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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,
#include "RTCRtpSender.h"
#include <stdint.h>
#include <vector>
#include <string>
#include <algorithm>
#include <utility>
#include <iterator>
#include <set>
#include <sstream>
#include "system_wrappers/include/clock.h"
#include "call/call.h"
#include "api/rtp_parameters.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "api/video_codecs/video_codec.h"
#include "api/video/video_codec_constants.h"
#include "call/audio_send_stream.h"
#include "call/video_send_stream.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "nsPIDOMWindow.h"
#include "nsString.h"
#include "MainThreadUtils.h"
#include "nsCOMPtr.h"
#include "nsContentUtils.h"
#include "nsCycleCollectionParticipant.h"
#include "nsDebug.h"
#include "nsISupports.h"
#include "nsLiteralString.h"
#include "nsStringFwd.h"
#include "nsTArray.h"
#include "nsThreadUtils.h"
#include "nsWrapperCache.h"
#include "mozilla/AbstractThread.h"
#include "mozilla/AlreadyAddRefed.h"
#include "mozilla/Assertions.h"
#include "mozilla/Attributes.h"
#include "mozilla/fallible.h"
#include "mozilla/Logging.h"
#include "mozilla/mozalloc_oom.h"
#include "mozilla/MozPromise.h"
#include "mozilla/OwningNonNull.h"
#include "mozilla/RefPtr.h"
#include "mozilla/StateWatching.h"
#include "mozilla/UniquePtr.h"
#include "mozilla/Unused.h"
#include "mozilla/StateMirroring.h"
#include "mozilla/Maybe.h"
#include "mozilla/dom/MediaStreamTrack.h"
#include "mozilla/dom/Promise.h"
#include "mozilla/dom/RTCRtpScriptTransform.h"
#include "mozilla/dom/VideoStreamTrack.h"
#include "mozilla/dom/RTCRtpSenderBinding.h"
#include "mozilla/dom/MediaStreamTrackBinding.h"
#include "mozilla/dom/Nullable.h"
#include "mozilla/dom/RTCRtpParametersBinding.h"
#include "mozilla/dom/RTCStatsReportBinding.h"
#include "mozilla/glean/GleanMetrics.h"
#include "js/RootingAPI.h"
#include "jsep/JsepTransceiver.h"
#include "RTCStatsReport.h"
#include "RTCRtpTransceiver.h"
#include "PeerConnectionImpl.h"
#include "libwebrtcglue/CodecConfig.h"
#include "libwebrtcglue/MediaConduitControl.h"
#include "libwebrtcglue/MediaConduitInterface.h"
#include "sdp/SdpAttribute.h"
#include "sdp/SdpEnum.h"
namespace mozilla::dom {
LazyLogModule gSenderLog("RTCRtpSender");
NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(RTCRtpSender)
NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(RTCRtpSender)
tmp->Unlink();
NS_IMPL_CYCLE_COLLECTION_UNLINK_PRESERVED_WRAPPER
NS_IMPL_CYCLE_COLLECTION_UNLINK_END
NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN(RTCRtpSender)
NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mWindow, mPc, mSenderTrack, mTransceiver,
mStreams, mTransform, mDtmf)
NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
NS_IMPL_CYCLE_COLLECTING_ADDREF(RTCRtpSender)
NS_IMPL_CYCLE_COLLECTING_RELEASE(RTCRtpSender)
NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(RTCRtpSender)
NS_WRAPPERCACHE_INTERFACE_MAP_ENTRY
NS_INTERFACE_MAP_ENTRY(nsISupports)
NS_INTERFACE_MAP_END
#define INIT_CANONICAL(name, val) \
name(AbstractThread::MainThread(), val, "RTCRtpSender::" #name " (Canonical)")
RTCRtpSender::RTCRtpSender(nsPIDOMWindowInner* aWindow, PeerConnectionImpl* aPc,
MediaTransportHandler* aTransportHandler,
AbstractThread* aCallThread,
nsISerialEventTarget* aStsThread,
MediaSessionConduit* aConduit,
dom::MediaStreamTrack* aTrack,
const Sequence<RTCRtpEncodingParameters>& aEncodings,
RTCRtpTransceiver* aTransceiver)
: mWatchManager(this, AbstractThread::MainThread()),
mWindow(aWindow),
mPc(aPc),
mSenderTrack(aTrack),
mTransportHandler(aTransportHandler),
mTransceiver(aTransceiver),
INIT_CANONICAL(mSsrcs, Ssrcs()),
INIT_CANONICAL(mVideoRtxSsrcs, Ssrcs()),
INIT_CANONICAL(mLocalRtpExtensions, RtpExtList()),
INIT_CANONICAL(mAudioCodec, Nothing()),
INIT_CANONICAL(mVideoCodec, Nothing()),
INIT_CANONICAL(mVideoRtpRtcpConfig, Nothing()),
INIT_CANONICAL(mVideoCodecMode, webrtc::VideoCodecMode::kRealtimeVideo),
INIT_CANONICAL(mCname, std::string()),
INIT_CANONICAL(mTransmitting, false),
INIT_CANONICAL(mFrameTransformerProxy, nullptr) {
mPipeline = MediaPipelineTransmit::Create(
mPc->GetHandle(), aTransportHandler, aCallThread, aStsThread,
aConduit->type() == MediaSessionConduit::VIDEO, aConduit);
mPipeline->InitControl(this);
if (aConduit->type() == MediaSessionConduit::AUDIO) {
mDtmf = new RTCDTMFSender(aWindow, mTransceiver);
}
mPipeline->SetTrack(mSenderTrack);
mozilla::glean::rtcrtpsender::count.Add(1);
if (mPc->ShouldAllowOldSetParameters()) {
mAllowOldSetParameters = true;
mozilla::glean::rtcrtpsender::count_setparameters_compat.Add(1);
}
if (aEncodings.Length()) {
// This sender was created by addTransceiver with sendEncodings.
mParameters.mEncodings = aEncodings;
mSimulcastEnvelopeSet = true;
mozilla::glean::rtcrtpsender::used_sendencodings.AddToNumerator(1);
} else {
// This sender was created by addTrack, sRD(offer), or addTransceiver
// without sendEncodings.
RTCRtpEncodingParameters defaultEncoding;
defaultEncoding.mActive = true;
if (aConduit->type() == MediaSessionConduit::VIDEO) {
defaultEncoding.mScaleResolutionDownBy.Construct(1.0f);
}
Unused << mParameters.mEncodings.AppendElement(defaultEncoding, fallible);
UpdateRestorableEncodings(mParameters.mEncodings);
MaybeGetJsepRids();
}
mParameters.mCodecs.Construct();
if (mDtmf) {
mWatchManager.Watch(mTransmitting, &RTCRtpSender::UpdateDtmfSender);
}
}
#undef INIT_CANONICAL
RTCRtpSender::~RTCRtpSender() = default;
JSObject* RTCRtpSender::WrapObject(JSContext* aCx,
JS::Handle<JSObject*> aGivenProto) {
return RTCRtpSender_Binding::Wrap(aCx, this, aGivenProto);
}
RTCDtlsTransport* RTCRtpSender::GetTransport() const {
if (!mTransceiver) {
return nullptr;
}
return mTransceiver->GetDtlsTransport();
}
RTCDTMFSender* RTCRtpSender::GetDtmf() const { return mDtmf; }
already_AddRefed<Promise> RTCRtpSender::GetStats(ErrorResult& aError) {
RefPtr<Promise> promise = MakePromise(aError);
if (aError.Failed()) {
return nullptr;
}
if (NS_WARN_IF(!mPipeline)) {
// (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();
}
if (!mSenderTrack) {
promise->MaybeResolve(new RTCStatsReport(mWindow));
return promise.forget();
}
mTransceiver->ChainToDomPromiseWithCodecStats(GetStatsInternal(), promise);
return promise.forget();
}
nsTArray<RefPtr<dom::RTCStatsPromise>> RTCRtpSender::GetStatsInternal(
bool aSkipIceStats) {
MOZ_ASSERT(NS_IsMainThread());
nsTArray<RefPtr<RTCStatsPromise>> promises(2);
if (!mSenderTrack || !mPipeline) {
return promises;
}
nsAutoString trackName;
if (auto track = mPipeline->GetTrack()) {
track->GetId(trackName);
}
{
// Add bandwidth estimation stats
promises.AppendElement(InvokeAsync(
mPipeline->mCallThread, __func__,
[conduit = mPipeline->mConduit, trackName]() mutable {
auto report = MakeUnique<dom::RTCStatsCollection>();
Maybe<webrtc::Call::Stats> stats = conduit->GetCallStats();
stats.apply([&](const auto aStats) {
dom::RTCBandwidthEstimationInternal bw;
bw.mTrackIdentifier = trackName;
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(
mPipeline->mCallThread, __func__, [pipeline = mPipeline, trackName] {
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()));
for (uint32_t ssrc : pipeline->mConduit->GetLocalSSRCs()) {
nsString localId = u"outbound_rtp_"_ns + idstr + u"_"_ns;
localId.AppendInt(ssrc);
nsString remoteId;
Maybe<uint16_t> base_seq =
pipeline->mConduit->RtpSendBaseSeqFor(ssrc);
auto constructCommonRemoteInboundRtpStats =
[&](RTCRemoteInboundRtpStreamStats& aRemote,
const webrtc::ReportBlockData& aRtcpData) {
remoteId = u"outbound_rtcp_"_ns + idstr + u"_"_ns;
remoteId.AppendInt(ssrc);
aRemote.mTimestamp.Construct(
RTCStatsTimestamp::FromNtp(
pipeline->GetTimestampMaker(),
webrtc::Timestamp::Micros(
aRtcpData.report_block_timestamp_utc().us()) +
webrtc::TimeDelta::Seconds(webrtc::kNtpJan1970))
.ToDom());
aRemote.mId.Construct(remoteId);
aRemote.mType.Construct(RTCStatsType::Remote_inbound_rtp);
aRemote.mSsrc = ssrc;
aRemote.mKind = kind;
aRemote.mMediaType.Construct(
kind); // mediaType is the old name for kind.
aRemote.mLocalId.Construct(localId);
if (base_seq) {
if (aRtcpData.extended_highest_sequence_number() <
*base_seq) {
aRemote.mPacketsReceived.Construct(0);
} else {
aRemote.mPacketsReceived.Construct(
aRtcpData.extended_highest_sequence_number() -
aRtcpData.cumulative_lost() - *base_seq + 1);
}
}
};
auto constructCommonOutboundRtpStats =
[&](RTCOutboundRtpStreamStats& aLocal) {
aLocal.mSsrc = ssrc;
aLocal.mTimestamp.Construct(
pipeline->GetTimestampMaker().GetNow().ToDom());
aLocal.mId.Construct(localId);
aLocal.mType.Construct(RTCStatsType::Outbound_rtp);
aLocal.mKind = kind;
aLocal.mMediaType.Construct(
kind); // mediaType is the old name for kind.
if (remoteId.Length()) {
aLocal.mRemoteId.Construct(remoteId);
}
};
auto constructCommonMediaSourceStats =
[&](RTCMediaSourceStats& aStats) {
nsString id = u"mediasource_"_ns + idstr + trackName;
aStats.mTimestamp.Construct(
pipeline->GetTimestampMaker().GetNow().ToDom());
aStats.mId.Construct(id);
aStats.mType.Construct(RTCStatsType::Media_source);
aStats.mTrackIdentifier = trackName;
aStats.mKind = kind;
};
asAudio.apply([&](auto& aConduit) {
Maybe<webrtc::AudioSendStream::Stats> audioStats =
aConduit->GetSenderStats();
if (audioStats.isNothing()) {
return;
}
if (audioStats->packets_sent == 0) {
// 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 receiver data, if present.
// ReceiverReports have less information than SenderReports, so fill
// in what we can.
Maybe<webrtc::ReportBlockData> reportBlockData;
{
if (const auto remoteSsrc = aConduit->GetRemoteSSRC();
remoteSsrc) {
for (auto& data : audioStats->report_block_datas) {
if (data.source_ssrc() == ssrc &&
data.sender_ssrc() == *remoteSsrc) {
reportBlockData.emplace(data);
break;
}
}
}
}
reportBlockData.apply([&](auto& aReportBlockData) {
RTCRemoteInboundRtpStreamStats remote;
constructCommonRemoteInboundRtpStats(remote, aReportBlockData);
if (audioStats->jitter_ms >= 0) {
remote.mJitter.Construct(audioStats->jitter_ms / 1000.0);
}
if (audioStats->packets_lost >= 0) {
remote.mPacketsLost.Construct(audioStats->packets_lost);
}
if (audioStats->rtt_ms >= 0) {
remote.mRoundTripTime.Construct(
static_cast<double>(audioStats->rtt_ms) / 1000.0);
}
remote.mFractionLost.Construct(audioStats->fraction_lost);
remote.mTotalRoundTripTime.Construct(
double(aReportBlockData.sum_rtts().ms()) / 1000);
remote.mRoundTripTimeMeasurements.Construct(
aReportBlockData.num_rtts());
if (!report->mRemoteInboundRtpStreamStats.AppendElement(
std::move(remote), fallible)) {
mozalloc_handle_oom(0);
}
});
// Then, fill in local side (with cross-link to remote only if
// present)
RTCOutboundRtpStreamStats local;
constructCommonOutboundRtpStats(local);
local.mPacketsSent.Construct(audioStats->packets_sent);
local.mBytesSent.Construct(audioStats->payload_bytes_sent);
local.mNackCount.Construct(
audioStats->rtcp_packet_type_counts.nack_packets);
local.mHeaderBytesSent.Construct(
audioStats->header_and_padding_bytes_sent);
local.mRetransmittedPacketsSent.Construct(
audioStats->retransmitted_packets_sent);
local.mRetransmittedBytesSent.Construct(
audioStats->retransmitted_bytes_sent);
/*
* Potential new stats that are now available upstream.
* Note: when we last tried exposing this we were getting
* targetBitrate for audio was ending up as 0. We did not
* investigate why.
local.mTargetBitrate.Construct(audioStats->target_bitrate_bps);
*/
if (!report->mOutboundRtpStreamStats.AppendElement(std::move(local),
fallible)) {
mozalloc_handle_oom(0);
}
RTCMediaSourceStats mediaSourceStats;
constructCommonMediaSourceStats(mediaSourceStats);
if (!report->mMediaSourceStats.AppendElement(
std::move(mediaSourceStats), fallible)) {
mozalloc_handle_oom(0);
}
});
asVideo.apply([&](auto& aConduit) {
Maybe<webrtc::VideoSendStream::Stats> videoStats =
aConduit->GetSenderStats();
if (videoStats.isNothing()) {
return;
}
Maybe<webrtc::VideoSendStream::StreamStats> streamStats;
auto kv = videoStats->substreams.find(ssrc);
if (kv != videoStats->substreams.end()) {
streamStats = Some(kv->second);
}
if (!streamStats) {
// 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;
}
aConduit->GetAssociatedLocalRtxSSRC(ssrc).apply(
[&](const auto rtxSsrc) {
auto kv = videoStats->substreams.find(rtxSsrc);
if (kv != videoStats->substreams.end()) {
streamStats->rtp_stats.Add(kv->second.rtp_stats);
}
});
if (streamStats->rtp_stats.first_packet_time ==
webrtc::Timestamp::PlusInfinity()) {
return;
}
// First, fill in remote stat with rtcp receiver data, if present.
// ReceiverReports have less information than SenderReports, so fill
// in what we can.
if (streamStats->report_block_data) {
const webrtc::ReportBlockData& rtcpReportData =
*streamStats->report_block_data;
RTCRemoteInboundRtpStreamStats remote;
remote.mJitter.Construct(
static_cast<double>(rtcpReportData.jitter()) /
webrtc::kVideoPayloadTypeFrequency);
remote.mPacketsLost.Construct(rtcpReportData.cumulative_lost());
if (rtcpReportData.has_rtt()) {
remote.mRoundTripTime.Construct(
static_cast<double>(rtcpReportData.last_rtt().ms()) /
1000.0);
}
constructCommonRemoteInboundRtpStats(remote, rtcpReportData);
remote.mTotalRoundTripTime.Construct(
streamStats->report_block_data->sum_rtts().ms() / 1000.0);
remote.mFractionLost.Construct(
static_cast<float>(rtcpReportData.fraction_lost_raw()) /
(1 << 8));
remote.mRoundTripTimeMeasurements.Construct(
streamStats->report_block_data->num_rtts());
if (!report->mRemoteInboundRtpStreamStats.AppendElement(
std::move(remote), fallible)) {
mozalloc_handle_oom(0);
}
}
// Then, fill in local side (with cross-link to remote only if
// present)
RTCOutboundRtpStreamStats local;
constructCommonOutboundRtpStats(local);
local.mPacketsSent.Construct(
streamStats->rtp_stats.transmitted.packets);
local.mBytesSent.Construct(
streamStats->rtp_stats.transmitted.payload_bytes);
local.mNackCount.Construct(
streamStats->rtcp_packet_type_counts.nack_packets);
local.mFirCount.Construct(
streamStats->rtcp_packet_type_counts.fir_packets);
local.mPliCount.Construct(
streamStats->rtcp_packet_type_counts.pli_packets);
local.mFramesEncoded.Construct(streamStats->frames_encoded);
if (streamStats->qp_sum) {
local.mQpSum.Construct(*streamStats->qp_sum);
}
local.mHeaderBytesSent.Construct(
streamStats->rtp_stats.transmitted.header_bytes +
streamStats->rtp_stats.transmitted.padding_bytes);
local.mRetransmittedPacketsSent.Construct(
streamStats->rtp_stats.retransmitted.packets);
local.mRetransmittedBytesSent.Construct(
streamStats->rtp_stats.retransmitted.payload_bytes);
local.mTotalEncodedBytesTarget.Construct(
videoStats->total_encoded_bytes_target);
local.mFrameWidth.Construct(streamStats->width);
local.mFrameHeight.Construct(streamStats->height);
local.mFramesPerSecond.Construct(streamStats->encode_frame_rate);
local.mFramesSent.Construct(streamStats->frames_encoded);
local.mHugeFramesSent.Construct(streamStats->huge_frames_sent);
local.mTotalEncodeTime.Construct(
double(streamStats->total_encode_time_ms) / 1000.);
/*
* Potential new stats that are now available upstream.
local.mTargetBitrate.Construct(videoStats->target_media_bitrate_bps);
*/
if (!report->mOutboundRtpStreamStats.AppendElement(std::move(local),
fallible)) {
mozalloc_handle_oom(0);
}
RTCVideoSourceStats videoSourceStats;
constructCommonMediaSourceStats(videoSourceStats);
// webrtc::VideoSendStream::Stats does not have width/height. We
// might be able to get this somewhere else?
// videoStats->frames is not documented, but looking at the
// implementation it appears to be the number of frames inputted to
// the encoder, which ought to work.
videoSourceStats.mFrames.Construct(videoStats->frames);
videoSourceStats.mFramesPerSecond.Construct(
videoStats->input_frame_rate);
auto resolution = aConduit->GetLastResolution();
resolution.apply(
[&](const VideoSessionConduit::Resolution& aResolution) {
videoSourceStats.mWidth.Construct(aResolution.width);
videoSourceStats.mHeight.Construct(aResolution.height);
});
if (!report->mVideoSourceStats.AppendElement(
std::move(videoSourceStats), fallible)) {
mozalloc_handle_oom(0);
}
});
}
return RTCStatsPromise::CreateAndResolve(std::move(report), __func__);
}));
if (!aSkipIceStats && GetJsepTransceiver().mTransport.mComponents) {
promises.AppendElement(mTransportHandler->GetIceStats(
GetJsepTransceiver().mTransport.mTransportId,
mPipeline->GetTimestampMaker().GetNow().ToDom()));
}
return promises;
}
void RTCRtpSender::GetCapabilities(const GlobalObject&, const nsAString& aKind,
Nullable<dom::RTCRtpCapabilities>& aResult) {
PeerConnectionImpl::GetCapabilities(aKind, aResult, sdp::Direction::kSend);
}
void RTCRtpSender::WarnAboutBadSetParameters(const nsCString& aError) {
nsCString warning(
"WARNING! Invalid setParameters call detected! The good news? Firefox "
"supports sendEncodings in addTransceiver now, so we ask that you switch "
"over to using the parameters code you use for other browsers. Thank you "
"for your patience and support. The specific error was: ");
warning += aError;
mPc->SendWarningToConsole(warning);
}
nsCString RTCRtpSender::GetEffectiveTLDPlus1() const {
return mPc->GetEffectiveTLDPlus1();
}
already_AddRefed<Promise> RTCRtpSender::SetParameters(
const dom::RTCRtpSendParameters& aParameters, ErrorResult& aError) {
dom::RTCRtpSendParameters paramsCopy(aParameters);
// When the setParameters method is called, the user agent MUST run the
// following steps:
// Let parameters be the method's first argument.
// Let sender be the RTCRtpSender object on which setParameters is invoked.
// Let transceiver be the RTCRtpTransceiver object associated with sender
// (i.e.sender is transceiver.[[Sender]]).
RefPtr<dom::Promise> p = MakePromise(aError);
if (aError.Failed()) {
return nullptr;
}
if (mPc->IsClosed()) {
p->MaybeRejectWithInvalidStateError("Peer connection is closed");
return p.forget();
}
// If transceiver.[[Stopping]] is true, return a promise rejected with a newly
// created InvalidStateError.
if (mTransceiver->Stopping()) {
p->MaybeRejectWithInvalidStateError(
"This sender's transceiver is stopping/stopped");
return p.forget();
}
// If sender.[[LastReturnedParameters]] is null, return a promise rejected
// with a newly created InvalidStateError.
if (!mLastReturnedParameters.isSome()) {
nsCString error;
if (mLastTransactionId.isSome() && paramsCopy.mTransactionId.WasPassed() &&
*mLastTransactionId == paramsCopy.mTransactionId.Value()) {
error =
"Event loop was relinquished between getParameters and setParameters "
"calls";
} else {
error = "Cannot call setParameters without first calling getParameters";
}
if (mAllowOldSetParameters) {
if (!mHaveWarnedBecauseNoGetParameters) {
mHaveWarnedBecauseNoGetParameters = true;
mozilla::glean::rtcrtpsender_setparameters::warn_no_getparameters
.AddToNumerator(1);
}
WarnAboutBadSetParameters(error);
} else {
if (!mHaveFailedBecauseNoGetParameters) {
mHaveFailedBecauseNoGetParameters = true;
mozilla::glean::rtcrtpsender_setparameters::fail_no_getparameters
.AddToNumerator(1);
}
p->MaybeRejectWithInvalidStateError(error);
return p.forget();
}
}
// According to the spec, our consistency checking is based on
// [[LastReturnedParameters]], but if we're letting
// [[LastReturnedParameters]]==null slide, we still want to do
// consistency checking on _something_ so we can warn implementers if they
// are messing that up also. Just find something, _anything_, to do that
// checking with.
Maybe<RTCRtpSendParameters> oldParams;
if (mAllowOldSetParameters) {
if (mLastReturnedParameters.isSome()) {
oldParams = mLastReturnedParameters;
} else if (mPendingParameters.isSome()) {
oldParams = mPendingParameters;
} else {
oldParams = Some(mParameters);
}
MOZ_ASSERT(oldParams.isSome());
} else {
oldParams = mLastReturnedParameters;
}
MOZ_ASSERT(oldParams.isSome());
// Validate parameters by running the following steps:
// Let encodings be parameters.encodings.
// Let codecs be parameters.codecs.
// Let N be the number of RTCRtpEncodingParameters stored in
// sender.[[SendEncodings]].
// If any of the following conditions are met,
// return a promise rejected with a newly created InvalidModificationError:
bool pendingRidChangeFromCompatMode = false;
// encodings.length is different from N.
if (paramsCopy.mEncodings.Length() != oldParams->mEncodings.Length()) {
nsCString error("Cannot change the number of encodings with setParameters");
if (!mAllowOldSetParameters) {
if (!mHaveFailedBecauseEncodingCountChange) {
mHaveFailedBecauseEncodingCountChange = true;
mozilla::glean::rtcrtpsender_setparameters::fail_length_changed
.AddToNumerator(1);
}
p->MaybeRejectWithInvalidModificationError(error);
return p.forget();
}
// Make sure we don't use the old rids in SyncToJsep while we wait for the
// queued task below to update mParameters.
pendingRidChangeFromCompatMode = true;
mSimulcastEnvelopeSet = true;
if (!mHaveWarnedBecauseEncodingCountChange) {
mHaveWarnedBecauseEncodingCountChange = true;
mozilla::glean::rtcrtpsender_setparameters::warn_length_changed
.AddToNumerator(1);
}
WarnAboutBadSetParameters(error);
} else {
// encodings has been re-ordered.
for (size_t i = 0; i < paramsCopy.mEncodings.Length(); ++i) {
const auto& oldEncoding = oldParams->mEncodings[i];
const auto& newEncoding = paramsCopy.mEncodings[i];
if (oldEncoding.mRid != newEncoding.mRid) {
nsCString error("Cannot change rid, or reorder encodings");
if (!mHaveFailedBecauseRidChange) {
mHaveFailedBecauseRidChange = true;
mozilla::glean::rtcrtpsender_setparameters::fail_rid_changed
.AddToNumerator(1);
}
p->MaybeRejectWithInvalidModificationError(error);
return p.forget();
}
}
}
// setParameters style.
if (!paramsCopy.mTransactionId.WasPassed()) {
nsCString error("transactionId is not set!");
if (!mAllowOldSetParameters) {
if (!mHaveFailedBecauseNoTransactionId) {
mHaveFailedBecauseNoTransactionId = true;
mozilla::glean::rtcrtpsender_setparameters::fail_no_transactionid
.AddToNumerator(1);
}
p->MaybeRejectWithTypeError(error);
return p.forget();
}
if (!mHaveWarnedBecauseNoTransactionId) {
mHaveWarnedBecauseNoTransactionId = true;
mozilla::glean::rtcrtpsender_setparameters::warn_no_transactionid
.AddToNumerator(1);
}
WarnAboutBadSetParameters(error);
} else if (oldParams->mTransactionId.WasPassed() &&
oldParams->mTransactionId != paramsCopy.mTransactionId) {
// Any parameter in parameters is marked as a Read-only parameter (such as
// RID) and has a value that is different from the corresponding parameter
// value in sender.[[LastReturnedParameters]]. Note that this also applies
// to transactionId.
// Don't throw this error if letting the "stale getParameters" case slide.
nsCString error(
"Cannot change transaction id: call getParameters, modify the result, "
"and then call setParameters");
if (!mHaveFailedBecauseStaleTransactionId) {
mHaveFailedBecauseStaleTransactionId = true;
mozilla::glean::rtcrtpsender_setparameters::fail_stale_transactionid
.AddToNumerator(1);
}
p->MaybeRejectWithInvalidModificationError(error);
return p.forget();
}
// This could conceivably happen if we are allowing the old setParameters
// behavior.
if (!paramsCopy.mEncodings.Length()) {
nsCString error("Cannot set an empty encodings array");
if (!mAllowOldSetParameters) {
if (!mHaveFailedBecauseNoEncodings) {
mHaveFailedBecauseNoEncodings = true;
mozilla::glean::rtcrtpsender_setparameters::fail_no_encodings
.AddToNumerator(1);
}
p->MaybeRejectWithInvalidModificationError(error);
return p.forget();
}
// TODO: Add some warning telemetry here
WarnAboutBadSetParameters(error);
// Just don't do this; it's stupid.
paramsCopy.mEncodings = oldParams->mEncodings;
}
if (!(oldParams->mCodecs == paramsCopy.mCodecs)) {
nsCString error("RTCRtpParameters.codecs is a read-only parameter");
if (!mAllowOldSetParameters) {
p->MaybeRejectWithInvalidModificationError(error);
return p.forget();
}
WarnAboutBadSetParameters(error);
}
// TODO: Verify remaining read-only parameters
// CheckAndRectifyEncodings handles the following steps:
// If transceiver kind is "audio", remove the scaleResolutionDownBy member
// from all encodings that contain one.
//
// If transceiver kind is "video", and any encoding in encodings contains a
// scaleResolutionDownBy member whose value is less than 1.0, return a
// promise rejected with a newly created RangeError.
//
// Verify that each encoding in encodings has a maxFramerate member whose
// value is greater than or equal to 0.0. If one of the maxFramerate values
// does not meet this requirement, return a promise rejected with a newly
// created RangeError.
ErrorResult rv;
CheckAndRectifyEncodings(paramsCopy.mEncodings, mTransceiver->IsVideo(), rv);
if (rv.Failed()) {
if (!mHaveFailedBecauseOtherError) {
mHaveFailedBecauseOtherError = true;
mozilla::glean::rtcrtpsender_setparameters::fail_other.AddToNumerator(1);
}
p->MaybeReject(std::move(rv));
return p.forget();
}
// If transceiver kind is "video", then for each encoding in encodings that
// doesn't contain a scaleResolutionDownBy member, add a
// scaleResolutionDownBy member with the value 1.0.
if (mTransceiver->IsVideo()) {
for (auto& encoding : paramsCopy.mEncodings) {
if (!encoding.mScaleResolutionDownBy.WasPassed()) {
encoding.mScaleResolutionDownBy.Construct(1.0);
}
}
}
// Let p be a new promise. (see above)
// In parallel, configure the media stack to use parameters to transmit
// sender.[[SenderTrack]].
// Right now this is infallible. That may change someday.
// We need to put this in a member variable, since MaybeUpdateConduit needs it
// This also allows PeerConnectionImpl to detect when there is a pending
// setParameters, which has implcations for the handling of
// setRemoteDescription.
mPendingRidChangeFromCompatMode = pendingRidChangeFromCompatMode;
mPendingParameters = Some(paramsCopy);
uint32_t serialNumber = ++mNumSetParametersCalls;
MaybeUpdateConduit();
// If the media stack is successfully configured with parameters,
// queue a task to run the following steps:
GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction(
__func__,
[this, self = RefPtr<RTCRtpSender>(this), p, paramsCopy, serialNumber] {
// Set sender.[[LastReturnedParameters]] to null.
mLastReturnedParameters = Nothing();
// Set sender.[[SendEncodings]] to parameters.encodings.
mParameters.mEncodings = paramsCopy.mEncodings;
UpdateRestorableEncodings(mParameters.mEncodings);
// Only clear mPendingParameters if it matches; there could have been
// back-to-back calls to setParameters, and we only want to clear this
// if no subsequent setParameters is pending.
if (serialNumber == mNumSetParametersCalls) {
mPendingParameters = Nothing();
// Ok, nothing has called SyncToJsep while this async task was
// pending. No need for special handling anymore.
mPendingRidChangeFromCompatMode = false;
}
MOZ_ASSERT(mParameters.mEncodings.Length());
// Resolve p with undefined.
p->MaybeResolveWithUndefined();
}));
// Return p.
return p.forget();
}
// static
void RTCRtpSender::CheckAndRectifyEncodings(
Sequence<RTCRtpEncodingParameters>& aEncodings, bool aVideo,
ErrorResult& aRv) {
// If any encoding contains a rid member whose value does not conform to the
// grammar requirements specified in Section 10 of [RFC8851], throw a
// TypeError.
for (const auto& encoding : aEncodings) {
if (encoding.mRid.WasPassed()) {
std::string utf8Rid = NS_ConvertUTF16toUTF8(encoding.mRid.Value()).get();
std::string error;
if (!SdpRidAttributeList::CheckRidValidity(utf8Rid, &error)) {
aRv.ThrowTypeError(nsCString(error));
return;
}
if (utf8Rid.size() > SdpRidAttributeList::kMaxRidLength) {
std::ostringstream ss;
ss << "Rid can be at most " << SdpRidAttributeList::kMaxRidLength
<< " characters long (due to internal limitations)";
aRv.ThrowTypeError(nsCString(ss.str()));
return;
}
}
}
if (aEncodings.Length() > 1) {
// If some but not all encodings contain a rid member, throw a TypeError.
// rid must be set if there is more than one encoding
// NOTE: Since rid is read-only, and the number of encodings cannot grow,
// this should never happen in setParameters.
for (const auto& encoding : aEncodings) {
if (!encoding.mRid.WasPassed()) {
aRv.ThrowTypeError("Missing rid");
return;
}
}
// If any encoding contains a rid member whose value is the same as that of
// a rid contained in another encoding in sendEncodings, throw a TypeError.
// NOTE: Since rid is read-only, and the number of encodings cannot grow,
// this should never happen in setParameters.
std::set<nsString> uniqueRids;
for (const auto& encoding : aEncodings) {
if (uniqueRids.count(encoding.mRid.Value())) {
aRv.ThrowTypeError("Duplicate rid");
return;
}
uniqueRids.insert(encoding.mRid.Value());
}
}
// If kind is "audio", remove the scaleResolutionDownBy member from all
// encodings that contain one.
if (!aVideo) {
for (auto& encoding : aEncodings) {
if (encoding.mScaleResolutionDownBy.WasPassed()) {
encoding.mScaleResolutionDownBy.Reset();
}
if (encoding.mMaxFramerate.WasPassed()) {
encoding.mMaxFramerate.Reset();
}
}
}
// If any encoding contains a scaleResolutionDownBy member whose value is
// less than 1.0, throw a RangeError.
for (const auto& encoding : aEncodings) {
if (encoding.mScaleResolutionDownBy.WasPassed()) {
if (encoding.mScaleResolutionDownBy.Value() < 1.0f) {
aRv.ThrowRangeError("scaleResolutionDownBy must be >= 1.0");
return;
}
}
}
// Verify that the value of each maxFramerate member in sendEncodings that is
// defined is greater than 0.0. If one of the maxFramerate values does not
// meet this requirement, throw a RangeError.
for (const auto& encoding : aEncodings) {
if (encoding.mMaxFramerate.WasPassed()) {
if (encoding.mMaxFramerate.Value() < 0.0f) {
aRv.ThrowRangeError("maxFramerate must be non-negative");
return;
}
}
}
}
void RTCRtpSender::GetParameters(RTCRtpSendParameters& aParameters) {
MOZ_ASSERT(mParameters.mEncodings.Length());
// If sender.[[LastReturnedParameters]] is not null, return
// sender.[[LastReturnedParameters]], and abort these steps.
if (mLastReturnedParameters.isSome()) {
aParameters = *mLastReturnedParameters;
return;
}
// Let result be a new RTCRtpSendParameters dictionary constructed as follows:
// transactionId is set to a new unique identifier
aParameters.mTransactionId.Construct(mPc->GenerateUUID());
// encodings is set to the value of the [[SendEncodings]] internal slot.
aParameters.mEncodings = mParameters.mEncodings;
// The headerExtensions sequence is populated based on the header extensions
// that have been negotiated for sending
// aParameters.mHeaderExtensions.Construct();
// rtcp.cname is set to the CNAME of the associated RTCPeerConnection.
// rtcp.reducedSize is set to true if reduced-size RTCP has been negotiated
// for sending, and false otherwise.
aParameters.mRtcp.Construct();
aParameters.mRtcp.Value().mCname.Construct();
aParameters.mRtcp.Value().mReducedSize.Construct(false);
aParameters.mHeaderExtensions.Construct();
if (mParameters.mCodecs.WasPassed()) {
aParameters.mCodecs.Construct(mParameters.mCodecs.Value());
}
// Set sender.[[LastReturnedParameters]] to result.
mLastReturnedParameters = Some(aParameters);
// Used to help with warning strings
mLastTransactionId = Some(aParameters.mTransactionId.Value());
// Queue a task that sets sender.[[LastReturnedParameters]] to null.
GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction(
__func__, [this, self = RefPtr<RTCRtpSender>(this)] {
mLastReturnedParameters = Nothing();
}));
}
bool operator==(const RTCRtpEncodingParameters& a1,
const RTCRtpEncodingParameters& a2) {
// webidl does not generate types that are equality comparable
return a1.mActive == a2.mActive && a1.mMaxBitrate == a2.mMaxBitrate &&
a1.mMaxFramerate == a2.mMaxFramerate && a1.mPriority == a2.mPriority &&
a1.mRid == a2.mRid &&
a1.mScaleResolutionDownBy == a2.mScaleResolutionDownBy;
}
// static
void RTCRtpSender::ApplyJsEncodingToConduitEncoding(
const RTCRtpEncodingParameters& aJsEncoding,
VideoCodecConfig::Encoding* aConduitEncoding) {
aConduitEncoding->active = aJsEncoding.mActive;
if (aJsEncoding.mMaxBitrate.WasPassed()) {
aConduitEncoding->constraints.maxBr = aJsEncoding.mMaxBitrate.Value();
}
if (aJsEncoding.mMaxFramerate.WasPassed()) {
aConduitEncoding->constraints.maxFps =
Some(aJsEncoding.mMaxFramerate.Value());
}
if (aJsEncoding.mScaleResolutionDownBy.WasPassed()) {
// Optional does not have a valueOr, despite being based on Maybe
// :(
aConduitEncoding->constraints.scaleDownBy =
aJsEncoding.mScaleResolutionDownBy.Value();
} else {
aConduitEncoding->constraints.scaleDownBy = 1.0f;
}
}
void RTCRtpSender::UpdateRestorableEncodings(
const Sequence<RTCRtpEncodingParameters>& aEncodings) {
MOZ_ASSERT(aEncodings.Length());
if (GetJsepTransceiver().mSendTrack.GetNegotiatedDetails()) {
// Once initial negotiation completes, we are no longer allowed to restore
// the unicast encoding.
mUnicastEncoding.reset();
} else if (mParameters.mEncodings.Length() == 1 &&
!mParameters.mEncodings[0].mRid.WasPassed()) {
// If we have not completed the initial negotiation, and we currently are
// ridless unicast, we need to save our unicast encoding in case a
// rollback occurs.
mUnicastEncoding = Some(mParameters.mEncodings[0]);
}
}
Sequence<RTCRtpEncodingParameters> RTCRtpSender::ToSendEncodings(
const std::vector<std::string>& aRids) const {
MOZ_ASSERT(!aRids.empty());
Sequence<RTCRtpEncodingParameters> result;
// If sendEncodings is given as input to this algorithm, and is non-empty,
// set the [[SendEncodings]] slot to sendEncodings.
for (const auto& rid : aRids) {
MOZ_ASSERT(!rid.empty());
RTCRtpEncodingParameters encoding;
encoding.mActive = true;
encoding.mRid.Construct(NS_ConvertUTF8toUTF16(rid.c_str()));
Unused << result.AppendElement(encoding, fallible);
}
// If sendEncodings is non-empty, set each encoding's scaleResolutionDownBy
// to 2^(length of sendEncodings - encoding index - 1).
if (mTransceiver->IsVideo()) {
double scale = 1.0f;
for (auto it = result.rbegin(); it != result.rend(); ++it) {
it->mScaleResolutionDownBy.Construct(scale);
scale *= 2;
}
}
return result;
}
void RTCRtpSender::MaybeGetJsepRids() {
MOZ_ASSERT(!mSimulcastEnvelopeSet);
MOZ_ASSERT(mParameters.mEncodings.Length());
auto jsepRids = GetJsepTransceiver().mSendTrack.GetRids();
if (!jsepRids.empty()) {
UpdateRestorableEncodings(mParameters.mEncodings);
if (jsepRids.size() != 1 || !jsepRids[0].empty()) {
// JSEP is using at least one rid. Stomp our single ridless encoding
mParameters.mEncodings = ToSendEncodings(jsepRids);
}
mSimulcastEnvelopeSet = true;
mSimulcastEnvelopeSetByJSEP = true;
}
}
Sequence<RTCRtpEncodingParameters> RTCRtpSender::GetMatchingEncodings(
const std::vector<std::string>& aRids) const {
Sequence<RTCRtpEncodingParameters> result;
if (!aRids.empty() && !aRids[0].empty()) {
// Simulcast, or unicast with rid
for (const auto& encoding : mParameters.mEncodings) {
for (const auto& rid : aRids) {
auto utf16Rid = NS_ConvertUTF8toUTF16(rid.c_str());
if (!encoding.mRid.WasPassed() || (utf16Rid == encoding.mRid.Value())) {
auto encodingCopy(encoding);
if (!encodingCopy.mRid.WasPassed()) {
encodingCopy.mRid.Construct(NS_ConvertUTF8toUTF16(rid.c_str()));
}
Unused << result.AppendElement(encodingCopy, fallible);
break;
}
}
}
}
// If we're allowing the old setParameters behavior, we _might_ be able to
// get into this situation even if there were rids above. Be extra careful.
// Under normal circumstances, this just handles the ridless case.
if (!result.Length()) {
// Unicast with no specified rid. Restore mUnicastEncoding, if
// it exists, otherwise pick the first encoding.
if (mUnicastEncoding.isSome()) {
Unused << result.AppendElement(*mUnicastEncoding, fallible);
} else {
Unused << result.AppendElement(mParameters.mEncodings[0], fallible);
}
}
return result;
}
void RTCRtpSender::SetStreams(
const Sequence<OwningNonNull<DOMMediaStream>>& aStreams, ErrorResult& aRv) {
if (mPc->IsClosed()) {
aRv.ThrowInvalidStateError(
"Cannot call setStreams if the peer connection is closed");
return;
}
SetStreamsImpl(aStreams);
mPc->UpdateNegotiationNeeded();
}
void RTCRtpSender::SetStreamsImpl(
const Sequence<OwningNonNull<DOMMediaStream>>& aStreams) {
mStreams.Clear();
std::set<nsString> ids;
for (const auto& stream : aStreams) {
nsString id;
stream->GetId(id);
if (!ids.count(id)) {
ids.insert(id);
mStreams.AppendElement(stream);
}
}
}
void RTCRtpSender::GetStreams(nsTArray<RefPtr<DOMMediaStream>>& aStreams) {
aStreams = mStreams.Clone();
}
class ReplaceTrackOperation final : public PeerConnectionImpl::Operation {
public:
ReplaceTrackOperation(PeerConnectionImpl* aPc,
const RefPtr<RTCRtpTransceiver>& aTransceiver,
const RefPtr<MediaStreamTrack>& aTrack,
ErrorResult& aError);
NS_DECL_ISUPPORTS_INHERITED
NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(ReplaceTrackOperation,
PeerConnectionImpl::Operation)
private:
MOZ_CAN_RUN_SCRIPT
RefPtr<dom::Promise> CallImpl(ErrorResult& aError) override;
~ReplaceTrackOperation() = default;
RefPtr<RTCRtpTransceiver> mTransceiver;
RefPtr<MediaStreamTrack> mNewTrack;
};
NS_IMPL_CYCLE_COLLECTION_INHERITED(ReplaceTrackOperation,
PeerConnectionImpl::Operation, mTransceiver,
mNewTrack)
NS_IMPL_ADDREF_INHERITED(ReplaceTrackOperation, PeerConnectionImpl::Operation)
NS_IMPL_RELEASE_INHERITED(ReplaceTrackOperation, PeerConnectionImpl::Operation)
NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(ReplaceTrackOperation)
NS_INTERFACE_MAP_END_INHERITING(PeerConnectionImpl::Operation)
ReplaceTrackOperation::ReplaceTrackOperation(
PeerConnectionImpl* aPc, const RefPtr<RTCRtpTransceiver>& aTransceiver,
const RefPtr<MediaStreamTrack>& aTrack, ErrorResult& aError)
: PeerConnectionImpl::Operation(aPc, aError),
mTransceiver(aTransceiver),
mNewTrack(aTrack) {}
RefPtr<dom::Promise> ReplaceTrackOperation::CallImpl(ErrorResult& aError) {
RefPtr<RTCRtpSender> sender = mTransceiver->Sender();
// If transceiver.[[Stopping]] is true, return a promise rejected with a
// newly created InvalidStateError.
if (mTransceiver->Stopped() || mTransceiver->Stopping()) {
RefPtr<dom::Promise> error = sender->MakePromise(aError);
if (aError.Failed()) {
return nullptr;
}
MOZ_LOG(gSenderLog, LogLevel::Debug,
("%s Cannot call replaceTrack when transceiver is stopping",
__FUNCTION__));
error->MaybeRejectWithInvalidStateError(
"Cannot call replaceTrack when transceiver is stopping");
return error;
}
// Let p be a new promise.
RefPtr<dom::Promise> p = sender->MakePromise(aError);
if (aError.Failed()) {
return nullptr;
}
if (!sender->SeamlessTrackSwitch(mNewTrack)) {
MOZ_LOG(gSenderLog, LogLevel::Info,
("%s Could not seamlessly replace track", __FUNCTION__));
p->MaybeRejectWithInvalidModificationError(
"Could not seamlessly replace track");
return p;
}
// Queue a task that runs the following steps:
GetMainThreadSerialEventTarget()->Dispatch(NS_NewRunnableFunction(
__func__, [p, sender, track = mNewTrack]() MOZ_CAN_RUN_SCRIPT_BOUNDARY {
// If connection.[[IsClosed]] is true, abort these steps.
// Set sender.[[SenderTrack]] to withTrack.
if (sender->SetSenderTrackWithClosedCheck(track)) {
// Resolve p with undefined.
p->MaybeResolveWithUndefined();
}
}));
// Return p.
return p;
}
already_AddRefed<dom::Promise> RTCRtpSender::ReplaceTrack(
dom::MediaStreamTrack* aWithTrack, ErrorResult& aError) {
// If withTrack is non-null and withTrack.kind differs from the transceiver
// kind of transceiver, return a promise rejected with a newly created
// TypeError.
if (aWithTrack) {
nsString newKind;
aWithTrack->GetKind(newKind);
nsString oldKind;
mTransceiver->GetKind(oldKind);
if (newKind != oldKind) {
RefPtr<dom::Promise> error = MakePromise(aError);
if (aError.Failed()) {
return nullptr;
}
error->MaybeRejectWithTypeError(
"Cannot replaceTrack with a different kind!");
return error.forget();
}
}
MOZ_LOG(gSenderLog, LogLevel::Debug,
("%s[%s]: %s (%p to %p)", mPc->GetHandle().c_str(), GetMid().c_str(),
__FUNCTION__, mSenderTrack.get(), aWithTrack));
// Return the result of chaining the following steps to connection's
// operations chain:
RefPtr<PeerConnectionImpl::Operation> op =
new ReplaceTrackOperation(mPc, mTransceiver, aWithTrack, aError);
if (aError.Failed()) {
return nullptr;
}
// Static analysis forces us to use a temporary.
auto pc = mPc;
return pc->Chain(op, aError);
}
nsPIDOMWindowInner* RTCRtpSender::GetParentObject() const { return mWindow; }
already_AddRefed<dom::Promise> RTCRtpSender::MakePromise(
ErrorResult& aError) const {
return mPc->MakePromise(aError);
}
bool RTCRtpSender::SeamlessTrackSwitch(
const RefPtr<MediaStreamTrack>& aWithTrack) {
// We do not actually update mSenderTrack here! Spec says that happens in a
// queued task after this is done (this happens in
// SetSenderTrackWithClosedCheck).
mPipeline->SetTrack(aWithTrack);
MaybeUpdateConduit();
// There may eventually be cases where a renegotiation is necessary to switch.
return true;
}
void RTCRtpSender::SetTrack(const RefPtr<MediaStreamTrack>& aTrack) {
// Used for RTCPeerConnection.removeTrack and RTCPeerConnection.addTrack
if (mTransceiver->Stopping()) {
return;
}
mSenderTrack = aTrack;
SeamlessTrackSwitch(aTrack);
if (aTrack) {
// RFC says (in the section on remote rollback):
// However, an RtpTransceiver MUST NOT be removed if a track was attached
// to the RtpTransceiver via the addTrack method.
mSenderTrackSetByAddTrack = true;
}
}
bool RTCRtpSender::SetSenderTrackWithClosedCheck(
const RefPtr<MediaStreamTrack>& aTrack) {
if (!mPc->IsClosed()) {
mSenderTrack = aTrack;
return true;
}
return false;
}
void RTCRtpSender::Shutdown() {
MOZ_ASSERT(NS_IsMainThread());
mWatchManager.Shutdown();
mPipeline->Shutdown();
mPipeline = nullptr;
if (mTransform) {
mTransform->GetProxy().SetSender(nullptr);
}
}
void RTCRtpSender::BreakCycles() {
mWindow = nullptr;
mPc = nullptr;
mSenderTrack = nullptr;
mTransceiver = nullptr;
mStreams.Clear();
mDtmf = nullptr;
}
void RTCRtpSender::Unlink() {
if (mTransceiver) {
mTransceiver->Unlink();
}
}
void RTCRtpSender::UpdateTransport() {
MOZ_ASSERT(NS_IsMainThread());
if (!mHaveSetupTransport) {
mPipeline->SetLevel(GetJsepTransceiver().GetLevel());
mHaveSetupTransport = true;
}
mPipeline->UpdateTransport_m(GetJsepTransceiver().mTransport.mTransportId,
nullptr);
}
void RTCRtpSender::MaybeUpdateConduit() {
// NOTE(pkerr) - the Call API requires the both local_ssrc and remote_ssrc be
// set to a non-zero value or the CreateVideo...Stream call will fail.
if (NS_WARN_IF(GetJsepTransceiver().mSendTrack.GetSsrcs().empty())) {
MOZ_ASSERT(
false,
"No local ssrcs! This is a bug in the jsep engine, and should never "
"happen!");
return;
}
if (!mPipeline) {
return;
}
bool wasTransmitting = mTransmitting;
if (mPipeline->mConduit->type() == MediaSessionConduit::VIDEO) {
Maybe<VideoConfig> newConfig = GetNewVideoConfig();
if (newConfig.isSome()) {
ApplyVideoConfig(*newConfig);
}
} else {
Maybe<AudioConfig> newConfig = GetNewAudioConfig();
if (newConfig.isSome()) {
ApplyAudioConfig(*newConfig);
}
}
if (!mSenderTrack && !wasTransmitting && mTransmitting) {
MOZ_LOG(gSenderLog, LogLevel::Debug,
("%s[%s]: %s Starting transmit conduit without send track!",
mPc->GetHandle().c_str(), GetMid().c_str(), __FUNCTION__));
}
}
void RTCRtpSender::UpdateParametersCodecs() {
mParameters.mCodecs.Reset();
mParameters.mCodecs.Construct();
if (GetJsepTransceiver().mSendTrack.GetNegotiatedDetails()) {
const JsepTrackNegotiatedDetails details(
*GetJsepTransceiver().mSendTrack.GetNegotiatedDetails());
if (details.GetEncodingCount()) {
for (const auto& jsepCodec : details.GetEncoding(0).GetCodecs()) {
RTCRtpCodecParameters codec;
RTCRtpTransceiver::ToDomRtpCodecParameters(*jsepCodec, &codec);
Unused << mParameters.mCodecs.Value().AppendElement(codec, fallible);
if (jsepCodec->Type() == SdpMediaSection::kVideo) {
const JsepVideoCodecDescription& videoJsepCodec =
static_cast<JsepVideoCodecDescription&>(*jsepCodec);
// In our JSEP implementation, RTX is an addon to an existing codec,
// not a codec object in its own right. webrtc-pc treats RTX as a
// separate codec, however.
if (videoJsepCodec.mRtxEnabled) {
RTCRtpCodecParameters rtx;
RTCRtpTransceiver::ToDomRtpCodecParametersRtx(videoJsepCodec, &rtx);
Unused << mParameters.mCodecs.Value().AppendElement(rtx, fallible);
}
}
}
}
}
}
void RTCRtpSender::SyncFromJsep(const JsepTransceiver& aJsepTransceiver) {
if (!mSimulcastEnvelopeSet) {
// JSEP is establishing the simulcast envelope for the first time, right now
// This is the addTrack (or addTransceiver without sendEncodings) case.
MaybeGetJsepRids();
} else if (!aJsepTransceiver.mSendTrack.GetNegotiatedDetails() ||
!aJsepTransceiver.mSendTrack.IsInHaveRemote()) {
// Spec says that we do not update our encodings until we're in stable,
// _unless_ this is the first negotiation.
std::vector<std::string> rids = aJsepTransceiver.mSendTrack.GetRids();
if (mSimulcastEnvelopeSetByJSEP && rids.empty()) {
// JSEP previously set the simulcast envelope, but now it has no opinion
// regarding unicast/simulcast. This can only happen on rollback of the
// initial remote offer.
mParameters.mEncodings = GetMatchingEncodings(rids);
MOZ_ASSERT(mParameters.mEncodings.Length());
mSimulcastEnvelopeSetByJSEP = false;
mSimulcastEnvelopeSet = false;
} else if (!rids.empty()) {
// JSEP has an opinion on the simulcast envelope, which trumps anything
// we have already.
mParameters.mEncodings = GetMatchingEncodings(rids);
MOZ_ASSERT(mParameters.mEncodings.Length());
}
}
UpdateParametersCodecs();
MaybeUpdateConduit();
}
void RTCRtpSender::SyncToJsep(JsepTransceiver& aJsepTransceiver) const {
std::vector<std::string> streamIds;
for (const auto& stream : mStreams) {
nsString wideStreamId;
stream->GetId(wideStreamId);
std::string streamId = NS_ConvertUTF16toUTF8(wideStreamId).get();
MOZ_ASSERT(!streamId.empty());
streamIds.push_back(streamId);
}
aJsepTransceiver.mSendTrack.UpdateStreamIds(streamIds);
if (mSimulcastEnvelopeSet) {
std::vector<std::string> rids;
Maybe<RTCRtpSendParameters> parameters;
if (mPendingRidChangeFromCompatMode) {
// *sigh* If we have just let a setParameters change our rids, but we have
// not yet updated mParameters because the queued task hasn't run yet,
// we want to set the _new_ rids on the JsepTrack. So, we are forced to
// grab them from mPendingParameters.
parameters = mPendingParameters;
} else {
parameters = Some(mParameters);
}
for (const auto& encoding : parameters->mEncodings) {
if (encoding.mRid.WasPassed()) {
rids.push_back(NS_ConvertUTF16toUTF8(encoding.mRid.Value()).get());
} else {
rids.push_back("");
}
}
aJsepTransceiver.mSendTrack.SetRids(rids);
}
if (mTransceiver->IsVideo()) {
aJsepTransceiver.mSendTrack.SetMaxEncodings(webrtc::kMaxSimulcastStreams);
} else {
aJsepTransceiver.mSendTrack.SetMaxEncodings(1);
}
if (mSenderTrackSetByAddTrack) {
aJsepTransceiver.SetOnlyExistsBecauseOfSetRemote(false);
}
}
Maybe<RTCRtpSender::VideoConfig> RTCRtpSender::GetNewVideoConfig() {
// It is possible for SDP to signal that there is a send track, but there not
// actually be a send track, according to the specification; all that needs to
// happen is for the transceiver to be configured to send...
if (!GetJsepTransceiver().mSendTrack.GetNegotiatedDetails()) {
return Nothing();
}
VideoConfig oldConfig;
oldConfig.mSsrcs = mSsrcs;
oldConfig.mLocalRtpExtensions = mLocalRtpExtensions;
oldConfig.mCname = mCname;
oldConfig.mTransmitting = mTransmitting;
oldConfig.mVideoRtxSsrcs = mVideoRtxSsrcs;
oldConfig.mVideoCodec = mVideoCodec;
oldConfig.mVideoRtpRtcpConfig = mVideoRtpRtcpConfig;
oldConfig.mVideoCodecMode = mVideoCodecMode;
VideoConfig newConfig(oldConfig);
UpdateBaseConfig(&newConfig);
newConfig.mVideoRtxSsrcs = GetJsepTransceiver().mSendTrack.GetRtxSsrcs();
const JsepTrackNegotiatedDetails details(
*GetJsepTransceiver().mSendTrack.GetNegotiatedDetails());
if (mSenderTrack) {
RefPtr<mozilla::dom::VideoStreamTrack> videotrack =
mSenderTrack->AsVideoStreamTrack();
if (!videotrack) {
MOZ_CRASH(
"In ConfigureVideoCodecMode, mSenderTrack is not video! This should "
"never happen!");
}
dom::MediaSourceEnum source = videotrack->GetSource().GetMediaSource();
switch (source) {
case dom::MediaSourceEnum::Browser:
case dom::MediaSourceEnum::Screen:
case dom::MediaSourceEnum::Window:
case dom::MediaSourceEnum::Application:
newConfig.mVideoCodecMode = webrtc::VideoCodecMode::kScreensharing;
break;
case dom::MediaSourceEnum::Camera:
case dom::MediaSourceEnum::Other:
// Other is used by canvas capture, which we treat as realtime video.
// This seems debatable, but we've been doing it this way for a long
// time, so this is likely fine.
newConfig.mVideoCodecMode = webrtc::VideoCodecMode::kRealtimeVideo;
break;
case dom::MediaSourceEnum::Microphone:
case dom::MediaSourceEnum::AudioCapture:
MOZ_ASSERT(false);
break;
}
}
std::vector<VideoCodecConfig> configs;
RTCRtpTransceiver::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(gSenderLog, LogLevel::Error,
("%s[%s]: %s No video codecs were negotiated (send).",
mPc->GetHandle().c_str(), GetMid().c_str(), __FUNCTION__));
return Nothing();
}
newConfig.mVideoCodec = Some(configs[0]);
// Spec says that we start using new parameters right away, _before_ we
// update the parameters that are visible to JS (ie; mParameters).
const RTCRtpSendParameters& parameters =
mPendingParameters.isSome() ? *mPendingParameters : mParameters;
for (VideoCodecConfig::Encoding& conduitEncoding :
newConfig.mVideoCodec->mEncodings) {
for (const RTCRtpEncodingParameters& jsEncoding : parameters.mEncodings) {
std::string rid;
if (jsEncoding.mRid.WasPassed()) {
rid = NS_ConvertUTF16toUTF8(jsEncoding.mRid.Value()).get();
}
if (conduitEncoding.rid == rid) {
ApplyJsEncodingToConduitEncoding(jsEncoding, &conduitEncoding);
break;
}
}
}
if (!mHaveLoggedUlpfecInfo) {
bool ulpfecNegotiated = false;
for (const auto& codec : configs) {
if (nsCRT::strcasestr(codec.mName.c_str(), "ulpfec")) {
ulpfecNegotiated = true;
}
}
mozilla::glean::codec_stats::ulpfec_negotiated
.Get(ulpfecNegotiated ? "negotiated"_ns : "not_negotiated"_ns)
.Add(1);
mHaveLoggedUlpfecInfo = true;
}
// Log codec information we are tracking
if (!mHaveLoggedOtherFec &&
!GetJsepTransceiver().mSendTrack.GetFecCodecName().empty()) {
mozilla::glean::codec_stats::other_fec_signaled
.Get(nsDependentCString(
GetJsepTransceiver().mSendTrack.GetFecCodecName().c_str()))
.Add(1);
mHaveLoggedOtherFec = true;
}
if (!mHaveLoggedVideoPreferredCodec &&
!GetJsepTransceiver().mSendTrack.GetVideoPreferredCodec().empty()) {
mozilla::glean::codec_stats::video_preferred_codec
.Get(nsDependentCString(
GetJsepTransceiver().mSendTrack.GetVideoPreferredCodec().c_str()))
.Add(1);
mHaveLoggedVideoPreferredCodec = true;
}
newConfig.mVideoRtpRtcpConfig = Some(details.GetRtpRtcpConfig());
if (newConfig == oldConfig) {
MOZ_LOG(gSenderLog, LogLevel::Debug,
("%s[%s]: %s No change in video config", mPc->GetHandle().c_str(),
GetMid().c_str(), __FUNCTION__));
return Nothing();
}
if (newConfig.mVideoCodec.isSome()) {
MOZ_ASSERT(newConfig.mSsrcs.size() ==
newConfig.mVideoCodec->mEncodings.size());
}
return Some(newConfig);
}
Maybe<RTCRtpSender::AudioConfig> RTCRtpSender::GetNewAudioConfig() {
AudioConfig oldConfig;
oldConfig.mSsrcs = mSsrcs;
oldConfig.mLocalRtpExtensions = mLocalRtpExtensions;
oldConfig.mCname = mCname;
oldConfig.mTransmitting = mTransmitting;
oldConfig.mAudioCodec = mAudioCodec;
AudioConfig newConfig(oldConfig);
UpdateBaseConfig(&newConfig);
if (GetJsepTransceiver().mSendTrack.GetNegotiatedDetails() &&
GetJsepTransceiver().mSendTrack.GetActive()) {
const auto& details(
*GetJsepTransceiver().mSendTrack.GetNegotiatedDetails());
std::vector<AudioCodecConfig> configs;
RTCRtpTransceiver::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(gSenderLog, LogLevel::Error,
("%s[%s]: %s No audio codecs were negotiated (send)",
mPc->GetHandle().c_str(), GetMid().c_str(), __FUNCTION__));
return Nothing();
}
std::vector<AudioCodecConfig> dtmfConfigs;
std::copy_if(
configs.begin(), configs.end(), std::back_inserter(dtmfConfigs),
[](const auto& value) { return value.mName == "telephone-event"; });
const AudioCodecConfig& sendCodec = configs[0];
if (!dtmfConfigs.empty()) {
// There is at least one telephone-event codec.
// We primarily choose the codec whose frequency matches the send codec.
// Secondarily we choose the one with the lowest frequency.
auto dtmfIterator =
std::find_if(dtmfConfigs.begin(), dtmfConfigs.end(),
[&sendCodec](const auto& dtmfCodec) {
return dtmfCodec.mFreq == sendCodec.mFreq;
});
if (dtmfIterator == dtmfConfigs.end()) {
dtmfIterator = std::min_element(
dtmfConfigs.begin(), dtmfConfigs.end(),
[](const auto& a, const auto& b) { return a.mFreq < b.mFreq; });
}
MOZ_ASSERT(dtmfIterator != dtmfConfigs.end());
newConfig.mDtmfPt = dtmfIterator->mType;
newConfig.mDtmfFreq = dtmfIterator->mFreq;
}
newConfig.mAudioCodec = Some(sendCodec);
}
if (!mHaveLoggedAudioPreferredCodec &&
!GetJsepTransceiver().mSendTrack.GetAudioPreferredCodec().empty()) {
mozilla::glean::codec_stats::audio_preferred_codec
.Get(nsDependentCString(
GetJsepTransceiver().mSendTrack.GetAudioPreferredCodec().c_str()))
.Add(1);
mHaveLoggedAudioPreferredCodec = true;
}
if (newConfig == oldConfig) {
MOZ_LOG(gSenderLog, LogLevel::Debug,
("%s[%s]: %s No change in audio config", mPc->GetHandle().c_str(),
GetMid().c_str(), __FUNCTION__));
return Nothing();
}
return Some(newConfig);
}
void RTCRtpSender::UpdateBaseConfig(BaseConfig* aConfig) {
aConfig->mSsrcs = GetJsepTransceiver().mSendTrack.GetSsrcs();
aConfig->mCname = GetJsepTransceiver().mSendTrack.GetCNAME();
if (GetJsepTransceiver().mSendTrack.GetNegotiatedDetails() &&
GetJsepTransceiver().mSendTrack.GetActive()) {
const auto& details(
*GetJsepTransceiver().mSendTrack.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);
});
aConfig->mLocalRtpExtensions = extmaps;
}
}
// RTCRtpTransceiver::IsSending is updated after negotiation completes, in a
// queued task (which we may be in right now). Don't use
// JsepTrack::GetActive, because that updates before the queued task, which
// is too early for some of the things we interact with here (eg;
// RTCDTMFSender).
aConfig->mTransmitting = mTransceiver->IsSending();
}
void RTCRtpSender::ApplyVideoConfig(const VideoConfig& aConfig) {
if (aConfig.mVideoCodec.isSome()) {
MOZ_ASSERT(aConfig.mSsrcs.size() == aConfig.mVideoCodec->mEncodings.size());
}
mSsrcs = aConfig.mSsrcs;
mCname = aConfig.mCname;
mLocalRtpExtensions = aConfig.mLocalRtpExtensions;
mVideoRtxSsrcs = aConfig.mVideoRtxSsrcs;
mVideoCodec = aConfig.mVideoCodec;
mVideoRtpRtcpConfig = aConfig.mVideoRtpRtcpConfig;
mVideoCodecMode = aConfig.mVideoCodecMode;
mTransmitting = aConfig.mTransmitting;
}
void RTCRtpSender::ApplyAudioConfig(const AudioConfig& aConfig) {
mSsrcs = aConfig.mSsrcs;
mCname = aConfig.mCname;
mLocalRtpExtensions = aConfig.mLocalRtpExtensions;
mAudioCodec = aConfig.mAudioCodec;
if (aConfig.mDtmfPt >= 0) {
mDtmf->SetPayloadType(aConfig.mDtmfPt, aConfig.mDtmfFreq);
}
mTransmitting = aConfig.mTransmitting;
}
void RTCRtpSender::Stop() {
MOZ_ASSERT(mTransceiver->Stopping());
mTransmitting = false;
}
bool RTCRtpSender::HasTrack(const dom::MediaStreamTrack* aTrack) const {
if (!mSenderTrack) {
return false;
}
if (!aTrack) {
return true;
}
return mSenderTrack.get() == aTrack;
}
RefPtr<MediaPipelineTransmit> RTCRtpSender::GetPipeline() const {
return mPipeline;
}
std::string RTCRtpSender::GetMid() const { return mTransceiver->GetMidAscii(); }
JsepTransceiver& RTCRtpSender::GetJsepTransceiver() {
return mTransceiver->GetJsepTransceiver();
}
void RTCRtpSender::UpdateDtmfSender() {
if (!mDtmf) {
return;
}
if (mTransmitting) {
return;
}
mDtmf->StopPlayout();
}
void RTCRtpSender::SetTransform(RTCRtpScriptTransform* aTransform,
ErrorResult& aError) {
if (aTransform == mTransform.get()) {
// Ok... smile and nod
// TODO: Depending on spec, this might throw
return;
}
if (aTransform && aTransform->IsClaimed()) {
aError.ThrowInvalidStateError("transform has already been used elsewhere");
return;
}
// Seamless switch for frames
if (aTransform) {
mFrameTransformerProxy = &aTransform->GetProxy();
} else {
mFrameTransformerProxy = nullptr;
}
if (mTransform) {
mTransform->GetProxy().SetSender(nullptr);
}
mTransform = const_cast<RTCRtpScriptTransform*>(aTransform);
if (mTransform) {
mTransform->GetProxy().SetSender(this);
mTransform->SetClaimed();
}
}
bool RTCRtpSender::GenerateKeyFrame(const Maybe<std::string>& aRid) {
if (!mTransform || !mPipeline || !mSenderTrack) {
return false;
}
mPipeline->mConduit->AsVideoSessionConduit().apply([&](const auto& conduit) {
conduit->GenerateKeyFrame(aRid, &mTransform->GetProxy());
});
return true;
}
} // namespace mozilla::dom
#undef LOGTAG