpl*>((RTCAudioTrack*)track);
rtc::scoped_refptr<WebRTCStatsObserver> rtc_observer =
WebRTCStatsObserver::Create(observer, "stats");
rtc_peerconnection_->GetStats(
rtc_observer.get(), impl->rtc_track().get(),
webrtc::PeerConnectionInterface::kStatsOutputLevelDebug);
bool RTCPeerConnectionImpl::GetStats(scoped_refptr<RTCRtpSender> sender,
OnStatsCollectorSuccess success,
OnStatsCollectorFailure failure) {
rtc::scoped_refptr<WebRTCStatsCollectorCallback> rtc_callback =
WebRTCStatsCollectorCallback::Create(success, failure);
if (!rtc_peerconnection_.get() || !rtc_peerconnection_factory_.get()) {
webrtc::MutexLock cs(callback_crt_sec_.get());
failure("Failed to initialize PeerConnection");
return false;
}
RTCRtpSenderImpl* impl = static_cast<RTCRtpSenderImpl*>(sender.get());
rtc_peerconnection_->GetStats(impl->rtc_rtp_sender(), rtc_callback);
return true;
}

bool RTCPeerConnectionImpl::GetStats(
const RTCVideoTrack* track,
scoped_refptr<TrackStatsObserver> observer) {
VideoTrackImpl* impl = static_cast<VideoTrackImpl*>((RTCVideoTrack*)track);
rtc::scoped_refptr<WebRTCStatsObserver> rtc_observer =
WebRTCStatsObserver::Create(observer, "recv");
rtc_peerconnection_->GetStats(
rtc_observer.get(), impl->rtc_track().get(),
webrtc::PeerConnectionInterface::kStatsOutputLevelDebug);
bool RTCPeerConnectionImpl::GetStats(scoped_refptr<RTCRtpReceiver> receiver,
OnStatsCollectorSuccess success,
OnStatsCollectorFailure failure) {
rtc::scoped_refptr<WebRTCStatsCollectorCallback> rtc_callback =
WebRTCStatsCollectorCallback::Create(success, failure);
if (!rtc_peerconnection_.get() || !rtc_peerconnection_factory_.get()) {
webrtc::MutexLock cs(callback_crt_sec_.get());
failure("Failed to initialize PeerConnection");
return false;
}
RTCRtpReceiverImpl* impl = static_cast<RTCRtpReceiverImpl*>(receiver.get());
rtc_peerconnection_->GetStats(impl->rtp_receiver(), rtc_callback);
return true;
}

Expand Down Expand Up @@ -821,8 +831,9 @@ scoped_refptr<RTCRtpSender> RTCPeerConnectionImpl::AddTrack(

bool RTCPeerConnectionImpl::RemoveTrack(scoped_refptr<RTCRtpSender> render) {
RTCRtpSenderImpl* impl = static_cast<RTCRtpSenderImpl*>(render.get());
webrtc::RTCError err = rtc_peerconnection_->RemoveTrackOrError(impl->rtc_rtp_sender());
if(err.ok()) {
webrtc::RTCError err =
rtc_peerconnection_->RemoveTrackOrError(impl->rtc_rtp_sender());
if (err.ok()) {
return true;
}
return false;
Expand Down Expand Up @@ -886,83 +897,6 @@ void WebRTCStatsCollectorCallback::OnStatsDelivered(
success_(reports);
}

void WebRTCStatsObserver::OnComplete(const webrtc::StatsReports& reports) {
MediaTrackStatistics stats;

for (auto report : reports) {
// for (auto kv : report->values())
// {
// LOG_WARNING(<< kv.second->display_name() << ": "
// << kv.second->ToString());
// }

const webrtc::StatsReport::Value* kv =
report->FindValue(webrtc::StatsReport::kStatsValueNameBytesReceived);
if (kv) {
stats.bytes_received = kv->int64_val();
}
kv = report->FindValue(webrtc::StatsReport::kStatsValueNameBytesSent);
if (kv) {
stats.bytes_sent = kv->int64_val();
}

kv = report->FindValue(webrtc::StatsReport::kStatsValueNameRtt);
if (kv) {
stats.rtt = kv->int64_val();
}

kv = report->FindValue(webrtc::StatsReport::kStatsValueNamePacketsSent);
if (kv) {
stats.packets_sent = kv->int_val();
}

kv = report->FindValue(webrtc::StatsReport::kStatsValueNameFrameRateSent);
if (kv) {
stats.frame_rate_sent = kv->int_val();
}

kv = report->FindValue(
webrtc::StatsReport::kStatsValueNameFrameRateReceived);
if (kv) {
stats.frame_rate_received = kv->int_val();
}

kv = report->FindValue(webrtc::StatsReport::kStatsValueNamePacketsReceived);
if (kv) {
stats.packets_received = kv->int_val();
}

kv = report->FindValue(webrtc::StatsReport::kStatsValueNamePacketsLost);
if (kv) {
stats.packets_lost = kv->int_val();
}

kv = report->FindValue(webrtc::StatsReport::kStatsValueNameSsrc);
if (kv) {
stats.ssrc = kv->int64_val();
}

kv = report->FindValue(webrtc::StatsReport::kStatsValueNameTrackId);
if (kv) {
stats.msid = string(kv->static_string_val());
}

kv = report->FindValue(webrtc::StatsReport::kStatsValueNameMediaType);
if (kv) {
stats.kind = string(kv->static_string_val());
}

stats.direction = direction_;
}

if (observer_)
observer_->OnComplete(stats);

observer_ = nullptr;

this->Release();
}

MediaRTCStatsImpl::MediaRTCStatsImpl(std::unique_ptr<webrtc::RTCStats> stats)
: stats_(std::move(stats)) {}

Expand Down
10 changes: 6 additions & 4 deletions src/rtc_peerconnection_impl.h
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Original file line number Diff line number Diff line change
Expand Up @@ -130,11 +130,13 @@ class RTCPeerConnectionImpl : public RTCPeerConnection,
const string label,
RTCDataChannelInit* dataChannelDict) override;

virtual bool GetStats(const RTCAudioTrack* track,
scoped_refptr<TrackStatsObserver> observer) override;
virtual bool GetStats(scoped_refptr<RTCRtpSender> sender,
OnStatsCollectorSuccess success,
OnStatsCollectorFailure failure) override;

virtual bool GetStats(const RTCVideoTrack* track,
scoped_refptr<TrackStatsObserver> observer) override;
virtual bool GetStats(scoped_refptr<RTCRtpReceiver> receiver,
OnStatsCollectorSuccess success,
OnStatsCollectorFailure failure) override;

virtual void GetStats(OnStatsCollectorSuccess success,
OnStatsCollectorFailure failure) override;
Expand Down