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fune/dom/media/webrtc/jsapi/RTCRtpReceiver.cpp

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/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
* You can obtain one at http://mozilla.org/MPL/2.0/. */
#include "RTCRtpReceiver.h"
#include "transport/logging.h"
#include "mozilla/dom/MediaStreamTrack.h"
#include "mozilla/dom/Promise.h"
#include "transportbridge/MediaPipeline.h"
#include "nsPIDOMWindow.h"
#include "PrincipalHandle.h"
#include "nsIPrincipal.h"
#include "mozilla/dom/Document.h"
#include "mozilla/NullPrincipal.h"
#include "MediaTrackGraph.h"
#include "RemoteTrackSource.h"
#include "libwebrtcglue/RtpRtcpConfig.h"
#include "nsString.h"
#include "mozilla/dom/AudioStreamTrack.h"
#include "mozilla/dom/VideoStreamTrack.h"
#include "MediaTransportHandler.h"
#include "jsep/JsepTransceiver.h"
#include "mozilla/dom/RTCRtpReceiverBinding.h"
#include "mozilla/dom/RTCRtpSourcesBinding.h"
#include "RTCStatsReport.h"
#include "mozilla/Preferences.h"
#include "PeerConnectionCtx.h"
#include "TransceiverImpl.h"
#include "libwebrtcglue/AudioConduit.h"
namespace mozilla::dom {
NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE(RTCRtpReceiver, mWindow, mPc, mTrack)
NS_IMPL_CYCLE_COLLECTING_ADDREF(RTCRtpReceiver)
NS_IMPL_CYCLE_COLLECTING_RELEASE(RTCRtpReceiver)
NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(RTCRtpReceiver)
NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY
NS_INTERFACE_MAP_ENTRY(nsISupports)
NS_INTERFACE_MAP_END
LazyLogModule gReceiverLog("RTCRtpReceiver");
static PrincipalHandle GetPrincipalHandle(nsPIDOMWindowInner* aWindow,
bool aPrivacyNeeded) {
// Set the principal used for creating the tracks. This makes the track
// data (audio/video samples) accessible to the receiving page. We're
// only certain that privacy hasn't been requested if we're connected.
nsCOMPtr<nsIScriptObjectPrincipal> winPrincipal = do_QueryInterface(aWindow);
RefPtr<nsIPrincipal> principal = winPrincipal->GetPrincipal();
if (NS_WARN_IF(!principal)) {
principal = NullPrincipal::CreateWithoutOriginAttributes();
} else if (aPrivacyNeeded) {
principal = NullPrincipal::CreateWithInheritedAttributes(principal);
}
return MakePrincipalHandle(principal);
}
static already_AddRefed<dom::MediaStreamTrack> CreateTrack(
nsPIDOMWindowInner* aWindow, bool aAudio,
const nsCOMPtr<nsIPrincipal>& aPrincipal) {
MediaTrackGraph* graph = MediaTrackGraph::GetInstance(
aAudio ? MediaTrackGraph::AUDIO_THREAD_DRIVER
: MediaTrackGraph::SYSTEM_THREAD_DRIVER,
aWindow, MediaTrackGraph::REQUEST_DEFAULT_SAMPLE_RATE,
MediaTrackGraph::DEFAULT_OUTPUT_DEVICE);
RefPtr<MediaStreamTrack> track;
RefPtr<RemoteTrackSource> trackSource;
if (aAudio) {
RefPtr<SourceMediaTrack> source =
graph->CreateSourceTrack(MediaSegment::AUDIO);
trackSource = new RemoteTrackSource(source, aPrincipal, u"remote audio"_ns);
track = new AudioStreamTrack(aWindow, source, trackSource);
} else {
RefPtr<SourceMediaTrack> source =
graph->CreateSourceTrack(MediaSegment::VIDEO);
trackSource = new RemoteTrackSource(source, aPrincipal, u"remote video"_ns);
track = new VideoStreamTrack(aWindow, source, trackSource);
}
// Spec says remote tracks start out muted.
trackSource->SetMuted(true);
return track.forget();
}
#define INIT_CANONICAL(name, val) \
name(AbstractThread::MainThread(), val, \
"RTCRtpReceiver::" #name " (Canonical)")
RTCRtpReceiver::RTCRtpReceiver(
nsPIDOMWindowInner* aWindow, bool aPrivacyNeeded, PeerConnectionImpl* aPc,
MediaTransportHandler* aTransportHandler, JsepTransceiver* aJsepTransceiver,
AbstractThread* aCallThread, nsISerialEventTarget* aStsThread,
MediaSessionConduit* aConduit, TransceiverImpl* aTransceiverImpl)
: mWindow(aWindow),
mPc(aPc),
mJsepTransceiver(aJsepTransceiver),
mCallThread(aCallThread),
mStsThread(aStsThread),
mTransportHandler(aTransportHandler),
mTransceiverImpl(aTransceiverImpl),
INIT_CANONICAL(mSsrc, 0),
INIT_CANONICAL(mVideoRtxSsrc, 0),
INIT_CANONICAL(mLocalRtpExtensions, RtpExtList()),
INIT_CANONICAL(mAudioCodecs, std::vector<AudioCodecConfig>()),
INIT_CANONICAL(mVideoCodecs, std::vector<VideoCodecConfig>()),
INIT_CANONICAL(mVideoRtpRtcpConfig, Nothing()),
INIT_CANONICAL(mReceiving, false) {
PrincipalHandle principalHandle = GetPrincipalHandle(aWindow, aPrivacyNeeded);
mTrack = CreateTrack(aWindow, aConduit->type() == MediaSessionConduit::AUDIO,
principalHandle.get());
// Until Bug 1232234 is fixed, we'll get extra RTCP BYES during renegotiation,
// so we'll disable muting on RTCP BYE and timeout for now.
if (Preferences::GetBool("media.peerconnection.mute_on_bye_or_timeout",
false)) {
mRtcpByeListener = aConduit->RtcpByeEvent().Connect(
GetMainThreadSerialEventTarget(), this, &RTCRtpReceiver::OnRtcpBye);
mRtcpTimeoutListener = aConduit->RtcpTimeoutEvent().Connect(
GetMainThreadSerialEventTarget(), this, &RTCRtpReceiver::OnRtcpTimeout);
}
if (aConduit->type() == MediaSessionConduit::AUDIO) {
mPipeline = new MediaPipelineReceiveAudio(
mPc->GetHandle(), aTransportHandler, aCallThread, mStsThread.get(),
*aConduit->AsAudioSessionConduit(), mTrack, principalHandle);
} else {
mPipeline = new MediaPipelineReceiveVideo(
mPc->GetHandle(), aTransportHandler, aCallThread, mStsThread.get(),
*aConduit->AsVideoSessionConduit(), mTrack, principalHandle);
}
}
#undef INIT_CANONICAL
RTCRtpReceiver::~RTCRtpReceiver() { MOZ_ASSERT(!mPipeline); }
JSObject* RTCRtpReceiver::WrapObject(JSContext* aCx,
JS::Handle<JSObject*> aGivenProto) {
return RTCRtpReceiver_Binding::Wrap(aCx, this, aGivenProto);
}
RTCDtlsTransport* RTCRtpReceiver::GetTransport() const {
if (!mTransceiverImpl) {
return nullptr;
}
return mTransceiverImpl->GetDtlsTransport();
}
already_AddRefed<Promise> RTCRtpReceiver::GetStats() {
nsCOMPtr<nsIGlobalObject> global = do_QueryInterface(mWindow);
ErrorResult rv;
RefPtr<Promise> promise = Promise::Create(global, rv);
if (NS_WARN_IF(rv.Failed())) {
rv.StealNSResult();
return nullptr;
}
if (NS_WARN_IF(!mTransceiverImpl)) {
// TODO(bug 1056433): When we stop nulling this out when the PC is closed
// (or when the transceiver is stopped), we can remove this code. We
// resolve instead of reject in order to make this eventual change in
// behavior a little smaller.
promise->MaybeResolve(new RTCStatsReport(mWindow));
return promise.forget();
}
mTransceiverImpl->ChainToDomPromiseWithCodecStats(GetStatsInternal(),
promise);
return promise.forget();
}
nsTArray<RefPtr<RTCStatsPromise>> RTCRtpReceiver::GetStatsInternal() {
MOZ_ASSERT(NS_IsMainThread());
nsTArray<RefPtr<RTCStatsPromise>> promises(3);
if (!mPipeline) {
return promises;
}
if (!mHaveStartedReceiving) {
return promises;
}
nsString recvTrackId;
MOZ_ASSERT(mTrack);
if (mTrack) {
mTrack->GetId(recvTrackId);
}
{
// Add bandwidth estimation stats
promises.AppendElement(InvokeAsync(
mCallThread, __func__,
[conduit = mPipeline->mConduit, recvTrackId]() mutable {
auto report = MakeUnique<dom::RTCStatsCollection>();
const Maybe<webrtc::Call::Stats> stats = conduit->GetCallStats();
stats.apply([&](const auto& aStats) {
dom::RTCBandwidthEstimationInternal bw;
bw.mTrackIdentifier = recvTrackId;
bw.mSendBandwidthBps.Construct(aStats.send_bandwidth_bps / 8);
bw.mMaxPaddingBps.Construct(aStats.max_padding_bitrate_bps / 8);
bw.mReceiveBandwidthBps.Construct(aStats.recv_bandwidth_bps / 8);
bw.mPacerDelayMs.Construct(aStats.pacer_delay_ms);
if (aStats.rtt_ms >= 0) {
bw.mRttMs.Construct(aStats.rtt_ms);
}
if (!report->mBandwidthEstimations.AppendElement(std::move(bw),
fallible)) {
mozalloc_handle_oom(0);
}
});
return RTCStatsPromise::CreateAndResolve(std::move(report), __func__);
}));
}
promises.AppendElement(
InvokeAsync(
mCallThread, __func__,
[pipeline = mPipeline, recvTrackId] {
auto report = MakeUnique<dom::RTCStatsCollection>();
auto asAudio = pipeline->mConduit->AsAudioSessionConduit();
auto asVideo = pipeline->mConduit->AsVideoSessionConduit();
nsString kind = asVideo.isNothing() ? u"audio"_ns : u"video"_ns;
nsString idstr = kind + u"_"_ns;
idstr.AppendInt(static_cast<uint32_t>(pipeline->Level()));
Maybe<uint32_t> ssrc = pipeline->mConduit->GetRemoteSSRC();
// Add frame history
asVideo.apply([&](const auto& conduit) {
if (conduit->AddFrameHistory(&report->mVideoFrameHistories)) {
auto& history = report->mVideoFrameHistories.LastElement();
history.mTrackIdentifier = recvTrackId;
}
});
// TODO(@@NG):ssrcs handle Conduits having multiple stats at the
// same level.
// This is pending spec work.
// Gather pipeline stats.
nsString localId = u"inbound_rtp_"_ns + idstr;
nsString remoteId;
auto constructCommonRemoteOutboundRtpStats =
[&](RTCRemoteOutboundRtpStreamStats& aRemote,
const DOMHighResTimeStamp& aTimestamp) {
remoteId = u"inbound_rtcp_"_ns + idstr;
aRemote.mTimestamp.Construct(aTimestamp);
aRemote.mId.Construct(remoteId);
aRemote.mType.Construct(RTCStatsType::Remote_outbound_rtp);
ssrc.apply(
[&](uint32_t aSsrc) { aRemote.mSsrc.Construct(aSsrc); });
aRemote.mMediaType.Construct(
kind); // mediaType is the old name for kind.
aRemote.mKind.Construct(kind);
aRemote.mLocalId.Construct(localId);
};
auto constructCommonInboundRtpStats =
[&](RTCInboundRtpStreamStats& aLocal) {
aLocal.mTimestamp.Construct(
pipeline->GetTimestampMaker().GetNow());
aLocal.mId.Construct(localId);
aLocal.mType.Construct(RTCStatsType::Inbound_rtp);
ssrc.apply(
[&](uint32_t aSsrc) { aLocal.mSsrc.Construct(aSsrc); });
aLocal.mMediaType.Construct(
kind); // mediaType is the old name for kind.
aLocal.mKind.Construct(kind);
if (remoteId.Length()) {
aLocal.mRemoteId.Construct(remoteId);
}
};
asAudio.apply([&](auto& aConduit) {
Maybe<webrtc::AudioReceiveStream::Stats> audioStats =
aConduit->GetReceiverStats();
if (audioStats.isNothing()) {
return;
}
if (!audioStats->last_packet_received_timestamp_ms) {
// By spec: "The lifetime of all RTP monitored objects starts
// when the RTP stream is first used: When the first RTP packet
// is sent or received on the SSRC it represents"
return;
}
// First, fill in remote stat with rtcp sender data, if present.
if (audioStats->last_sender_report_timestamp_ms) {
RTCRemoteOutboundRtpStreamStats remote;
constructCommonRemoteOutboundRtpStats(
remote,
aConduit->GetTimestampMaker().ConvertNtpToDomTime(
webrtc::Timestamp::Millis(
*audioStats->last_sender_report_timestamp_ms) +
webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970)));
remote.mPacketsSent.Construct(
audioStats->sender_reports_packets_sent);
remote.mBytesSent.Construct(
audioStats->sender_reports_bytes_sent);
remote.mRemoteTimestamp.Construct(
*audioStats->last_sender_report_remote_timestamp_ms);
if (!report->mRemoteOutboundRtpStreamStats.AppendElement(
std::move(remote), fallible)) {
mozalloc_handle_oom(0);
}
}
// Then, fill in local side (with cross-link to remote only if
// present)
RTCInboundRtpStreamStats local;
constructCommonInboundRtpStats(local);
local.mJitter.Construct(audioStats->jitter_ms / 1000.0);
local.mPacketsLost.Construct(audioStats->packets_lost);
local.mPacketsReceived.Construct(audioStats->packets_rcvd);
local.mPacketsDiscarded.Construct(audioStats->packets_discarded);
local.mBytesReceived.Construct(audioStats->payload_bytes_rcvd);
/*
* Potential new stats that are now available upstream.
if (audioStats->last_packet_received_timestamp_ms) {
local.mLastPacketReceivedTimestamp.Construct(
aConduit->GetTimestampMaker().ConvertNtpToDomTime(
webrtc::Timestamp::Millis(
*audioStats->last_packet_received_timestamp_ms) +
webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970)));
}
local.mHeaderBytesReceived.Construct(
audioStats->header_and_padding_bytes_rcvd);
local.mFecPacketsReceived.Construct(
audioStats->fec_packets_received);
local.mFecPacketsDiscarded.Construct(
audioStats->fec_packets_discarded);
if (audioStats->estimated_playout_ntp_timestamp_ms) {
local.mEstimatedPlayoutTimestamp.Construct(
aConduit->GetTimestampMaker().ConvertNtpToDomTime(
webrtc::Timestamp::Millis(
*audioStats->estimated_playout_ntp_timestamp_ms)));
}
local.mJitterBufferDelay.Construct(
audioStats->jitter_buffer_delay_seconds);
local.mJitterBufferEmittedCount.Construct(
audioStats->jitter_buffer_emitted_count);
local.mTotalSamplesReceived.Construct(
audioStats->total_samples_received);
local.mConcealedSamples.Construct(audioStats->concealed_samples);
local.mSilentConcealedSamples.Construct(
audioStats->silent_concealed_samples);
local.mConcealmentEvents.Construct(
audioStats->concealment_events);
local.mInsertedSamplesForDeceleration.Construct(
audioStats->inserted_samples_for_deceleration);
local.mRemovedSamplesForAcceleration.Construct(
audioStats->removed_samples_for_acceleration);
if (audioStats->audio_level >= 0 &&
audioStats->audio_level <= 32767) {
local.mAudioLevel.Construct(audioStats->audio_level / 32767.0);
}
local.mTotalAudioEnergy.Construct(
audioStats->total_output_energy);
local.mTotalSamplesDuration.Construct(
audioStats->total_output_duration);
*/
if (!report->mInboundRtpStreamStats.AppendElement(
std::move(local), fallible)) {
mozalloc_handle_oom(0);
}
});
asVideo.apply([&](auto& aConduit) {
Maybe<webrtc::VideoReceiveStream::Stats> videoStats =
aConduit->GetReceiverStats();
if (videoStats.isNothing()) {
return;
}
if (!videoStats->rtp_stats.last_packet_received_timestamp_ms) {
// By spec: "The lifetime of all RTP monitored objects starts
// when the RTP stream is first used: When the first RTP packet
// is sent or received on the SSRC it represents"
return;
}
// First, fill in remote stat with rtcp sender data, if present.
if (videoStats->rtcp_sender_ntp_timestamp_ms) {
RTCRemoteOutboundRtpStreamStats remote;
constructCommonRemoteOutboundRtpStats(
remote, aConduit->GetTimestampMaker().ConvertNtpToDomTime(
webrtc::Timestamp::Millis(
videoStats->rtcp_sender_ntp_timestamp_ms)));
remote.mPacketsSent.Construct(
videoStats->rtcp_sender_packets_sent);
remote.mBytesSent.Construct(
videoStats->rtcp_sender_octets_sent);
remote.mRemoteTimestamp.Construct(
(webrtc::Timestamp::Millis(
videoStats->rtcp_sender_remote_ntp_timestamp_ms) -
webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970))
.ms());
if (!report->mRemoteOutboundRtpStreamStats.AppendElement(
std::move(remote), fallible)) {
mozalloc_handle_oom(0);
}
}
// Then, fill in local side (with cross-link to remote only if
// present)
RTCInboundRtpStreamStats local;
constructCommonInboundRtpStats(local);
local.mJitter.Construct(
static_cast<double>(videoStats->rtp_stats.jitter) /
webrtc::kVideoPayloadTypeFrequency);
local.mPacketsLost.Construct(videoStats->rtp_stats.packets_lost);
local.mPacketsReceived.Construct(
videoStats->rtp_stats.packet_counter.packets);
local.mPacketsDiscarded.Construct(videoStats->packets_discarded);
local.mDiscardedPackets.Construct(videoStats->packets_discarded);
local.mBytesReceived.Construct(
videoStats->rtp_stats.packet_counter.payload_bytes);
// Fill in packet type statistics
local.mNackCount.Construct(
videoStats->rtcp_packet_type_counts.nack_packets);
local.mFirCount.Construct(
videoStats->rtcp_packet_type_counts.fir_packets);
local.mPliCount.Construct(
videoStats->rtcp_packet_type_counts.pli_packets);
// Lastly, fill in video decoder stats
local.mFramesDecoded.Construct(videoStats->frames_decoded);
local.mFramesPerSecond.Construct(videoStats->decode_frame_rate);
/*
* Potential new stats that are now available upstream.
local.mFrameWidth.Construct(videoStats->width);
local.mFrameheight.Construct(videoStats->height);
if (videoStats->qp_sum) {
local.mQpSum.Construct(*videoStats->qp_sum.value);
}
local.mTotalDecodeTime.Construct(
double(videoStats->total_decode_time_ms) / 1000);
local.mTotalInterFrameDelay.Construct(
videoStats->total_inter_frame_delay);
local.mTotalSquaredInterFrameDelay.Construct(
videoStats->total_squared_inter_frame_delay);
if (videoStats->rtp_stats.last_packet_received_timestamp_ms) {
local.mLastPacketReceiveTimestamp.Construct(
aConduit->GetTimestampMaker().ConvertNtpToDomTime(
webrtc::Timestamp::Millis(
*videoStats->rtp_stats
.last_packet_received_timestamp_ms) +
webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970)));
}
local.mHeaderBytesReceived.Construct(
videoStats->rtp_stats.packet_counter.header_bytes +
videoStats->rtp_stats.packet_counter.padding_bytes);
if (videoStats->estimated_playout_ntp_timestamp_ms) {
local.mEstimatedPlayoutTimestamp.Construct(
aConduit->GetTimestampMaker().ConvertNtpToDomTime(
webrtc::Timestamp::Millis(
*videoStats->estimated_playout_ntp_timestamp_ms)));
}
local.mJitterBufferDelay.Construct(
videoStats->jitter_buffer_delay_seconds);
local.mJitterBufferEmittedCount.Construct(
videoStats->jitter_buffer_emitted_count);
local.mFramesReceived.Construct(
videoStats->frame_counts.key_frames +
videoStats->frame_counts.delta_frames);
// Not including frames dropped in the rendering pipe, which
// is not of webrtc's concern anyway?!
local.mFramesDropped.Construct(videoStats->frames_dropped);
*/
if (!report->mInboundRtpStreamStats.AppendElement(
std::move(local), fallible)) {
mozalloc_handle_oom(0);
}
});
return RTCStatsPromise::CreateAndResolve(std::move(report),
__func__);
})
->Then(
mStsThread, __func__,
[pipeline = mPipeline](UniquePtr<RTCStatsCollection> aReport) {
// Fill in Contributing Source statistics
if (!aReport->mInboundRtpStreamStats.IsEmpty() &&
aReport->mInboundRtpStreamStats[0].mId.WasPassed()) {
pipeline->GetContributingSourceStats(
aReport->mInboundRtpStreamStats[0].mId.Value(),
aReport->mRtpContributingSourceStats);
}
return RTCStatsPromise::CreateAndResolve(std::move(aReport),
__func__);
},
[] {
MOZ_CRASH("Unexpected reject");
return RTCStatsPromise::CreateAndReject(NS_ERROR_UNEXPECTED,
__func__);
}));
if (mJsepTransceiver->mTransport.mComponents) {
promises.AppendElement(mTransportHandler->GetIceStats(
mJsepTransceiver->mTransport.mTransportId,
mPipeline->GetTimestampMaker().GetNow()));
}
return promises;
}
void RTCRtpReceiver::GetContributingSources(
nsTArray<RTCRtpContributingSource>& aSources) {
// Duplicate code...
if (mPipeline && mPipeline->mConduit) {
nsTArray<dom::RTCRtpSourceEntry> sources;
mPipeline->mConduit->GetRtpSources(sources);
sources.RemoveElementsBy([](const dom::RTCRtpSourceEntry& aEntry) {
return aEntry.mSourceType != dom::RTCRtpSourceEntryType::Contributing;
});
aSources.ReplaceElementsAt(0, aSources.Length(), sources.Elements(),
sources.Length());
}
}
void RTCRtpReceiver::GetSynchronizationSources(
nsTArray<dom::RTCRtpSynchronizationSource>& aSources) {
// Duplicate code...
if (mPipeline && mPipeline->mConduit) {
nsTArray<dom::RTCRtpSourceEntry> sources;
mPipeline->mConduit->GetRtpSources(sources);
sources.RemoveElementsBy([](const dom::RTCRtpSourceEntry& aEntry) {
return aEntry.mSourceType != dom::RTCRtpSourceEntryType::Synchronization;
});
aSources.ReplaceElementsAt(0, aSources.Length(), sources.Elements(),
sources.Length());
}
}
nsPIDOMWindowInner* RTCRtpReceiver::GetParentObject() const { return mWindow; }
void RTCRtpReceiver::Shutdown() {
MOZ_ASSERT(NS_IsMainThread());
if (mPipeline) {
mPipeline->Shutdown();
mPipeline = nullptr;
}
mTransceiverImpl = nullptr;
mCallThread = nullptr;
mRtcpByeListener.DisconnectIfExists();
mRtcpTimeoutListener.DisconnectIfExists();
}
void RTCRtpReceiver::UpdateTransport() {
MOZ_ASSERT(NS_IsMainThread());
if (!mHaveSetupTransport) {
mPipeline->SetLevel(mJsepTransceiver->GetLevel());
mHaveSetupTransport = true;
}
UniquePtr<MediaPipelineFilter> filter;
auto const& details = mJsepTransceiver->mRecvTrack.GetNegotiatedDetails();
if (mJsepTransceiver->HasBundleLevel() && details) {
std::vector<webrtc::RtpExtension> extmaps;
details->ForEachRTPHeaderExtension(
[&extmaps](const SdpExtmapAttributeList::Extmap& extmap) {
extmaps.emplace_back(extmap.extensionname, extmap.entry);
});
filter = MakeUnique<MediaPipelineFilter>(extmaps);
// Add remote SSRCs so we can distinguish which RTP packets actually
// belong to this pipeline (also RTCP sender reports).
for (uint32_t ssrc : mJsepTransceiver->mRecvTrack.GetSsrcs()) {
filter->AddRemoteSSRC(ssrc);
}
for (uint32_t ssrc : mJsepTransceiver->mRecvTrack.GetRtxSsrcs()) {
filter->AddRemoteSSRC(ssrc);
}
auto mid = Maybe<std::string>();
if (GetMid() != "") {
mid = Some(GetMid());
}
filter->SetRemoteMediaStreamId(mid);
// Add unique payload types as a last-ditch fallback
auto uniquePts = mJsepTransceiver->mRecvTrack.GetNegotiatedDetails()
->GetUniquePayloadTypes();
for (unsigned char& uniquePt : uniquePts) {
filter->AddUniquePT(uniquePt);
}
}
mPipeline->UpdateTransport_m(mJsepTransceiver->mTransport.mTransportId,
std::move(filter));
}
void RTCRtpReceiver::UpdateConduit() {
mReceiving = false;
Stop();
if (mPipeline->mConduit->type() == MediaSessionConduit::VIDEO) {
UpdateVideoConduit();
} else {
UpdateAudioConduit();
}
if ((mReceiving = mJsepTransceiver->mRecvTrack.GetActive())) {
Start();
}
}
void RTCRtpReceiver::UpdateVideoConduit() {
RefPtr<VideoSessionConduit> conduit =
*mPipeline->mConduit->AsVideoSessionConduit();
// NOTE(pkerr) - this is new behavior. Needed because the
// CreateVideoReceiveStream method of the Call API will assert (in debug)
// and fail if a value is not provided for the remote_ssrc that will be used
// by the far-end sender.
if (!mJsepTransceiver->mRecvTrack.GetSsrcs().empty()) {
MOZ_LOG(gReceiverLog, LogLevel::Debug,
("%s[%s]: %s Setting remote SSRC %u", mPc->GetHandle().c_str(),
GetMid().c_str(), __FUNCTION__,
mJsepTransceiver->mRecvTrack.GetSsrcs().front()));
uint32_t rtxSsrc = mJsepTransceiver->mRecvTrack.GetRtxSsrcs().empty()
? 0
: mJsepTransceiver->mRecvTrack.GetRtxSsrcs().front();
mSsrc = mJsepTransceiver->mRecvTrack.GetSsrcs().front();
mVideoRtxSsrc = rtxSsrc;
// TODO (bug 1423041) once we pay attention to receiving MID's in RTP
// packets (see bug 1405495) we could make this depending on the presence of
// MID in the RTP packets instead of relying on the signaling.
// In any case, do not disable SSRC changes if no SSRCs were negotiated
if (mJsepTransceiver->HasBundleLevel() &&
(!mJsepTransceiver->mRecvTrack.GetNegotiatedDetails() ||
!mJsepTransceiver->mRecvTrack.GetNegotiatedDetails()->GetExt(
webrtc::RtpExtension::kMidUri))) {
mCallThread->Dispatch(
NewRunnableMethod("VideoSessionConduit::DisableSsrcChanges", conduit,
&VideoSessionConduit::DisableSsrcChanges));
}
}
if (mJsepTransceiver->mRecvTrack.GetNegotiatedDetails() &&
mJsepTransceiver->mRecvTrack.GetActive()) {
const auto& details(*mJsepTransceiver->mRecvTrack.GetNegotiatedDetails());
{
std::vector<webrtc::RtpExtension> extmaps;
// @@NG read extmap from track
details.ForEachRTPHeaderExtension(
[&extmaps](const SdpExtmapAttributeList::Extmap& extmap) {
extmaps.emplace_back(extmap.extensionname, extmap.entry);
});
mLocalRtpExtensions = extmaps;
}
std::vector<VideoCodecConfig> configs;
TransceiverImpl::NegotiatedDetailsToVideoCodecConfigs(details, &configs);
if (configs.empty()) {
// TODO: Are we supposed to plumb this error back to JS? This does not
// seem like a failure to set an answer, it just means that codec
// negotiation failed. For now, we're just doing the same thing we do
// if negotiation as a whole failed.
MOZ_LOG(gReceiverLog, LogLevel::Error,
("%s[%s]: %s No video codecs were negotiated (recv).",
mPc->GetHandle().c_str(), GetMid().c_str(), __FUNCTION__));
return;
}
mVideoCodecs = configs;
mVideoRtpRtcpConfig = Some(details.GetRtpRtcpConfig());
}
}
void RTCRtpReceiver::UpdateAudioConduit() {
RefPtr<AudioSessionConduit> conduit =
*mPipeline->mConduit->AsAudioSessionConduit();
if (!mJsepTransceiver->mRecvTrack.GetSsrcs().empty()) {
MOZ_LOG(gReceiverLog, LogLevel::Debug,
("%s[%s]: %s Setting remote SSRC %u", mPc->GetHandle().c_str(),
GetMid().c_str(), __FUNCTION__,
mJsepTransceiver->mRecvTrack.GetSsrcs().front()));
mSsrc = mJsepTransceiver->mRecvTrack.GetSsrcs().front();
// TODO (bug 1423041) once we pay attention to receiving MID's in RTP
// packets (see bug 1405495) we could make this depending on the presence of
// MID in the RTP packets instead of relying on the signaling.
// In any case, do not disable SSRC changes if no SSRCs were negotiated
if (mJsepTransceiver->HasBundleLevel() &&
(!mJsepTransceiver->mRecvTrack.GetNegotiatedDetails() ||
!mJsepTransceiver->mRecvTrack.GetNegotiatedDetails()->GetExt(
webrtc::RtpExtension::kMidUri))) {
mCallThread->Dispatch(
NewRunnableMethod("AudioSessionConduit::DisableSsrcChanges", conduit,
&AudioSessionConduit::DisableSsrcChanges));
}
}
if (mJsepTransceiver->mRecvTrack.GetNegotiatedDetails() &&
mJsepTransceiver->mRecvTrack.GetActive()) {
const auto& details(*mJsepTransceiver->mRecvTrack.GetNegotiatedDetails());
std::vector<AudioCodecConfig> configs;
TransceiverImpl::NegotiatedDetailsToAudioCodecConfigs(details, &configs);
if (configs.empty()) {
// TODO: Are we supposed to plumb this error back to JS? This does not
// seem like a failure to set an answer, it just means that codec
// negotiation failed. For now, we're just doing the same thing we do
// if negotiation as a whole failed.
MOZ_LOG(gReceiverLog, LogLevel::Error,
("%s[%s]: %s No audio codecs were negotiated (recv)",
mPc->GetHandle().c_str(), GetMid().c_str(), __FUNCTION__));
return;
}
// Ensure conduit knows about extensions prior to creating streams
{
std::vector<webrtc::RtpExtension> extmaps;
// @@NG read extmap from track
details.ForEachRTPHeaderExtension(
[&extmaps](const SdpExtmapAttributeList::Extmap& extmap) {
extmaps.emplace_back(extmap.extensionname, extmap.entry);
});
mLocalRtpExtensions = extmaps;
}
mAudioCodecs = configs;
}
}
void RTCRtpReceiver::Stop() {
if (mPipeline) {
mPipeline->Stop();
}
}
void RTCRtpReceiver::Start() {
mPipeline->Start();
mHaveStartedReceiving = true;
}
bool RTCRtpReceiver::HasTrack(const dom::MediaStreamTrack* aTrack) const {
return !aTrack || (mTrack == aTrack);
}
void RTCRtpReceiver::UpdateStreams(StreamAssociationChanges* aChanges) {
// We don't sort and use set_difference, because we need to report the
// added/removed streams in the order that they appear in the SDP.
std::set<std::string> newIds(
mJsepTransceiver->mRecvTrack.GetStreamIds().begin(),
mJsepTransceiver->mRecvTrack.GetStreamIds().end());
MOZ_ASSERT(mJsepTransceiver->mRecvTrack.GetRemoteSetSendBit() ||
newIds.empty());
bool needsTrackEvent = false;
for (const auto& id : mStreamIds) {
if (!newIds.count(id)) {
aChanges->mStreamAssociationsRemoved.push_back({mTrack, id});
}
}
std::set<std::string> oldIds(mStreamIds.begin(), mStreamIds.end());
for (const auto& id : mJsepTransceiver->mRecvTrack.GetStreamIds()) {
if (!oldIds.count(id)) {
needsTrackEvent = true;
aChanges->mStreamAssociationsAdded.push_back({mTrack, id});
}
}
mStreamIds = mJsepTransceiver->mRecvTrack.GetStreamIds();
if (mRemoteSetSendBit != mJsepTransceiver->mRecvTrack.GetRemoteSetSendBit()) {
mRemoteSetSendBit = mJsepTransceiver->mRecvTrack.GetRemoteSetSendBit();
if (mRemoteSetSendBit) {
needsTrackEvent = true;
} else {
aChanges->mTracksToMute.push_back(mTrack);
}
}
if (needsTrackEvent) {
aChanges->mTrackEvents.push_back({this, mStreamIds});
}
}
// test-only: adds fake CSRCs and audio data
void RTCRtpReceiver::MozInsertAudioLevelForContributingSource(
const uint32_t aSource, const DOMHighResTimeStamp aTimestamp,
const uint32_t aRtpTimestamp, const bool aHasLevel, const uint8_t aLevel) {
if (!mPipeline || mPipeline->IsVideo() || !mPipeline->mConduit) {
return;
}
mPipeline->mConduit->InsertAudioLevelForContributingSource(
aSource, aTimestamp, aRtpTimestamp, aHasLevel, aLevel);
}
void RTCRtpReceiver::OnRtcpBye() { SetReceiveTrackMuted(true); }
void RTCRtpReceiver::OnRtcpTimeout() { SetReceiveTrackMuted(true); }
void RTCRtpReceiver::SetReceiveTrackMuted(bool aMuted) {
if (mTrack) {
// This sets the muted state for mTrack and all its clones.
static_cast<RemoteTrackSource&>(mTrack->GetSource()).SetMuted(aMuted);
}
}
std::string RTCRtpReceiver::GetMid() const {
if (mJsepTransceiver->IsAssociated()) {
return mJsepTransceiver->GetMid();
}
return std::string();
}
} // namespace mozilla::dom
#undef LOGTAG
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