Revert "Reland "Propagate media transport to media channel.""
This reverts commit da65ed2adcfa57ff3288ce01c1602c973fcab00d.
Reason for revert: Breaks downstream project
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}
TBR=steveanton@webrtc.org,oprypin@webrtc.org,nisse@webrtc.org,sukhanov@webrtc.org,psla@webrtc.org,sukhanov@google.com
Change-Id: I284bab7230e931cda9ee65cb780a8e7d46fa9072
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/106520
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25223}diff --git a/api/test/fake_media_transport.h b/api/test/fake_media_transport.h
index 563ed90..48970a5 100644
--- a/api/test/fake_media_transport.h
+++ b/api/test/fake_media_transport.h
@@ -62,6 +62,9 @@
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 f1471b6..cba3be3 100644
--- a/media/base/mediachannel.cc
+++ b/media/base/mediachannel.cc
@@ -16,18 +16,15 @@
VideoOptions::~VideoOptions() = default;
MediaChannel::MediaChannel(const MediaConfig& config)
- : enable_dscp_(config.enable_dscp) {}
+ : enable_dscp_(config.enable_dscp), network_interface_(NULL) {}
-MediaChannel::MediaChannel() : enable_dscp_(false) {}
+MediaChannel::MediaChannel() : enable_dscp_(false), network_interface_(NULL) {}
MediaChannel::~MediaChannel() {}
-void MediaChannel::SetInterface(
- NetworkInterface* iface,
- webrtc::MediaTransportInterface* media_transport) {
+void MediaChannel::SetInterface(NetworkInterface* iface) {
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 9948d96e..ff3368c 100644
--- a/media/base/mediachannel.h
+++ b/media/base/mediachannel.h
@@ -22,7 +22,6 @@
#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"
@@ -184,14 +183,8 @@
MediaChannel();
~MediaChannel() override;
- // 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);
+ // Sets the abstract interface class for sending RTP/RTCP data.
+ virtual void SetInterface(NetworkInterface* iface);
// Called when a RTP packet is received.
virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) = 0;
@@ -258,10 +251,6 @@
return network_interface_->SetOption(type, opt, option);
}
- webrtc::MediaTransportInterface* media_transport() {
- return media_transport_;
- }
-
protected:
virtual rtc::DiffServCodePoint PreferredDscp() const;
@@ -294,8 +283,7 @@
// from any MediaEngine threads. This critical section is to protect accessing
// of network_interface_ object.
rtc::CriticalSection network_interface_crit_;
- NetworkInterface* network_interface_ = nullptr;
- webrtc::MediaTransportInterface* media_transport_ = nullptr;
+ NetworkInterface* network_interface_;
};
// The stats information is structured as follows:
diff --git a/media/base/rtpdataengine_unittest.cc b/media/base/rtpdataengine_unittest.cc
index a636ea4..c15c55f 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(), /*media_transport=*/nullptr);
+ channel->SetInterface(iface_.get());
channel->SignalDataReceived.connect(receiver_.get(),
&FakeDataReceiver::OnDataReceived);
return channel;
diff --git a/media/engine/webrtcvideoengine.cc b/media/engine/webrtcvideoengine.cc
index d9ef860..8b17ccd 100644
--- a/media/engine/webrtcvideoengine.cc
+++ b/media/engine/webrtcvideoengine.cc
@@ -1423,13 +1423,8 @@
network_route.packet_overhead);
}
-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);
+void WebRtcVideoChannel::SetInterface(NetworkInterface* iface) {
+ MediaChannel::SetInterface(iface);
// 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 f1c232e..27911b6 100644
--- a/media/engine/webrtcvideoengine.h
+++ b/media/engine/webrtcvideoengine.h
@@ -153,8 +153,7 @@
void OnReadyToSend(bool ready) override;
void OnNetworkRouteChanged(const std::string& transport_name,
const rtc::NetworkRoute& network_route) override;
- void SetInterface(NetworkInterface* iface,
- webrtc::MediaTransportInterface* media_transport) override;
+ void SetInterface(NetworkInterface* iface) override;
// Implemented for VideoMediaChannelTest.
bool sending() const { return sending_; }
diff --git a/media/engine/webrtcvideoengine_unittest.cc b/media/engine/webrtcvideoengine_unittest.cc
index 2675d92..e799b88 100644
--- a/media/engine/webrtcvideoengine_unittest.cc
+++ b/media/engine/webrtcvideoengine_unittest.cc
@@ -1260,7 +1260,7 @@
channel_->OnReadyToSend(true);
EXPECT_TRUE(channel_.get() != NULL);
network_interface_.SetDestination(channel_.get());
- channel_->SetInterface(&network_interface_, /*media_transport=*/nullptr);
+ channel_->SetInterface(&network_interface_);
cricket::VideoRecvParameters parameters;
parameters.codecs = engine_.codecs();
channel_->SetRecvParameters(parameters);
@@ -4597,14 +4597,14 @@
channel.reset(static_cast<cricket::WebRtcVideoChannel*>(
engine_.CreateChannel(call_.get(), config, VideoOptions())));
- channel->SetInterface(network_interface.get(), /*media_transport=*/nullptr);
+ channel->SetInterface(network_interface.get());
// 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())));
- channel->SetInterface(network_interface.get(), /*media_transport=*/nullptr);
+ channel->SetInterface(network_interface.get());
EXPECT_EQ(rtc::DSCP_AF41, network_interface->dscp());
// Packets should also self-identify their dscp in PacketOptions.
@@ -4618,7 +4618,7 @@
config.enable_dscp = false;
channel.reset(static_cast<cricket::WebRtcVideoChannel*>(
engine_.CreateChannel(call_.get(), config, VideoOptions())));
- channel->SetInterface(network_interface.get(), /*media_transport=*/nullptr);
+ channel->SetInterface(network_interface.get());
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
}
diff --git a/media/engine/webrtcvoiceengine_unittest.cc b/media/engine/webrtcvoiceengine_unittest.cc
index 0e586e1..bb816ee 100644
--- a/media/engine/webrtcvoiceengine_unittest.cc
+++ b/media/engine/webrtcvoiceengine_unittest.cc
@@ -3027,14 +3027,14 @@
channel.reset(static_cast<cricket::WebRtcVoiceMediaChannel*>(
engine_->CreateChannel(&call_, config, cricket::AudioOptions())));
- channel->SetInterface(&network_interface, /*media_transport=*/nullptr);
+ channel->SetInterface(&network_interface);
// 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::WebRtcVoiceMediaChannel*>(
engine_->CreateChannel(&call_, config, cricket::AudioOptions())));
- channel->SetInterface(&network_interface, /*media_transport=*/nullptr);
+ channel->SetInterface(&network_interface);
EXPECT_EQ(rtc::DSCP_EF, network_interface.dscp());
// Packets should also self-identify their dscp in PacketOptions.
@@ -3047,11 +3047,11 @@
config.enable_dscp = false;
channel.reset(static_cast<cricket::WebRtcVoiceMediaChannel*>(
engine_->CreateChannel(&call_, config, cricket::AudioOptions())));
- channel->SetInterface(&network_interface, /*media_transport=*/nullptr);
+ channel->SetInterface(&network_interface);
// Default value when DSCP is disabled should be DSCP_DEFAULT.
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp());
- channel->SetInterface(nullptr, nullptr);
+ channel->SetInterface(nullptr);
}
TEST_F(WebRtcVoiceEngineTestFake, SetOutputVolume) {
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index bac1ffd..c9d3fa4 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -516,7 +516,6 @@
":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 91d8cbf..cd1534f 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -155,21 +155,19 @@
rtp_transport_->SignalSentPacket.disconnect(this);
}
-void BaseChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport,
- webrtc::MediaTransportInterface* media_transport) {
+void BaseChannel::Init_w(webrtc::RtpTransportInternal* rtp_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_transport);
+ media_channel_->SetInterface(this);
}
void BaseChannel::Deinit() {
RTC_DCHECK(worker_thread_->IsCurrent());
- media_channel_->SetInterface(/*iface=*/nullptr,
- /*media_transport=*/nullptr);
+ media_channel_->SetInterface(NULL);
// 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.
@@ -1038,7 +1036,7 @@
}
void RtpDataChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport) {
- BaseChannel::Init_w(rtp_transport, /*media_transport=*/nullptr);
+ BaseChannel::Init_w(rtp_transport);
media_channel()->SignalDataReceived.connect(this,
&RtpDataChannel::OnDataReceived);
media_channel()->SignalReadyToSend.connect(
diff --git a/pc/channel.h b/pc/channel.h
index cbd9994..535f64e 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -42,7 +42,6 @@
namespace webrtc {
class AudioSinkInterface;
-class MediaTransportInterface;
} // namespace webrtc
namespace cricket {
@@ -85,8 +84,7 @@
bool srtp_required,
webrtc::CryptoOptions crypto_options);
virtual ~BaseChannel();
- void Init_w(webrtc::RtpTransportInternal* rtp_transport,
- webrtc::MediaTransportInterface* media_transport);
+ void Init_w(webrtc::RtpTransportInternal* rtp_transport);
// Deinit may be called multiple times and is simply ignored if it's already
// done.
@@ -164,11 +162,6 @@
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);
@@ -314,11 +307,6 @@
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 f17032c..94f38c9 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, /*media_transport=*/nullptr);
+ channel->Init_w(rtp_transport);
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, /*media_transport=*/nullptr);
+ channel->Init_w(rtp_transport);
return channel;
}
diff --git a/pc/channelmanager.cc b/pc/channelmanager.cc
index eb27bd7..1c80719 100644
--- a/pc/channelmanager.cc
+++ b/pc/channelmanager.cc
@@ -156,7 +156,6 @@
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,
@@ -165,8 +164,8 @@
if (!worker_thread_->IsCurrent()) {
return worker_thread_->Invoke<VoiceChannel*>(RTC_FROM_HERE, [&] {
return CreateVoiceChannel(call, media_config, rtp_transport,
- media_transport, signaling_thread, content_name,
- srtp_required, crypto_options, options);
+ signaling_thread, content_name, srtp_required,
+ crypto_options, options);
});
}
@@ -188,7 +187,7 @@
absl::WrapUnique(media_channel), content_name, srtp_required,
crypto_options);
- voice_channel->Init_w(rtp_transport, media_transport);
+ voice_channel->Init_w(rtp_transport);
VoiceChannel* voice_channel_ptr = voice_channel.get();
voice_channels_.push_back(std::move(voice_channel));
@@ -254,9 +253,7 @@
worker_thread_, network_thread_, signaling_thread,
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);
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 5cafd8c..6430f8e 100644
--- a/pc/channelmanager.h
+++ b/pc/channelmanager.h
@@ -80,16 +80,14 @@
// 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,
- webrtc::MediaTransportInterface* media_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,
+ 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 6e9cab6..053166b 100644
--- a/pc/channelmanager_unittest.cc
+++ b/pc/channelmanager_unittest.cc
@@ -11,7 +11,6 @@
#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"
@@ -62,21 +61,9 @@
return dtls_srtp_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) {
+ void TestCreateDestroyChannels(webrtc::RtpTransportInternal* rtp_transport) {
cricket::VoiceChannel* voice_channel = cm_->CreateVoiceChannel(
- &fake_call_, cricket::MediaConfig(), rtp_transport, media_transport,
+ &fake_call_, cricket::MediaConfig(), rtp_transport,
rtc::Thread::Current(), cricket::CN_AUDIO, kDefaultSrtpRequired,
webrtc::CryptoOptions(), AudioOptions());
EXPECT_TRUE(voice_channel != nullptr);
@@ -103,7 +90,6 @@
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.
@@ -168,15 +154,7 @@
TEST_F(ChannelManagerTest, CreateDestroyChannels) {
EXPECT_TRUE(cm_->Init());
auto rtp_transport = CreateDtlsSrtpTransport();
- 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());
+ TestCreateDestroyChannels(rtp_transport.get());
}
TEST_F(ChannelManagerTest, CreateDestroyChannelsOnThread) {
@@ -186,7 +164,7 @@
EXPECT_TRUE(cm_->set_network_thread(network_.get()));
EXPECT_TRUE(cm_->Init());
auto rtp_transport = CreateDtlsSrtpTransport();
- TestCreateDestroyChannels(rtp_transport.get(), /*media_transport=*/nullptr);
+ TestCreateDestroyChannels(rtp_transport.get());
}
} // namespace cricket
diff --git a/pc/peerconnection.cc b/pc/peerconnection.cc
index 78a9605..087a4ae 100644
--- a/pc/peerconnection.cc
+++ b/pc/peerconnection.cc
@@ -939,18 +939,6 @@
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));
@@ -5539,11 +5527,11 @@
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
cricket::VoiceChannel* PeerConnection::CreateVoiceChannel(
const std::string& mid) {
- RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
- MediaTransportInterface* media_transport = GetMediaTransport(mid);
-
+ RtpTransportInternal* rtp_transport =
+ transport_controller_->GetRtpTransport(mid);
+ RTC_DCHECK(rtp_transport);
cricket::VoiceChannel* voice_channel = channel_manager()->CreateVoiceChannel(
- call_.get(), configuration_.media_config, rtp_transport, media_transport,
+ call_.get(), configuration_.media_config, rtp_transport,
signaling_thread(), mid, SrtpRequired(),
factory_->options().crypto_options, audio_options_);
if (!voice_channel) {
@@ -5561,9 +5549,9 @@
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
cricket::VideoChannel* PeerConnection::CreateVideoChannel(
const std::string& mid) {
- RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
-
- // TODO(sukhanov): Propagate media_transport to video channel.
+ RtpTransportInternal* rtp_transport =
+ transport_controller_->GetRtpTransport(mid);
+ RTC_DCHECK(rtp_transport);
cricket::VideoChannel* video_channel = channel_manager()->CreateVideoChannel(
call_.get(), configuration_.media_config, rtp_transport,
signaling_thread(), mid, SrtpRequired(),
@@ -5598,7 +5586,9 @@
channel->OnTransportChannelCreated();
}
} else {
- RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
+ RtpTransportInternal* rtp_transport =
+ transport_controller_->GetRtpTransport(mid);
+ RTC_DCHECK(rtp_transport);
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 0cd976a..b42d620 100644
--- a/pc/peerconnection.h
+++ b/pc/peerconnection.h
@@ -915,22 +915,6 @@
// 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 6b50091..6f4fe1e 100644
--- a/pc/peerconnection_media_unittest.cc
+++ b/pc/peerconnection_media_unittest.cc
@@ -15,7 +15,6 @@
#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"
@@ -72,26 +71,13 @@
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();
-
- 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 pc_factory = CreateModularPeerConnectionFactory(
+ rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
+ std::move(media_engine), CreateCallFactory(),
+ CreateRtcEventLogFactory());
auto fake_port_allocator = absl::make_unique<cricket::FakePortAllocator>(
rtc::Thread::Current(), nullptr);
@@ -1086,69 +1072,6 @@
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 70b1601..fad839c 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(),
- /*media_transport=*/nullptr, rtc::Thread::Current(), cricket::CN_AUDIO,
- srtp_required, webrtc::CryptoOptions(), cricket::AudioOptions());
+ 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,
