Diff
Modified: branches/safari-611-branch/LayoutTests/ChangeLog (278900 => 278901)
--- branches/safari-611-branch/LayoutTests/ChangeLog 2021-06-15 21:49:24 UTC (rev 278900)
+++ branches/safari-611-branch/LayoutTests/ChangeLog 2021-06-15 21:49:28 UTC (rev 278901)
@@ -1,3 +1,50 @@
+2021-06-15 Alan Coon <alanc...@apple.com>
+
+ Cherry-pick r278410. rdar://problem/79355285
+
+ Cherry-pick webrtc fix to correctly handle audio track state in case of renegotiation
+ https://bugs.webkit.org/show_bug.cgi?id=226577
+
+ Reviewed by Eric Carlson.
+
+ LayoutTests/imported/w3c:
+
+ * web-platform-tests/webrtc/receiver-track-live.https-expected.txt:
+
+ Source/ThirdParty/libwebrtc:
+
+ Cherry-pick https://webrtc.googlesource.com/src/+/c335b0e63bff56ca0fbfa617dee6a644c85df164%5E%21/.
+ We need to do small changes to peer_connection.cc given the upstream fix is based on a newer version
+ which has some code moved from peer_connection.cc to rtp_transmission_manager.cc.
+
+ * Source/webrtc/pc/audio_rtp_receiver.cc:
+ * Source/webrtc/pc/audio_rtp_receiver.h:
+ * Source/webrtc/pc/peer_connection.cc:
+ * Source/webrtc/pc/peer_connection_rtp_unittest.cc:
+ * Source/webrtc/pc/remote_audio_source.cc:
+ * Source/webrtc/pc/remote_audio_source.h:
+ * Source/webrtc/pc/rtp_sender_receiver_unittest.cc:
+
+ LayoutTests:
+
+ Update test now that we have the correct behavior.
+
+ * webrtc/receiver-track-should-stay-live-even-if-receiver-is-inactive.html:
+
+
+ git-svn-id: https://svn.webkit.org/repository/webkit/trunk@278410 268f45cc-cd09-0410-ab3c-d52691b4dbfc
+
+ 2021-06-03 Youenn Fablet <you...@apple.com>
+
+ Cherry-pick webrtc fix to correctly handle audio track state in case of renegotiation
+ https://bugs.webkit.org/show_bug.cgi?id=226577
+
+ Reviewed by Eric Carlson.
+
+ Update test now that we have the correct behavior.
+
+ * webrtc/receiver-track-should-stay-live-even-if-receiver-is-inactive.html:
+
2021-06-10 Russell Epstein <repst...@apple.com>
Cherry-pick r278177. rdar://problem/79144899
Modified: branches/safari-611-branch/LayoutTests/imported/w3c/ChangeLog (278900 => 278901)
--- branches/safari-611-branch/LayoutTests/imported/w3c/ChangeLog 2021-06-15 21:49:24 UTC (rev 278900)
+++ branches/safari-611-branch/LayoutTests/imported/w3c/ChangeLog 2021-06-15 21:49:28 UTC (rev 278901)
@@ -1,3 +1,48 @@
+2021-06-15 Alan Coon <alanc...@apple.com>
+
+ Cherry-pick r278410. rdar://problem/79355285
+
+ Cherry-pick webrtc fix to correctly handle audio track state in case of renegotiation
+ https://bugs.webkit.org/show_bug.cgi?id=226577
+
+ Reviewed by Eric Carlson.
+
+ LayoutTests/imported/w3c:
+
+ * web-platform-tests/webrtc/receiver-track-live.https-expected.txt:
+
+ Source/ThirdParty/libwebrtc:
+
+ Cherry-pick https://webrtc.googlesource.com/src/+/c335b0e63bff56ca0fbfa617dee6a644c85df164%5E%21/.
+ We need to do small changes to peer_connection.cc given the upstream fix is based on a newer version
+ which has some code moved from peer_connection.cc to rtp_transmission_manager.cc.
+
+ * Source/webrtc/pc/audio_rtp_receiver.cc:
+ * Source/webrtc/pc/audio_rtp_receiver.h:
+ * Source/webrtc/pc/peer_connection.cc:
+ * Source/webrtc/pc/peer_connection_rtp_unittest.cc:
+ * Source/webrtc/pc/remote_audio_source.cc:
+ * Source/webrtc/pc/remote_audio_source.h:
+ * Source/webrtc/pc/rtp_sender_receiver_unittest.cc:
+
+ LayoutTests:
+
+ Update test now that we have the correct behavior.
+
+ * webrtc/receiver-track-should-stay-live-even-if-receiver-is-inactive.html:
+
+
+ git-svn-id: https://svn.webkit.org/repository/webkit/trunk@278410 268f45cc-cd09-0410-ab3c-d52691b4dbfc
+
+ 2021-06-03 Youenn Fablet <you...@apple.com>
+
+ Cherry-pick webrtc fix to correctly handle audio track state in case of renegotiation
+ https://bugs.webkit.org/show_bug.cgi?id=226577
+
+ Reviewed by Eric Carlson.
+
+ * web-platform-tests/webrtc/receiver-track-live.https-expected.txt:
+
2021-05-07 Russell Epstein <repst...@apple.com>
Cherry-pick r276627. rdar://problem/77580982
Modified: branches/safari-611-branch/LayoutTests/imported/w3c/web-platform-tests/webrtc/receiver-track-live.https-expected.txt (278900 => 278901)
--- branches/safari-611-branch/LayoutTests/imported/w3c/web-platform-tests/webrtc/receiver-track-live.https-expected.txt 2021-06-15 21:49:24 UTC (rev 278900)
+++ branches/safari-611-branch/LayoutTests/imported/w3c/web-platform-tests/webrtc/receiver-track-live.https-expected.txt 2021-06-15 21:49:28 UTC (rev 278901)
@@ -1,7 +1,7 @@
PASS Setup audio call
-FAIL Inactivate the audio transceiver assert_equals: expected "live" but got "ended"
-FAIL Reactivate the audio transceiver assert_equals: expected "live" but got "ended"
+PASS Inactivate the audio transceiver
+PASS Reactivate the audio transceiver
PASS Clean-up
Modified: branches/safari-611-branch/LayoutTests/webrtc/receiver-track-should-stay-live-even-if-receiver-is-inactive.html (278900 => 278901)
--- branches/safari-611-branch/LayoutTests/webrtc/receiver-track-should-stay-live-even-if-receiver-is-inactive.html 2021-06-15 21:49:24 UTC (rev 278900)
+++ branches/safari-611-branch/LayoutTests/webrtc/receiver-track-should-stay-live-even-if-receiver-is-inactive.html 2021-06-15 21:49:28 UTC (rev 278901)
@@ -46,8 +46,7 @@
await pc2.setLocalDescription(answer);
await pc1.setRemoteDescription(answer);
- // FIXME: The track should be live but is ended due to a bug in our backend.
- assert_equals(remoteTrack.readyState, "ended");
+ assert_equals(remoteTrack.readyState, "live");
}, "Inactivate the audio transceiver");
promise_test(async (test) => {
@@ -59,8 +58,7 @@
await pc2.setLocalDescription(answer);
await pc1.setRemoteDescription(answer);
- // FIXME: The track should be live but is ended due to a bug in our backend.
- assert_equals(remoteTrack.readyState, "ended");
+ assert_equals(remoteTrack.readyState, "live");
}, "Reactivate the audio transceiver");
</script>
</body>
Modified: branches/safari-611-branch/Source/ThirdParty/libwebrtc/ChangeLog (278900 => 278901)
--- branches/safari-611-branch/Source/ThirdParty/libwebrtc/ChangeLog 2021-06-15 21:49:24 UTC (rev 278900)
+++ branches/safari-611-branch/Source/ThirdParty/libwebrtc/ChangeLog 2021-06-15 21:49:28 UTC (rev 278901)
@@ -1,5 +1,60 @@
2021-06-15 Alan Coon <alanc...@apple.com>
+ Cherry-pick r278410. rdar://problem/79355285
+
+ Cherry-pick webrtc fix to correctly handle audio track state in case of renegotiation
+ https://bugs.webkit.org/show_bug.cgi?id=226577
+
+ Reviewed by Eric Carlson.
+
+ LayoutTests/imported/w3c:
+
+ * web-platform-tests/webrtc/receiver-track-live.https-expected.txt:
+
+ Source/ThirdParty/libwebrtc:
+
+ Cherry-pick https://webrtc.googlesource.com/src/+/c335b0e63bff56ca0fbfa617dee6a644c85df164%5E%21/.
+ We need to do small changes to peer_connection.cc given the upstream fix is based on a newer version
+ which has some code moved from peer_connection.cc to rtp_transmission_manager.cc.
+
+ * Source/webrtc/pc/audio_rtp_receiver.cc:
+ * Source/webrtc/pc/audio_rtp_receiver.h:
+ * Source/webrtc/pc/peer_connection.cc:
+ * Source/webrtc/pc/peer_connection_rtp_unittest.cc:
+ * Source/webrtc/pc/remote_audio_source.cc:
+ * Source/webrtc/pc/remote_audio_source.h:
+ * Source/webrtc/pc/rtp_sender_receiver_unittest.cc:
+
+ LayoutTests:
+
+ Update test now that we have the correct behavior.
+
+ * webrtc/receiver-track-should-stay-live-even-if-receiver-is-inactive.html:
+
+
+ git-svn-id: https://svn.webkit.org/repository/webkit/trunk@278410 268f45cc-cd09-0410-ab3c-d52691b4dbfc
+
+ 2021-06-03 Youenn Fablet <you...@apple.com>
+
+ Cherry-pick webrtc fix to correctly handle audio track state in case of renegotiation
+ https://bugs.webkit.org/show_bug.cgi?id=226577
+
+ Reviewed by Eric Carlson.
+
+ Cherry-pick https://webrtc.googlesource.com/src/+/c335b0e63bff56ca0fbfa617dee6a644c85df164%5E%21/.
+ We need to do small changes to peer_connection.cc given the upstream fix is based on a newer version
+ which has some code moved from peer_connection.cc to rtp_transmission_manager.cc.
+
+ * Source/webrtc/pc/audio_rtp_receiver.cc:
+ * Source/webrtc/pc/audio_rtp_receiver.h:
+ * Source/webrtc/pc/peer_connection.cc:
+ * Source/webrtc/pc/peer_connection_rtp_unittest.cc:
+ * Source/webrtc/pc/remote_audio_source.cc:
+ * Source/webrtc/pc/remote_audio_source.h:
+ * Source/webrtc/pc/rtp_sender_receiver_unittest.cc:
+
+2021-06-15 Alan Coon <alanc...@apple.com>
+
Cherry-pick r278272. rdar://problem/79355341
Use tighter bitrate allocation rules for WebRTC H264 software encoder
Modified: branches/safari-611-branch/Source/ThirdParty/libwebrtc/Source/webrtc/pc/audio_rtp_receiver.cc (278900 => 278901)
--- branches/safari-611-branch/Source/ThirdParty/libwebrtc/Source/webrtc/pc/audio_rtp_receiver.cc 2021-06-15 21:49:24 UTC (rev 278900)
+++ branches/safari-611-branch/Source/ThirdParty/libwebrtc/Source/webrtc/pc/audio_rtp_receiver.cc 2021-06-15 21:49:28 UTC (rev 278901)
@@ -30,18 +30,25 @@
AudioRtpReceiver::AudioRtpReceiver(rtc::Thread* worker_thread,
std::string receiver_id,
- std::vector<std::string> stream_ids)
+ std::vector<std::string> stream_ids,
+ bool is_unified_plan)
: AudioRtpReceiver(worker_thread,
receiver_id,
- CreateStreamsFromIds(std::move(stream_ids))) {}
+ CreateStreamsFromIds(std::move(stream_ids)),
+ is_unified_plan) {}
AudioRtpReceiver::AudioRtpReceiver(
rtc::Thread* worker_thread,
const std::string& receiver_id,
- const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams)
+ const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams,
+ bool is_unified_plan)
: worker_thread_(worker_thread),
id_(receiver_id),
- source_(new rtc::RefCountedObject<RemoteAudioSource>(worker_thread)),
+ source_(new rtc::RefCountedObject<RemoteAudioSource>(
+ worker_thread,
+ is_unified_plan
+ ? RemoteAudioSource::OnAudioChannelGoneAction::kSurvive
+ : RemoteAudioSource::OnAudioChannelGoneAction::kEnd)),
track_(AudioTrackProxyWithInternal<AudioTrack>::Create(
rtc::Thread::Current(),
AudioTrack::Create(receiver_id, source_))),
@@ -139,6 +146,7 @@
if (stopped_) {
return;
}
+ source_->SetState(MediaSourceInterface::kEnded);
if (media_channel_) {
// Allow that SetOutputVolume fail. This is the normal case when the
// underlying media channel has already been deleted.
Modified: branches/safari-611-branch/Source/ThirdParty/libwebrtc/Source/webrtc/pc/audio_rtp_receiver.h (278900 => 278901)
--- branches/safari-611-branch/Source/ThirdParty/libwebrtc/Source/webrtc/pc/audio_rtp_receiver.h 2021-06-15 21:49:24 UTC (rev 278900)
+++ branches/safari-611-branch/Source/ThirdParty/libwebrtc/Source/webrtc/pc/audio_rtp_receiver.h 2021-06-15 21:49:28 UTC (rev 278901)
@@ -39,12 +39,14 @@
public:
AudioRtpReceiver(rtc::Thread* worker_thread,
std::string receiver_id,
- std::vector<std::string> stream_ids);
+ std::vector<std::string> stream_ids,
+ bool is_unified_plan);
// TODO(https://crbug.com/webrtc/9480): Remove this when streams() is removed.
AudioRtpReceiver(
rtc::Thread* worker_thread,
const std::string& receiver_id,
- const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams);
+ const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams,
+ bool is_unified_plan);
virtual ~AudioRtpReceiver();
// ObserverInterface implementation
Modified: branches/safari-611-branch/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peer_connection.cc (278900 => 278901)
--- branches/safari-611-branch/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peer_connection.cc 2021-06-15 21:49:24 UTC (rev 278900)
+++ branches/safari-611-branch/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peer_connection.cc 2021-06-15 21:49:28 UTC (rev 278901)
@@ -1210,7 +1210,7 @@
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(), new AudioRtpReceiver(worker_thread(), receiver_id,
- std::vector<std::string>({})));
+ std::vector<std::string>({}), IsUnifiedPlan()));
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
} else {
RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
@@ -2253,7 +2253,7 @@
// TODO(https://crbug.com/webrtc/9480): When we remove remote_streams(), use
// the constructor taking stream IDs instead.
auto* audio_receiver = new AudioRtpReceiver(
- worker_thread(), remote_sender_info.sender_id, streams);
+ worker_thread(), remote_sender_info.sender_id, streams, IsUnifiedPlan());
audio_receiver->SetMediaChannel(voice_media_channel());
if (remote_sender_info.sender_id == kDefaultAudioSenderId) {
audio_receiver->SetupUnsignaledMediaChannel();
Modified: branches/safari-611-branch/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peer_connection_rtp_unittest.cc (278900 => 278901)
--- branches/safari-611-branch/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peer_connection_rtp_unittest.cc 2021-06-15 21:49:24 UTC (rev 278900)
+++ branches/safari-611-branch/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peer_connection_rtp_unittest.cc 2021-06-15 21:49:28 UTC (rev 278901)
@@ -779,7 +779,57 @@
EXPECT_EQ(receivers[0]->streams()[0]->id(), kStreamId1);
EXPECT_EQ(receivers[0]->streams()[1]->id(), kStreamId2);
}
+TEST_F(PeerConnectionRtpTestUnifiedPlan, TracksDoNotEndWhenSsrcChanges) {
+ constexpr uint32_t kFirstMungedSsrc = 1337u;
+ auto caller = CreatePeerConnection();
+ auto callee = CreatePeerConnection();
+
+ // Caller offers to receive audio and video.
+ RtpTransceiverInit init;
+ init.direction = RtpTransceiverDirection::kRecvOnly;
+ caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init);
+ caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init);
+
+ // Callee wants to send audio and video tracks.
+ callee->AddTrack(callee->CreateAudioTrack("audio_track"), {});
+ callee->AddTrack(callee->CreateVideoTrack("video_track"), {});
+
+ // Do inittial offer/answer exchange.
+ ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
+ ASSERT_TRUE(
+ caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
+ ASSERT_EQ(caller->observer()->add_track_events_.size(), 2u);
+ ASSERT_EQ(caller->pc()->GetReceivers().size(), 2u);
+
+ // Do a follow-up offer/answer exchange where the SSRCs are modified.
+ ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
+ auto answer = callee->CreateAnswer();
+ auto& contents = answer->description()->contents();
+ ASSERT_TRUE(!contents.empty());
+ for (size_t i = 0; i < contents.size(); ++i) {
+ auto& mutable_streams = contents[i].media_description()->mutable_streams();
+ ASSERT_EQ(mutable_streams.size(), 1u);
+ mutable_streams[0].ssrcs = {kFirstMungedSsrc + static_cast<uint32_t>(i)};
+ }
+ ASSERT_TRUE(
+ callee->SetLocalDescription(CloneSessionDescription(answer.get())));
+ ASSERT_TRUE(
+ caller->SetRemoteDescription(CloneSessionDescription(answer.get())));
+
+ // No furher track events should fire because we never changed direction, only
+ // SSRCs.
+ ASSERT_EQ(caller->observer()->add_track_events_.size(), 2u);
+ // We should have the same number of receivers as before.
+ auto receivers = caller->pc()->GetReceivers();
+ ASSERT_EQ(receivers.size(), 2u);
+ // The tracks are still alive.
+ EXPECT_EQ(receivers[0]->track()->state(),
+ MediaStreamTrackInterface::TrackState::kLive);
+ EXPECT_EQ(receivers[1]->track()->state(),
+ MediaStreamTrackInterface::TrackState::kLive);
+}
+
// Tests that with Unified Plan if the the stream id changes for a track when
// when setting a new remote description, that the media stream is updated
// appropriately for the receiver.
Modified: branches/safari-611-branch/Source/ThirdParty/libwebrtc/Source/webrtc/pc/remote_audio_source.cc (278900 => 278901)
--- branches/safari-611-branch/Source/ThirdParty/libwebrtc/Source/webrtc/pc/remote_audio_source.cc 2021-06-15 21:49:24 UTC (rev 278900)
+++ branches/safari-611-branch/Source/ThirdParty/libwebrtc/Source/webrtc/pc/remote_audio_source.cc 2021-06-15 21:49:28 UTC (rev 278901)
@@ -51,9 +51,12 @@
const rtc::scoped_refptr<RemoteAudioSource> source_;
};
-RemoteAudioSource::RemoteAudioSource(rtc::Thread* worker_thread)
+RemoteAudioSource::RemoteAudioSource(
+ rtc::Thread* worker_thread,
+ OnAudioChannelGoneAction on_audio_channel_gone_action)
: main_thread_(rtc::Thread::Current()),
worker_thread_(worker_thread),
+ on_audio_channel_gone_action_(on_audio_channel_gone_action),
state_(MediaSourceInterface::kLive) {
RTC_DCHECK(main_thread_);
RTC_DCHECK(worker_thread_);
@@ -92,6 +95,13 @@
});
}
+void RemoteAudioSource::SetState(SourceState new_state) {
+ if (state_ != new_state) {
+ state_ = new_state;
+ FireOnChanged();
+ }
+}
+
MediaSourceInterface::SourceState RemoteAudioSource::state() const {
RTC_DCHECK(main_thread_->IsCurrent());
return state_;
@@ -158,6 +168,9 @@
}
void RemoteAudioSource::OnAudioChannelGone() {
+ if (on_audio_channel_gone_action_ != OnAudioChannelGoneAction::kEnd) {
+ return;
+ }
// Called when the audio channel is deleted. It may be the worker thread
// in libjingle or may be a different worker thread.
// This object needs to live long enough for the cleanup logic in OnMessage to
@@ -172,8 +185,7 @@
void RemoteAudioSource::OnMessage(rtc::Message* msg) {
RTC_DCHECK(main_thread_->IsCurrent());
sinks_.clear();
- state_ = MediaSourceInterface::kEnded;
- FireOnChanged();
+ SetState(MediaSourceInterface::kEnded);
// Will possibly delete this RemoteAudioSource since it is reference counted
// in the message.
delete msg->pdata;
Modified: branches/safari-611-branch/Source/ThirdParty/libwebrtc/Source/webrtc/pc/remote_audio_source.h (278900 => 278901)
--- branches/safari-611-branch/Source/ThirdParty/libwebrtc/Source/webrtc/pc/remote_audio_source.h 2021-06-15 21:49:24 UTC (rev 278900)
+++ branches/safari-611-branch/Source/ThirdParty/libwebrtc/Source/webrtc/pc/remote_audio_source.h 2021-06-15 21:49:28 UTC (rev 278901)
@@ -34,8 +34,22 @@
class RemoteAudioSource : public Notifier<AudioSourceInterface>,
rtc::MessageHandler {
public:
- explicit RemoteAudioSource(rtc::Thread* worker_thread);
+ // In Unified Plan, receivers map to m= sections and their tracks and sources
+ // survive SSRCs being reconfigured. The life cycle of the remote audio source
+ // is associated with the life cycle of the m= section, and thus even if an
+ // audio channel is destroyed the RemoteAudioSource should kSurvive.
+ //
+ // In Plan B however, remote audio sources map 1:1 with an SSRCs and if an
+ // audio channel is destroyed, the RemoteAudioSource should kEnd.
+ enum class OnAudioChannelGoneAction {
+ kSurvive,
+ kEnd,
+ };
+ explicit RemoteAudioSource(
+ rtc::Thread* worker_thread,
+ OnAudioChannelGoneAction on_audio_channel_gone_action);
+
// Register and unregister remote audio source with the underlying media
// engine.
void Start(cricket::VoiceMediaChannel* media_channel,
@@ -42,6 +56,7 @@
absl::optional<uint32_t> ssrc);
void Stop(cricket::VoiceMediaChannel* media_channel,
absl::optional<uint32_t> ssrc);
+ void SetState(SourceState new_state);
// MediaSourceInterface implementation.
MediaSourceInterface::SourceState state() const override;
@@ -68,6 +83,7 @@
rtc::Thread* const main_thread_;
rtc::Thread* const worker_thread_;
+ const OnAudioChannelGoneAction on_audio_channel_gone_action_;
std::list<AudioObserver*> audio_observers_;
Mutex sink_lock_;
std::list<AudioTrackSinkInterface*> sinks_;
Modified: branches/safari-611-branch/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtp_sender_receiver_unittest.cc (278900 => 278901)
--- branches/safari-611-branch/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtp_sender_receiver_unittest.cc 2021-06-15 21:49:24 UTC (rev 278900)
+++ branches/safari-611-branch/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtp_sender_receiver_unittest.cc 2021-06-15 21:49:28 UTC (rev 278901)
@@ -289,7 +289,8 @@
void CreateAudioRtpReceiver(
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams = {}) {
audio_rtp_receiver_ =
- new AudioRtpReceiver(rtc::Thread::Current(), kAudioTrackId, streams);
+ new AudioRtpReceiver(rtc::Thread::Current(), kAudioTrackId, streams,
+ /*is_unified_plan=*/true);
audio_rtp_receiver_->SetMediaChannel(voice_media_channel_);
audio_rtp_receiver_->SetupMediaChannel(kAudioSsrc);
audio_track_ = audio_rtp_receiver_->audio_track();
