Reland "Reland "Propagate media transport to media channel.""
This is a reland of da65ed2adcfa57ff3288ce01c1602c973fcab00d
Original change's description:
> Reland "Propagate media transport to media channel."
>
> This reverts commit 37cf2455a420124b341ad06ac27fa3c4dbd29d3c.
>
> Reason for revert: <INSERT REASONING HERE>
>
> Original change's description:
> > Revert "Propagate media transport to media channel."
> >
> > This reverts commit 8c16f745ab92cb6d305283e87fa8a661ae500ce4.
> >
> > Reason for revert: Breaks downstream project
> >
> > Original change's description:
> > > Propagate media transport to media channel.
> > >
> > > 1. Pass media transport factory to JSEP transport controller.
> > > 2. Pass media transport to voice media channel.
> > > 3. Add basic unit test that make sure if peer connection is created with media transport, it is propagated to voice media channel.
> > >
> > > Change-Id: Ie922db78ade0efd893e019cd2b4441a9947a2f71
> > > Bug: webrtc:9719
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/105542
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > > Reviewed-by: Peter Slatala <psla@webrtc.org>
> > > Commit-Queue: Anton Sukhanov <sukhanov@google.com>
> > > Cr-Commit-Position: refs/heads/master@{#25152}
> >
> > TBR=steveanton@webrtc.org,nisse@webrtc.org,psla@webrtc.org,sukhanov@google.com
> >
> > # Not skipping CQ checks because original CL landed > 1 day ago.
> >
> > Bug: webrtc:9719
> > Change-Id: Ic78cdc142a2145682ad74eac8b72c71c50f0a5c1
> > Reviewed-on: https://webrtc-review.googlesource.com/c/105840
> > Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> > Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25154}
>
> TBR=steveanton@webrtc.org,oprypin@webrtc.org,nisse@webrtc.org,sukhanov@webrtc.org,psla@webrtc.org,sukhanov@google.com
>
> Change-Id: I505ff3451eae81573531faef155ff35d7f894022
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9719
> Reviewed-on: https://webrtc-review.googlesource.com/c/106500
> Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
> Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25220}
Bug: webrtc:9719
Tbr: Steve Anton <steveanton@webrtc.org>
Tbr: Niels Moller <nisse@webrtc.org>
Change-Id: Ib45691ba8be9abb89ff8c6dac1861bdf59be4c8d
Reviewed-on: https://webrtc-review.googlesource.com/c/106561
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25240}diff --git a/api/test/fake_media_transport.h b/api/test/fake_media_transport.h
index 48970a5..563ed90 100644
--- a/api/test/fake_media_transport.h
+++ b/api/test/fake_media_transport.h
@@ -62,9 +62,6 @@
rtc::PacketTransportInternal* packet_transport,
rtc::Thread* network_thread,
bool is_caller) override {
- RTC_CHECK(network_thread != nullptr);
- RTC_CHECK(packet_transport != nullptr);
-
std::unique_ptr<MediaTransportInterface> media_transport =
absl::make_unique<FakeMediaTransport>(is_caller);
diff --git a/media/base/mediachannel.cc b/media/base/mediachannel.cc
index cba3be3..f1471b6 100644
--- a/media/base/mediachannel.cc
+++ b/media/base/mediachannel.cc
@@ -16,15 +16,18 @@
VideoOptions::~VideoOptions() = default;
MediaChannel::MediaChannel(const MediaConfig& config)
- : enable_dscp_(config.enable_dscp), network_interface_(NULL) {}
+ : enable_dscp_(config.enable_dscp) {}
-MediaChannel::MediaChannel() : enable_dscp_(false), network_interface_(NULL) {}
+MediaChannel::MediaChannel() : enable_dscp_(false) {}
MediaChannel::~MediaChannel() {}
-void MediaChannel::SetInterface(NetworkInterface* iface) {
+void MediaChannel::SetInterface(
+ NetworkInterface* iface,
+ webrtc::MediaTransportInterface* media_transport) {
rtc::CritScope cs(&network_interface_crit_);
network_interface_ = iface;
+ media_transport_ = media_transport;
SetDscp(enable_dscp_ ? PreferredDscp() : rtc::DSCP_DEFAULT);
}
diff --git a/media/base/mediachannel.h b/media/base/mediachannel.h
index ff3368c..9948d96e 100644
--- a/media/base/mediachannel.h
+++ b/media/base/mediachannel.h
@@ -22,6 +22,7 @@
#include "api/audio_options.h"
#include "api/crypto/framedecryptorinterface.h"
#include "api/crypto/frameencryptorinterface.h"
+#include "api/media_transport_interface.h"
#include "api/rtcerror.h"
#include "api/rtpparameters.h"
#include "api/rtpreceiverinterface.h"
@@ -183,8 +184,14 @@
MediaChannel();
~MediaChannel() override;
- // Sets the abstract interface class for sending RTP/RTCP data.
- virtual void SetInterface(NetworkInterface* iface);
+ // Sets the abstract interface class for sending RTP/RTCP data and
+ // interface for media transport (experimental). If media transport is
+ // provided, it should be used instead of RTP/RTCP.
+ // TODO(sukhanov): Currently media transport can co-exist with RTP/RTCP, but
+ // in the future we will refactor code to send all frames with media
+ // transport.
+ virtual void SetInterface(NetworkInterface* iface,
+ webrtc::MediaTransportInterface* media_transport);
// Called when a RTP packet is received.
virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) = 0;
@@ -251,6 +258,10 @@
return network_interface_->SetOption(type, opt, option);
}
+ webrtc::MediaTransportInterface* media_transport() {
+ return media_transport_;
+ }
+
protected:
virtual rtc::DiffServCodePoint PreferredDscp() const;
@@ -283,7 +294,8 @@
// from any MediaEngine threads. This critical section is to protect accessing
// of network_interface_ object.
rtc::CriticalSection network_interface_crit_;
- NetworkInterface* network_interface_;
+ NetworkInterface* network_interface_ = nullptr;
+ webrtc::MediaTransportInterface* media_transport_ = nullptr;
};
// The stats information is structured as follows:
diff --git a/media/base/rtpdataengine_unittest.cc b/media/base/rtpdataengine_unittest.cc
index c15c55f..a636ea4 100644
--- a/media/base/rtpdataengine_unittest.cc
+++ b/media/base/rtpdataengine_unittest.cc
@@ -73,7 +73,7 @@
cricket::MediaConfig config;
cricket::RtpDataMediaChannel* channel =
static_cast<cricket::RtpDataMediaChannel*>(dme->CreateChannel(config));
- channel->SetInterface(iface_.get());
+ channel->SetInterface(iface_.get(), /*media_transport=*/nullptr);
channel->SignalDataReceived.connect(receiver_.get(),
&FakeDataReceiver::OnDataReceived);
return channel;
diff --git a/media/engine/webrtcvideoengine.cc b/media/engine/webrtcvideoengine.cc
index de1a886..94380dd 100644
--- a/media/engine/webrtcvideoengine.cc
+++ b/media/engine/webrtcvideoengine.cc
@@ -1425,8 +1425,13 @@
network_route.packet_overhead);
}
-void WebRtcVideoChannel::SetInterface(NetworkInterface* iface) {
- MediaChannel::SetInterface(iface);
+void WebRtcVideoChannel::SetInterface(
+ NetworkInterface* iface,
+ webrtc::MediaTransportInterface* media_transport) {
+ // TODO(sukhanov): Video is not currently supported with media transport.
+ RTC_CHECK(media_transport == nullptr);
+
+ MediaChannel::SetInterface(iface, media_transport);
// Set the RTP recv/send buffer to a bigger size.
// The group here can be either a positive integer with an explicit size, in
diff --git a/media/engine/webrtcvideoengine.h b/media/engine/webrtcvideoengine.h
index cf37e45..7de68a9 100644
--- a/media/engine/webrtcvideoengine.h
+++ b/media/engine/webrtcvideoengine.h
@@ -156,7 +156,8 @@
void OnReadyToSend(bool ready) override;
void OnNetworkRouteChanged(const std::string& transport_name,
const rtc::NetworkRoute& network_route) override;
- void SetInterface(NetworkInterface* iface) override;
+ void SetInterface(NetworkInterface* iface,
+ webrtc::MediaTransportInterface* media_transport) override;
// Implemented for VideoMediaChannelTest.
bool sending() const { return sending_; }
diff --git a/media/engine/webrtcvideoengine_unittest.cc b/media/engine/webrtcvideoengine_unittest.cc
index ce2bbd9..1f946cb 100644
--- a/media/engine/webrtcvideoengine_unittest.cc
+++ b/media/engine/webrtcvideoengine_unittest.cc
@@ -1261,7 +1261,7 @@
channel_->OnReadyToSend(true);
EXPECT_TRUE(channel_.get() != NULL);
network_interface_.SetDestination(channel_.get());
- channel_->SetInterface(&network_interface_);
+ channel_->SetInterface(&network_interface_, /*media_transport=*/nullptr);
cricket::VideoRecvParameters parameters;
parameters.codecs = engine_.codecs();
channel_->SetRecvParameters(parameters);
@@ -4599,14 +4599,14 @@
channel.reset(static_cast<cricket::WebRtcVideoChannel*>(engine_.CreateChannel(
call_.get(), config, VideoOptions(), webrtc::CryptoOptions())));
- channel->SetInterface(network_interface.get());
+ channel->SetInterface(network_interface.get(), /*media_transport=*/nullptr);
// Default value when DSCP is disabled should be DSCP_DEFAULT.
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
config.enable_dscp = true;
channel.reset(static_cast<cricket::WebRtcVideoChannel*>(engine_.CreateChannel(
call_.get(), config, VideoOptions(), webrtc::CryptoOptions())));
- channel->SetInterface(network_interface.get());
+ channel->SetInterface(network_interface.get(), /*media_transport=*/nullptr);
EXPECT_EQ(rtc::DSCP_AF41, network_interface->dscp());
// Packets should also self-identify their dscp in PacketOptions.
@@ -4620,7 +4620,7 @@
config.enable_dscp = false;
channel.reset(static_cast<cricket::WebRtcVideoChannel*>(engine_.CreateChannel(
call_.get(), config, VideoOptions(), webrtc::CryptoOptions())));
- channel->SetInterface(network_interface.get());
+ channel->SetInterface(network_interface.get(), /*media_transport=*/nullptr);
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
}
diff --git a/media/engine/webrtcvoiceengine_unittest.cc b/media/engine/webrtcvoiceengine_unittest.cc
index 8d20b72..36c0923 100644
--- a/media/engine/webrtcvoiceengine_unittest.cc
+++ b/media/engine/webrtcvoiceengine_unittest.cc
@@ -3031,7 +3031,7 @@
channel.reset(
static_cast<cricket::WebRtcVoiceMediaChannel*>(engine_->CreateChannel(
&call_, config, cricket::AudioOptions(), webrtc::CryptoOptions())));
- channel->SetInterface(&network_interface);
+ channel->SetInterface(&network_interface, /*media_transport=*/nullptr);
// Default value when DSCP is disabled should be DSCP_DEFAULT.
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp());
@@ -3039,7 +3039,7 @@
channel.reset(
static_cast<cricket::WebRtcVoiceMediaChannel*>(engine_->CreateChannel(
&call_, config, cricket::AudioOptions(), webrtc::CryptoOptions())));
- channel->SetInterface(&network_interface);
+ channel->SetInterface(&network_interface, /*media_transport=*/nullptr);
EXPECT_EQ(rtc::DSCP_EF, network_interface.dscp());
// Packets should also self-identify their dscp in PacketOptions.
@@ -3053,11 +3053,11 @@
channel.reset(
static_cast<cricket::WebRtcVoiceMediaChannel*>(engine_->CreateChannel(
&call_, config, cricket::AudioOptions(), webrtc::CryptoOptions())));
- channel->SetInterface(&network_interface);
+ channel->SetInterface(&network_interface, /*media_transport=*/nullptr);
// Default value when DSCP is disabled should be DSCP_DEFAULT.
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp());
- channel->SetInterface(nullptr);
+ channel->SetInterface(nullptr, nullptr);
}
TEST_F(WebRtcVoiceEngineTestFake, SetOutputVolume) {
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index c9d3fa4..bac1ffd 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -516,6 +516,7 @@
":pc_test_utils",
"..:webrtc_common",
"../api:callfactory_api",
+ "../api:fake_media_transport",
"../api:libjingle_peerconnection_test_api",
"../api:rtc_stats_api",
"../api/audio_codecs:audio_codecs_api",
diff --git a/pc/channel.cc b/pc/channel.cc
index cd1534f..91d8cbf 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -155,19 +155,21 @@
rtp_transport_->SignalSentPacket.disconnect(this);
}
-void BaseChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport) {
+void BaseChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport,
+ webrtc::MediaTransportInterface* media_transport) {
RTC_DCHECK_RUN_ON(worker_thread_);
network_thread_->Invoke<void>(
RTC_FROM_HERE, [this, rtp_transport] { SetRtpTransport(rtp_transport); });
// Both RTP and RTCP channels should be set, we can call SetInterface on
// the media channel and it can set network options.
- media_channel_->SetInterface(this);
+ media_channel_->SetInterface(this, media_transport);
}
void BaseChannel::Deinit() {
RTC_DCHECK(worker_thread_->IsCurrent());
- media_channel_->SetInterface(NULL);
+ media_channel_->SetInterface(/*iface=*/nullptr,
+ /*media_transport=*/nullptr);
// Packets arrive on the network thread, processing packets calls virtual
// functions, so need to stop this process in Deinit that is called in
// derived classes destructor.
@@ -1036,7 +1038,7 @@
}
void RtpDataChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport) {
- BaseChannel::Init_w(rtp_transport);
+ BaseChannel::Init_w(rtp_transport, /*media_transport=*/nullptr);
media_channel()->SignalDataReceived.connect(this,
&RtpDataChannel::OnDataReceived);
media_channel()->SignalReadyToSend.connect(
diff --git a/pc/channel.h b/pc/channel.h
index c7a842e..8e38cd8 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -42,6 +42,7 @@
namespace webrtc {
class AudioSinkInterface;
+class MediaTransportInterface;
} // namespace webrtc
namespace cricket {
@@ -84,7 +85,8 @@
bool srtp_required,
webrtc::CryptoOptions crypto_options);
virtual ~BaseChannel();
- void Init_w(webrtc::RtpTransportInternal* rtp_transport);
+ void Init_w(webrtc::RtpTransportInternal* rtp_transport,
+ webrtc::MediaTransportInterface* media_transport);
// Deinit may be called multiple times and is simply ignored if it's already
// done.
@@ -162,6 +164,11 @@
return nullptr;
}
+ // Returns media transport, can be null if media transport is not available.
+ webrtc::MediaTransportInterface* media_transport() {
+ return media_transport_;
+ }
+
// From RtpTransport - public for testing only
void OnTransportReadyToSend(bool ready);
@@ -307,6 +314,11 @@
webrtc::RtpTransportInternal* rtp_transport_ = nullptr;
+ // Optional media transport (experimental).
+ // If provided, audio and video will be sent through media_transport instead
+ // of RTP/RTCP. Currently media_transport can co-exist with rtp_transport.
+ webrtc::MediaTransportInterface* media_transport_ = nullptr;
+
std::vector<std::pair<rtc::Socket::Option, int> > socket_options_;
std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_;
bool writable_ = false;
diff --git a/pc/channel_unittest.cc b/pc/channel_unittest.cc
index 94f38c9..f17032c 100644
--- a/pc/channel_unittest.cc
+++ b/pc/channel_unittest.cc
@@ -251,7 +251,7 @@
auto channel = absl::make_unique<typename T::Channel>(
worker_thread, network_thread, signaling_thread, engine, std::move(ch),
cricket::CN_AUDIO, (flags & DTLS) != 0, webrtc::CryptoOptions());
- channel->Init_w(rtp_transport);
+ channel->Init_w(rtp_transport, /*media_transport=*/nullptr);
return channel;
}
@@ -1546,7 +1546,7 @@
auto channel = absl::make_unique<cricket::VideoChannel>(
worker_thread, network_thread, signaling_thread, std::move(ch),
cricket::CN_VIDEO, (flags & DTLS) != 0, webrtc::CryptoOptions());
- channel->Init_w(rtp_transport);
+ channel->Init_w(rtp_transport, /*media_transport=*/nullptr);
return channel;
}
diff --git a/pc/channelmanager.cc b/pc/channelmanager.cc
index 3ef9a42..e976cbc 100644
--- a/pc/channelmanager.cc
+++ b/pc/channelmanager.cc
@@ -156,6 +156,7 @@
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,
@@ -164,8 +165,8 @@
if (!worker_thread_->IsCurrent()) {
return worker_thread_->Invoke<VoiceChannel*>(RTC_FROM_HERE, [&] {
return CreateVoiceChannel(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);
});
}
@@ -187,7 +188,7 @@
absl::WrapUnique(media_channel), content_name, srtp_required,
crypto_options);
- voice_channel->Init_w(rtp_transport);
+ voice_channel->Init_w(rtp_transport, media_transport);
VoiceChannel* voice_channel_ptr = voice_channel.get();
voice_channels_.push_back(std::move(voice_channel));
@@ -253,7 +254,9 @@
worker_thread_, network_thread_, signaling_thread,
absl::WrapUnique(media_channel), content_name, srtp_required,
crypto_options);
- video_channel->Init_w(rtp_transport);
+
+ // TODO(sukhanov): Add media_transport support for video channel.
+ video_channel->Init_w(rtp_transport, /*media_transport=*/nullptr);
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 6430f8e..5cafd8c 100644
--- a/pc/channelmanager.h
+++ b/pc/channelmanager.h
@@ -80,14 +80,16 @@
// call the appropriate Destroy*Channel method when done.
// Creates a voice channel, to be associated with the specified session.
- VoiceChannel* CreateVoiceChannel(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 AudioOptions& options);
+ VoiceChannel* CreateVoiceChannel(
+ 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 AudioOptions& options);
// Destroys a voice channel created by CreateVoiceChannel.
void DestroyVoiceChannel(VoiceChannel* voice_channel);
diff --git a/pc/channelmanager_unittest.cc b/pc/channelmanager_unittest.cc
index 053166b..6e9cab6 100644
--- a/pc/channelmanager_unittest.cc
+++ b/pc/channelmanager_unittest.cc
@@ -11,6 +11,7 @@
#include <memory>
#include <utility>
+#include "api/test/fake_media_transport.h"
#include "media/base/fakemediaengine.h"
#include "media/base/testutils.h"
#include "media/engine/fakewebrtccall.h"
@@ -61,9 +62,21 @@
return dtls_srtp_transport;
}
- void TestCreateDestroyChannels(webrtc::RtpTransportInternal* rtp_transport) {
+ std::unique_ptr<webrtc::MediaTransportInterface> CreateMediaTransport(
+ rtc::PacketTransportInternal* packet_transport) {
+ auto media_transport_result =
+ fake_media_transport_factory_.CreateMediaTransport(packet_transport,
+ network_.get(),
+ /*is_caller=*/true);
+ RTC_CHECK(media_transport_result.ok());
+ return media_transport_result.MoveValue();
+ }
+
+ void TestCreateDestroyChannels(
+ webrtc::RtpTransportInternal* rtp_transport,
+ webrtc::MediaTransportInterface* media_transport) {
cricket::VoiceChannel* voice_channel = cm_->CreateVoiceChannel(
- &fake_call_, cricket::MediaConfig(), rtp_transport,
+ &fake_call_, cricket::MediaConfig(), rtp_transport, media_transport,
rtc::Thread::Current(), cricket::CN_AUDIO, kDefaultSrtpRequired,
webrtc::CryptoOptions(), AudioOptions());
EXPECT_TRUE(voice_channel != nullptr);
@@ -90,6 +103,7 @@
cricket::FakeDataEngine* fdme_;
std::unique_ptr<cricket::ChannelManager> cm_;
cricket::FakeCall fake_call_;
+ webrtc::FakeMediaTransportFactory fake_media_transport_factory_;
};
// Test that we startup/shutdown properly.
@@ -154,7 +168,15 @@
TEST_F(ChannelManagerTest, CreateDestroyChannels) {
EXPECT_TRUE(cm_->Init());
auto rtp_transport = CreateDtlsSrtpTransport();
- TestCreateDestroyChannels(rtp_transport.get());
+ TestCreateDestroyChannels(rtp_transport.get(), /*media_transport=*/nullptr);
+}
+
+TEST_F(ChannelManagerTest, CreateDestroyChannelsWithMediaTransport) {
+ EXPECT_TRUE(cm_->Init());
+ auto rtp_transport = CreateDtlsSrtpTransport();
+ auto media_transport =
+ CreateMediaTransport(rtp_transport->rtcp_packet_transport());
+ TestCreateDestroyChannels(rtp_transport.get(), media_transport.get());
}
TEST_F(ChannelManagerTest, CreateDestroyChannelsOnThread) {
@@ -164,7 +186,7 @@
EXPECT_TRUE(cm_->set_network_thread(network_.get()));
EXPECT_TRUE(cm_->Init());
auto rtp_transport = CreateDtlsSrtpTransport();
- TestCreateDestroyChannels(rtp_transport.get());
+ TestCreateDestroyChannels(rtp_transport.get(), /*media_transport=*/nullptr);
}
} // namespace cricket
diff --git a/pc/peerconnection.cc b/pc/peerconnection.cc
index 357e472..ca4d13d 100644
--- a/pc/peerconnection.cc
+++ b/pc/peerconnection.cc
@@ -939,6 +939,18 @@
config.enable_external_auth = true;
#endif
config.active_reset_srtp_params = configuration.active_reset_srtp_params;
+
+ if (configuration.use_media_transport) {
+ if (!factory_->media_transport_factory()) {
+ RTC_DCHECK(false)
+ << "PeerConnecton is initialized with use_media_transport = true, "
+ << "but media transport factory is not set in PeerConnectioFactory";
+ return false;
+ }
+
+ config.media_transport_factory = factory_->media_transport_factory();
+ }
+
transport_controller_.reset(new JsepTransportController(
signaling_thread(), network_thread(), port_allocator_.get(),
async_resolver_factory_.get(), config));
@@ -5527,11 +5539,11 @@
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
cricket::VoiceChannel* PeerConnection::CreateVoiceChannel(
const std::string& mid) {
- RtpTransportInternal* rtp_transport =
- transport_controller_->GetRtpTransport(mid);
- RTC_DCHECK(rtp_transport);
+ RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
+ MediaTransportInterface* media_transport = GetMediaTransport(mid);
+
cricket::VoiceChannel* voice_channel = channel_manager()->CreateVoiceChannel(
- call_.get(), configuration_.media_config, rtp_transport,
+ call_.get(), configuration_.media_config, rtp_transport, media_transport,
signaling_thread(), mid, SrtpRequired(),
factory_->options().crypto_options, audio_options_);
if (!voice_channel) {
@@ -5549,9 +5561,9 @@
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
cricket::VideoChannel* PeerConnection::CreateVideoChannel(
const std::string& mid) {
- RtpTransportInternal* rtp_transport =
- transport_controller_->GetRtpTransport(mid);
- RTC_DCHECK(rtp_transport);
+ RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
+
+ // TODO(sukhanov): Propagate media_transport to video channel.
cricket::VideoChannel* video_channel = channel_manager()->CreateVideoChannel(
call_.get(), configuration_.media_config, rtp_transport,
signaling_thread(), mid, SrtpRequired(),
@@ -5586,9 +5598,7 @@
channel->OnTransportChannelCreated();
}
} else {
- RtpTransportInternal* rtp_transport =
- transport_controller_->GetRtpTransport(mid);
- RTC_DCHECK(rtp_transport);
+ RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
rtp_data_channel_ = channel_manager()->CreateRtpDataChannel(
configuration_.media_config, rtp_transport, signaling_thread(), mid,
SrtpRequired(), factory_->options().crypto_options);
diff --git a/pc/peerconnection.h b/pc/peerconnection.h
index b42d620..0cd976a 100644
--- a/pc/peerconnection.h
+++ b/pc/peerconnection.h
@@ -915,6 +915,22 @@
// Returns the observer. Will crash on CHECK if the observer is removed.
PeerConnectionObserver* Observer() const;
+ // Returns rtp transport, result can not be nullptr.
+ RtpTransportInternal* GetRtpTransport(const std::string& mid) {
+ auto rtp_transport = transport_controller_->GetRtpTransport(mid);
+ RTC_DCHECK(rtp_transport);
+ return rtp_transport;
+ }
+
+ // Returns media transport, if PeerConnection was created with configuration
+ // to use media transport. Otherwise returns nullptr.
+ MediaTransportInterface* GetMediaTransport(const std::string& mid) {
+ auto media_transport = transport_controller_->GetMediaTransport(mid);
+ RTC_DCHECK(configuration_.use_media_transport ==
+ (media_transport != nullptr));
+ return media_transport;
+ }
+
sigslot::signal1<DataChannel*> SignalDataChannelCreated_;
// Storing the factory as a scoped reference pointer ensures that the memory
diff --git a/pc/peerconnection_media_unittest.cc b/pc/peerconnection_media_unittest.cc
index 6f4fe1e..6b50091 100644
--- a/pc/peerconnection_media_unittest.cc
+++ b/pc/peerconnection_media_unittest.cc
@@ -15,6 +15,7 @@
#include <tuple>
#include "api/call/callfactoryinterface.h"
+#include "api/test/fake_media_transport.h"
#include "logging/rtc_event_log/rtc_event_log_factory.h"
#include "media/base/fakemediaengine.h"
#include "p2p/base/fakeportallocator.h"
@@ -71,13 +72,26 @@
return CreatePeerConnection(RTCConfiguration());
}
+ // Creates PeerConnectionFactory and PeerConnection for given configuration.
+ // Note that PeerConnectionFactory is created with MediaTransportFactory,
+ // because some tests pass config.use_media_transport = true.
WrapperPtr CreatePeerConnection(const RTCConfiguration& config) {
auto media_engine = absl::make_unique<FakeMediaEngine>();
auto* media_engine_ptr = media_engine.get();
- auto pc_factory = CreateModularPeerConnectionFactory(
- rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
- std::move(media_engine), CreateCallFactory(),
- CreateRtcEventLogFactory());
+
+ PeerConnectionFactoryDependencies factory_dependencies;
+
+ factory_dependencies.network_thread = rtc::Thread::Current();
+ factory_dependencies.worker_thread = rtc::Thread::Current();
+ factory_dependencies.signaling_thread = rtc::Thread::Current();
+ factory_dependencies.media_engine = std::move(media_engine);
+ factory_dependencies.call_factory = CreateCallFactory();
+ factory_dependencies.event_log_factory = CreateRtcEventLogFactory();
+ factory_dependencies.media_transport_factory =
+ absl::make_unique<FakeMediaTransportFactory>();
+
+ auto pc_factory =
+ CreateModularPeerConnectionFactory(std::move(factory_dependencies));
auto fake_port_allocator = absl::make_unique<cricket::FakePortAllocator>(
rtc::Thread::Current(), nullptr);
@@ -1072,6 +1086,69 @@
audio_options.combined_audio_video_bwe);
}
+TEST_P(PeerConnectionMediaTest, MediaTransportPropagatedToVoiceEngine) {
+ RTCConfiguration config;
+
+ // Setup PeerConnection to use media transport.
+ config.use_media_transport = true;
+
+ auto caller = CreatePeerConnectionWithAudioVideo(config);
+ auto callee = CreatePeerConnectionWithAudioVideo(config);
+
+ ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
+ auto answer = callee->CreateAnswer();
+ ASSERT_TRUE(callee->SetLocalDescription(std::move(answer)));
+
+ auto caller_voice = caller->media_engine()->GetVoiceChannel(0);
+ auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
+ ASSERT_TRUE(caller_voice);
+ ASSERT_TRUE(callee_voice);
+
+ // Make sure media transport is propagated to voice channel.
+ FakeMediaTransport* caller_voice_media_transport =
+ static_cast<FakeMediaTransport*>(caller_voice->media_transport());
+ FakeMediaTransport* callee_voice_media_transport =
+ static_cast<FakeMediaTransport*>(callee_voice->media_transport());
+ ASSERT_NE(nullptr, caller_voice_media_transport);
+ ASSERT_NE(nullptr, callee_voice_media_transport);
+
+ // Make sure media transport is created with correct is_caller.
+ EXPECT_TRUE(caller_voice_media_transport->is_caller());
+ EXPECT_FALSE(callee_voice_media_transport->is_caller());
+
+ // TODO(sukhanov): Propagate media transport to video channel. This test
+ // will fail once media transport is propagated to video channel and it will
+ // serve as a reminder to add a test for video channel propagation.
+ auto caller_video = caller->media_engine()->GetVideoChannel(0);
+ auto callee_video = callee->media_engine()->GetVideoChannel(0);
+ ASSERT_EQ(nullptr, caller_video->media_transport());
+ ASSERT_EQ(nullptr, callee_video->media_transport());
+}
+
+TEST_P(PeerConnectionMediaTest, MediaTransportNotPropagatedToVoiceEngine) {
+ auto caller = CreatePeerConnectionWithAudioVideo();
+ auto callee = CreatePeerConnectionWithAudioVideo();
+
+ ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
+ auto answer = callee->CreateAnswer();
+ ASSERT_TRUE(callee->SetLocalDescription(std::move(answer)));
+
+ auto caller_voice = caller->media_engine()->GetVoiceChannel(0);
+ auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
+ ASSERT_TRUE(caller_voice);
+ ASSERT_TRUE(callee_voice);
+
+ // Since we did not setup PeerConnection to use media transport, media
+ // transport should not be created / propagated to the voice engine.
+ ASSERT_EQ(nullptr, caller_voice->media_transport());
+ ASSERT_EQ(nullptr, callee_voice->media_transport());
+
+ auto caller_video = caller->media_engine()->GetVideoChannel(0);
+ auto callee_video = callee->media_engine()->GetVideoChannel(0);
+ ASSERT_EQ(nullptr, caller_video->media_transport());
+ ASSERT_EQ(nullptr, callee_video->media_transport());
+}
+
INSTANTIATE_TEST_CASE_P(PeerConnectionMediaTest,
PeerConnectionMediaTest,
Values(SdpSemantics::kPlanB,
diff --git a/pc/rtpsenderreceiver_unittest.cc b/pc/rtpsenderreceiver_unittest.cc
index fad839c..70b1601 100644
--- a/pc/rtpsenderreceiver_unittest.cc
+++ b/pc/rtpsenderreceiver_unittest.cc
@@ -80,8 +80,8 @@
voice_channel_ = channel_manager_.CreateVoiceChannel(
&fake_call_, cricket::MediaConfig(), rtp_transport_.get(),
- rtc::Thread::Current(), cricket::CN_AUDIO, srtp_required,
- webrtc::CryptoOptions(), cricket::AudioOptions());
+ /*media_transport=*/nullptr, rtc::Thread::Current(), cricket::CN_AUDIO,
+ 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,
