Enable use of MediaTransportInterface for video streams.
Bug: webrtc:9719
Change-Id: I8c6027b4b15ed641e42fd210b3ea87d121508a69
Reviewed-on: https://webrtc-review.googlesource.com/c/111751
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26219}diff --git a/pc/channelmanager.cc b/pc/channelmanager.cc
index 0c86cf6..d5cbd9c 100644
--- a/pc/channelmanager.cc
+++ b/pc/channelmanager.cc
@@ -226,6 +226,7 @@
webrtc::Call* call,
const cricket::MediaConfig& media_config,
webrtc::RtpTransportInternal* rtp_transport,
+ webrtc::MediaTransportInterface* media_transport,
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
@@ -234,8 +235,8 @@
if (!worker_thread_->IsCurrent()) {
return worker_thread_->Invoke<VideoChannel*>(RTC_FROM_HERE, [&] {
return CreateVideoChannel(call, media_config, rtp_transport,
- signaling_thread, content_name, srtp_required,
- crypto_options, options);
+ media_transport, signaling_thread, content_name,
+ srtp_required, crypto_options, options);
});
}
@@ -257,8 +258,7 @@
absl::WrapUnique(media_channel), content_name, srtp_required,
crypto_options);
- // TODO(sukhanov): Add media_transport support for video channel.
- video_channel->Init_w(rtp_transport, /*media_transport=*/nullptr);
+ video_channel->Init_w(rtp_transport, media_transport);
VideoChannel* video_channel_ptr = video_channel.get();
video_channels_.push_back(std::move(video_channel));
diff --git a/pc/channelmanager.h b/pc/channelmanager.h
index 2b7a30a..8bd30d1 100644
--- a/pc/channelmanager.h
+++ b/pc/channelmanager.h
@@ -107,14 +107,16 @@
// Creates a video channel, synced with the specified voice channel, and
// associated with the specified session.
// Version of the above that takes PacketTransportInternal.
- VideoChannel* CreateVideoChannel(webrtc::Call* call,
- const cricket::MediaConfig& media_config,
- webrtc::RtpTransportInternal* rtp_transport,
- rtc::Thread* signaling_thread,
- const std::string& content_name,
- bool srtp_required,
- const webrtc::CryptoOptions& crypto_options,
- const VideoOptions& options);
+ VideoChannel* CreateVideoChannel(
+ webrtc::Call* call,
+ const cricket::MediaConfig& media_config,
+ webrtc::RtpTransportInternal* rtp_transport,
+ webrtc::MediaTransportInterface* media_transport,
+ rtc::Thread* signaling_thread,
+ const std::string& content_name,
+ bool srtp_required,
+ const webrtc::CryptoOptions& crypto_options,
+ const VideoOptions& options);
// Destroys a video channel created by CreateVideoChannel.
void DestroyVideoChannel(VideoChannel* video_channel);
diff --git a/pc/channelmanager_unittest.cc b/pc/channelmanager_unittest.cc
index 4d16614..ed09b72 100644
--- a/pc/channelmanager_unittest.cc
+++ b/pc/channelmanager_unittest.cc
@@ -86,7 +86,7 @@
webrtc::CryptoOptions(), AudioOptions());
EXPECT_TRUE(voice_channel != nullptr);
cricket::VideoChannel* video_channel = cm_->CreateVideoChannel(
- &fake_call_, cricket::MediaConfig(), rtp_transport,
+ &fake_call_, cricket::MediaConfig(), rtp_transport, media_transport,
rtc::Thread::Current(), cricket::CN_VIDEO, kDefaultSrtpRequired,
webrtc::CryptoOptions(), VideoOptions());
EXPECT_TRUE(video_channel != nullptr);
diff --git a/pc/peerconnection.cc b/pc/peerconnection.cc
index 518a2e6..2cf14a1 100644
--- a/pc/peerconnection.cc
+++ b/pc/peerconnection.cc
@@ -5922,10 +5922,13 @@
cricket::VideoChannel* PeerConnection::CreateVideoChannel(
const std::string& mid) {
RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
+ MediaTransportInterface* media_transport = nullptr;
+ if (configuration_.use_media_transport) {
+ media_transport = GetMediaTransport(mid);
+ }
- // TODO(sukhanov): Propagate media_transport to video channel.
cricket::VideoChannel* video_channel = channel_manager()->CreateVideoChannel(
- call_.get(), configuration_.media_config, rtp_transport,
+ call_.get(), configuration_.media_config, rtp_transport, media_transport,
signaling_thread(), mid, SrtpRequired(), GetCryptoOptions(),
video_options_);
if (!video_channel) {
diff --git a/pc/peerconnection_integrationtest.cc b/pc/peerconnection_integrationtest.cc
index ea1b5ea..0bd3d09 100644
--- a/pc/peerconnection_integrationtest.cc
+++ b/pc/peerconnection_integrationtest.cc
@@ -3598,6 +3598,41 @@
EXPECT_GE(first_stats.sent_audio_frames, second_stats.received_audio_frames);
}
+TEST_P(PeerConnectionIntegrationTest, MediaTransportBidirectionalVideo) {
+ PeerConnectionInterface::RTCConfiguration rtc_config;
+ rtc_config.use_media_transport = true;
+ rtc_config.enable_dtls_srtp = false; // SDES is required for media transport.
+ ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndMediaTransportFactory(
+ rtc_config, rtc_config, loopback_media_transports()->first_factory(),
+ loopback_media_transports()->second_factory()));
+ ConnectFakeSignaling();
+
+ caller()->AddVideoTrack();
+ callee()->AddVideoTrack();
+ // Start offer/answer exchange and wait for it to complete.
+ caller()->CreateAndSetAndSignalOffer();
+ ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+
+ // Ensure that the media transport is ready.
+ loopback_media_transports()->SetState(webrtc::MediaTransportState::kWritable);
+ loopback_media_transports()->FlushAsyncInvokes();
+
+ MediaExpectations media_expectations;
+ media_expectations.ExpectBidirectionalVideo();
+ ASSERT_TRUE(ExpectNewFrames(media_expectations));
+
+ webrtc::MediaTransportPair::Stats first_stats =
+ loopback_media_transports()->FirstStats();
+ webrtc::MediaTransportPair::Stats second_stats =
+ loopback_media_transports()->SecondStats();
+
+ EXPECT_GT(first_stats.received_video_frames, 0);
+ EXPECT_GE(second_stats.sent_video_frames, first_stats.received_video_frames);
+
+ EXPECT_GT(second_stats.received_video_frames, 0);
+ EXPECT_GE(first_stats.sent_video_frames, second_stats.received_video_frames);
+}
+
// Test that the ICE connection and gathering states eventually reach
// "complete".
TEST_P(PeerConnectionIntegrationTest, IceStatesReachCompletion) {
diff --git a/pc/rtpsenderreceiver_unittest.cc b/pc/rtpsenderreceiver_unittest.cc
index 037446f..ec1e949 100644
--- a/pc/rtpsenderreceiver_unittest.cc
+++ b/pc/rtpsenderreceiver_unittest.cc
@@ -108,8 +108,8 @@
srtp_required, webrtc::CryptoOptions(), cricket::AudioOptions());
video_channel_ = channel_manager_.CreateVideoChannel(
&fake_call_, cricket::MediaConfig(), rtp_transport_.get(),
- rtc::Thread::Current(), cricket::CN_VIDEO, srtp_required,
- webrtc::CryptoOptions(), cricket::VideoOptions());
+ /*media_transport=*/nullptr, rtc::Thread::Current(), cricket::CN_VIDEO,
+ srtp_required, webrtc::CryptoOptions(), cricket::VideoOptions());
voice_channel_->Enable(true);
video_channel_->Enable(true);
voice_media_channel_ = media_engine_->GetVoiceChannel(0);