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webrtc_m130/webrtc/ortc/ortcfactory_integrationtest.cc
deadbeef e814a0dee0 Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. 
This CL adds the following interfaces:
* RtpTransportController
* RtpTransport
* RtpSender
* RtpReceiver

They're implemented on top of the "BaseChannel" object, which is normally used
in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result
of this, there are several limitations:

* You can only have one of each type of sender and receiver (audio/video) on top
  of the same transport controller.
* The sender/receiver with the same media type must use the same RTP transport.
* You can't change the transport after creating the sender or receiver.
* Some of the parameters aren't supported.

Later, these "adapter" objects will be gradually replaced by real objects that don't
have these limitations, as "BaseChannel", "MediaChannel" and related code is
restructured. In this CL, we essentially have:

ORTC adapter objects -> BaseChannel -> Media engine
PeerConnection -> BaseChannel -> Media engine

And later we hope to have simply:

PeerConnection -> "Real" ORTC objects -> Media engine

See the linked bug for more context.

BUG=webrtc:7013
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2675173003
Cr-Commit-Position: refs/heads/master@{#16842}
2017-02-26 02:15:09 +00:00

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/*
* Copyright 2017 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <memory>
#include <utility> // For std::pair, std::move.
#include "webrtc/api/ortc/ortcfactoryinterface.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/fakenetwork.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/physicalsocketserver.h"
#include "webrtc/base/virtualsocketserver.h"
#include "webrtc/ortc/testrtpparameters.h"
#include "webrtc/p2p/base/udptransport.h"
#include "webrtc/pc/test/fakeaudiocapturemodule.h"
#include "webrtc/pc/test/fakeperiodicvideocapturer.h"
#include "webrtc/pc/test/fakevideotrackrenderer.h"
namespace {
const int kDefaultTimeout = 10000; // 10 seconds.
// Default number of audio/video frames to wait for before considering a test a
// success.
const int kDefaultNumFrames = 3;
const rtc::IPAddress kIPv4LocalHostAddress =
rtc::IPAddress(0x7F000001); // 127.0.0.1
} // namespace
namespace webrtc {
// Used to test that things work end-to-end when using the default
// implementations of threads/etc. provided by OrtcFactory, with the exception
// of using a virtual network.
//
// By default, the virtual network manager doesn't enumerate any networks, but
// sockets can still be created in this state.
class OrtcFactoryIntegrationTest : public testing::Test {
public:
OrtcFactoryIntegrationTest()
: virtual_socket_server_(&physical_socket_server_),
network_thread_(&virtual_socket_server_),
fake_audio_capture_module1_(FakeAudioCaptureModule::Create()),
fake_audio_capture_module2_(FakeAudioCaptureModule::Create()) {
// Sockets are bound to the ANY address, so this is needed to tell the
// virtual network which address to use in this case.
virtual_socket_server_.SetDefaultRoute(kIPv4LocalHostAddress);
network_thread_.Start();
// Need to create after network thread is started.
ortc_factory1_ = OrtcFactoryInterface::Create(
&network_thread_, nullptr, &fake_network_manager_,
nullptr, fake_audio_capture_module1_)
.MoveValue();
ortc_factory2_ = OrtcFactoryInterface::Create(
&network_thread_, nullptr, &fake_network_manager_,
nullptr, fake_audio_capture_module2_)
.MoveValue();
}
protected:
typedef std::pair<std::unique_ptr<UdpTransportInterface>,
std::unique_ptr<UdpTransportInterface>>
UdpTransportPair;
typedef std::pair<std::unique_ptr<RtpTransportInterface>,
std::unique_ptr<RtpTransportInterface>>
RtpTransportPair;
typedef std::pair<std::unique_ptr<RtpTransportControllerInterface>,
std::unique_ptr<RtpTransportControllerInterface>>
RtpTransportControllerPair;
// Helper function that creates one UDP transport each for |ortc_factory1_|
// and |ortc_factory2_|, and connects them.
UdpTransportPair CreateAndConnectUdpTransportPair() {
auto transport1 = ortc_factory1_->CreateUdpTransport(AF_INET).MoveValue();
auto transport2 = ortc_factory2_->CreateUdpTransport(AF_INET).MoveValue();
transport1->SetRemoteAddress(
rtc::SocketAddress(virtual_socket_server_.GetDefaultRoute(AF_INET),
transport2->GetLocalAddress().port()));
transport2->SetRemoteAddress(
rtc::SocketAddress(virtual_socket_server_.GetDefaultRoute(AF_INET),
transport1->GetLocalAddress().port()));
return {std::move(transport1), std::move(transport2)};
}
// Creates one transport controller each for |ortc_factory1_| and
// |ortc_factory2_|.
RtpTransportControllerPair CreateRtpTransportControllerPair() {
return {ortc_factory1_->CreateRtpTransportController().MoveValue(),
ortc_factory2_->CreateRtpTransportController().MoveValue()};
}
// Helper function that creates a pair of RtpTransports between
// |ortc_factory1_| and |ortc_factory2_|. Expected to be called with the
// result of CreateAndConnectUdpTransportPair. |rtcp_udp_transports| can be
// empty if RTCP muxing is used. |transport_controllers| can be empty if
// these transports are being created using a default transport controller.
RtpTransportPair CreateRtpTransportPair(
const RtcpParameters& rtcp_parameters,
const UdpTransportPair& rtp_udp_transports,
const UdpTransportPair& rtcp_udp_transports,
const RtpTransportControllerPair& transport_controllers) {
auto transport_result1 = ortc_factory1_->CreateRtpTransport(
rtcp_parameters, rtp_udp_transports.first.get(),
rtcp_udp_transports.first.get(), transport_controllers.first.get());
auto transport_result2 = ortc_factory2_->CreateRtpTransport(
rtcp_parameters, rtp_udp_transports.second.get(),
rtcp_udp_transports.second.get(), transport_controllers.second.get());
return {transport_result1.MoveValue(), transport_result2.MoveValue()};
}
// For convenience when |rtcp_udp_transports| and |transport_controllers|
// aren't needed.
RtpTransportPair CreateRtpTransportPair(
const RtcpParameters& rtcp_parameters,
const UdpTransportPair& rtp_udp_transports) {
return CreateRtpTransportPair(rtcp_parameters, rtp_udp_transports,
UdpTransportPair(),
RtpTransportControllerPair());
}
// Ends up using fake audio capture module, which was passed into OrtcFactory
// on creation.
rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack(
const std::string& id,
OrtcFactoryInterface* ortc_factory) {
// Disable echo cancellation to make test more efficient.
cricket::AudioOptions options;
options.echo_cancellation.emplace(true);
rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
ortc_factory->CreateAudioSource(options);
return ortc_factory->CreateAudioTrack(id, source);
}
// Stores created capturer in |fake_video_capturers_|.
rtc::scoped_refptr<webrtc::VideoTrackInterface>
CreateLocalVideoTrackAndFakeCapturer(const std::string& id,
OrtcFactoryInterface* ortc_factory) {
cricket::FakeVideoCapturer* fake_capturer =
new webrtc::FakePeriodicVideoCapturer();
fake_video_capturers_.push_back(fake_capturer);
rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source =
ortc_factory->CreateVideoSource(
std::unique_ptr<cricket::VideoCapturer>(fake_capturer));
return rtc::scoped_refptr<webrtc::VideoTrackInterface>(
ortc_factory->CreateVideoTrack(id, source));
}
rtc::PhysicalSocketServer physical_socket_server_;
rtc::VirtualSocketServer virtual_socket_server_;
rtc::Thread network_thread_;
rtc::FakeNetworkManager fake_network_manager_;
rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module1_;
rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module2_;
std::unique_ptr<OrtcFactoryInterface> ortc_factory1_;
std::unique_ptr<OrtcFactoryInterface> ortc_factory2_;
// Actually owned by video tracks.
std::vector<cricket::FakeVideoCapturer*> fake_video_capturers_;
};
// Very basic end-to-end test with a single pair of audio RTP sender and
// receiver.
//
// Uses muxed RTCP, and minimal parameters with a hard-coded config that's
// known to work.
TEST_F(OrtcFactoryIntegrationTest, BasicOneWayAudioRtpSenderAndReceiver) {
auto udp_transports = CreateAndConnectUdpTransportPair();
auto rtp_transports =
CreateRtpTransportPair(MakeRtcpMuxParameters(), udp_transports);
auto sender_result = ortc_factory1_->CreateRtpSender(
cricket::MEDIA_TYPE_AUDIO, rtp_transports.first.get());
auto receiver_result = ortc_factory2_->CreateRtpReceiver(
cricket::MEDIA_TYPE_AUDIO, rtp_transports.second.get());
ASSERT_TRUE(sender_result.ok());
ASSERT_TRUE(receiver_result.ok());
auto sender = sender_result.MoveValue();
auto receiver = receiver_result.MoveValue();
RTCError error =
sender->SetTrack(CreateLocalAudioTrack("audio", ortc_factory1_.get()));
EXPECT_TRUE(error.ok());
RtpParameters opus_parameters = MakeMinimalOpusParameters();
EXPECT_TRUE(receiver->Receive(opus_parameters).ok());
EXPECT_TRUE(sender->Send(opus_parameters).ok());
// Sender and receiver are connected and configured; audio frames should be
// able to flow at this point.
EXPECT_TRUE_WAIT(
fake_audio_capture_module2_->frames_received() > kDefaultNumFrames,
kDefaultTimeout);
}
// Very basic end-to-end test with a single pair of video RTP sender and
// receiver.
//
// Uses muxed RTCP, and minimal parameters with a hard-coded config that's
// known to work.
TEST_F(OrtcFactoryIntegrationTest, BasicOneWayVideoRtpSenderAndReceiver) {
auto udp_transports = CreateAndConnectUdpTransportPair();
auto rtp_transports =
CreateRtpTransportPair(MakeRtcpMuxParameters(), udp_transports);
auto sender_result = ortc_factory1_->CreateRtpSender(
cricket::MEDIA_TYPE_VIDEO, rtp_transports.first.get());
auto receiver_result = ortc_factory2_->CreateRtpReceiver(
cricket::MEDIA_TYPE_VIDEO, rtp_transports.second.get());
ASSERT_TRUE(sender_result.ok());
ASSERT_TRUE(receiver_result.ok());
auto sender = sender_result.MoveValue();
auto receiver = receiver_result.MoveValue();
RTCError error = sender->SetTrack(
CreateLocalVideoTrackAndFakeCapturer("video", ortc_factory1_.get()));
EXPECT_TRUE(error.ok());
RtpParameters vp8_parameters = MakeMinimalVp8Parameters();
EXPECT_TRUE(receiver->Receive(vp8_parameters).ok());
EXPECT_TRUE(sender->Send(vp8_parameters).ok());
FakeVideoTrackRenderer fake_renderer(
static_cast<VideoTrackInterface*>(receiver->GetTrack().get()));
// Sender and receiver are connected and configured; video frames should be
// able to flow at this point.
EXPECT_TRUE_WAIT(fake_renderer.num_rendered_frames() > kDefaultNumFrames,
kDefaultTimeout);
}
// Test that if the track is changed while sending, the sender seamlessly
// transitions to sending it and frames are received end-to-end.
//
// Only doing this for video, since given that audio is sourced from a single
// fake audio capture module, the audio track is just a dummy object.
// TODO(deadbeef): Change this when possible.
TEST_F(OrtcFactoryIntegrationTest, SetTrackWhileSending) {
auto udp_transports = CreateAndConnectUdpTransportPair();
auto rtp_transports =
CreateRtpTransportPair(MakeRtcpMuxParameters(), udp_transports);
auto sender_result = ortc_factory1_->CreateRtpSender(
cricket::MEDIA_TYPE_VIDEO, rtp_transports.first.get());
auto receiver_result = ortc_factory2_->CreateRtpReceiver(
cricket::MEDIA_TYPE_VIDEO, rtp_transports.second.get());
ASSERT_TRUE(sender_result.ok());
ASSERT_TRUE(receiver_result.ok());
auto sender = sender_result.MoveValue();
auto receiver = receiver_result.MoveValue();
RTCError error = sender->SetTrack(
CreateLocalVideoTrackAndFakeCapturer("video_1", ortc_factory1_.get()));
EXPECT_TRUE(error.ok());
RtpParameters vp8_parameters = MakeMinimalVp8Parameters();
EXPECT_TRUE(receiver->Receive(vp8_parameters).ok());
EXPECT_TRUE(sender->Send(vp8_parameters).ok());
FakeVideoTrackRenderer fake_renderer(
static_cast<VideoTrackInterface*>(receiver->GetTrack().get()));
// Expect for some initial number of frames to be received.
EXPECT_TRUE_WAIT(fake_renderer.num_rendered_frames() > kDefaultNumFrames,
kDefaultTimeout);
// Stop the old capturer, set a new track, and verify new frames are received
// from the new track. Stopping the old capturer ensures that we aren't
// actually still getting frames from it.
fake_video_capturers_[0]->Stop();
int prev_num_frames = fake_renderer.num_rendered_frames();
error = sender->SetTrack(
CreateLocalVideoTrackAndFakeCapturer("video_2", ortc_factory1_.get()));
EXPECT_TRUE(error.ok());
EXPECT_TRUE_WAIT(
fake_renderer.num_rendered_frames() > kDefaultNumFrames + prev_num_frames,
kDefaultTimeout);
}
// End-to-end test with two pairs of RTP senders and receivers, for audio and
// video.
//
// Uses muxed RTCP, and minimal parameters with hard-coded configs that are
// known to work.
TEST_F(OrtcFactoryIntegrationTest,
BasicTwoWayAudioVideoRtpSendersAndReceivers) {
auto udp_transports = CreateAndConnectUdpTransportPair();
auto rtp_transports =
CreateRtpTransportPair(MakeRtcpMuxParameters(), udp_transports);
// Create all the senders and receivers (four per endpoint).
auto audio_sender_result1 = ortc_factory1_->CreateRtpSender(
cricket::MEDIA_TYPE_AUDIO, rtp_transports.first.get());
auto video_sender_result1 = ortc_factory1_->CreateRtpSender(
cricket::MEDIA_TYPE_VIDEO, rtp_transports.first.get());
auto audio_receiver_result1 = ortc_factory1_->CreateRtpReceiver(
cricket::MEDIA_TYPE_AUDIO, rtp_transports.first.get());
auto video_receiver_result1 = ortc_factory1_->CreateRtpReceiver(
cricket::MEDIA_TYPE_VIDEO, rtp_transports.first.get());
ASSERT_TRUE(audio_sender_result1.ok());
ASSERT_TRUE(video_sender_result1.ok());
ASSERT_TRUE(audio_receiver_result1.ok());
ASSERT_TRUE(video_receiver_result1.ok());
auto audio_sender1 = audio_sender_result1.MoveValue();
auto video_sender1 = video_sender_result1.MoveValue();
auto audio_receiver1 = audio_receiver_result1.MoveValue();
auto video_receiver1 = video_receiver_result1.MoveValue();
auto audio_sender_result2 = ortc_factory2_->CreateRtpSender(
cricket::MEDIA_TYPE_AUDIO, rtp_transports.second.get());
auto video_sender_result2 = ortc_factory2_->CreateRtpSender(
cricket::MEDIA_TYPE_VIDEO, rtp_transports.second.get());
auto audio_receiver_result2 = ortc_factory2_->CreateRtpReceiver(
cricket::MEDIA_TYPE_AUDIO, rtp_transports.second.get());
auto video_receiver_result2 = ortc_factory2_->CreateRtpReceiver(
cricket::MEDIA_TYPE_VIDEO, rtp_transports.second.get());
ASSERT_TRUE(audio_sender_result2.ok());
ASSERT_TRUE(video_sender_result2.ok());
ASSERT_TRUE(audio_receiver_result2.ok());
ASSERT_TRUE(video_receiver_result2.ok());
auto audio_sender2 = audio_sender_result2.MoveValue();
auto video_sender2 = video_sender_result2.MoveValue();
auto audio_receiver2 = audio_receiver_result2.MoveValue();
auto video_receiver2 = video_receiver_result2.MoveValue();
// Add fake tracks.
RTCError error = audio_sender1->SetTrack(
CreateLocalAudioTrack("audio", ortc_factory1_.get()));
EXPECT_TRUE(error.ok());
error = video_sender1->SetTrack(
CreateLocalVideoTrackAndFakeCapturer("video", ortc_factory1_.get()));
EXPECT_TRUE(error.ok());
error = audio_sender2->SetTrack(
CreateLocalAudioTrack("audio", ortc_factory2_.get()));
EXPECT_TRUE(error.ok());
error = video_sender2->SetTrack(
CreateLocalVideoTrackAndFakeCapturer("video", ortc_factory2_.get()));
EXPECT_TRUE(error.ok());
// "sent_X_parameters1" are the parameters that endpoint 1 sends with and
// endpoint 2 receives with.
RtpParameters sent_opus_parameters1 =
MakeMinimalOpusParametersWithSsrc(0xdeadbeef);
RtpParameters sent_vp8_parameters1 =
MakeMinimalVp8ParametersWithSsrc(0xbaadfeed);
RtpParameters sent_opus_parameters2 =
MakeMinimalOpusParametersWithSsrc(0x13333337);
RtpParameters sent_vp8_parameters2 =
MakeMinimalVp8ParametersWithSsrc(0x12345678);
// Configure the senders' and receivers' parameters.
EXPECT_TRUE(audio_receiver1->Receive(sent_opus_parameters2).ok());
EXPECT_TRUE(video_receiver1->Receive(sent_vp8_parameters2).ok());
EXPECT_TRUE(audio_receiver2->Receive(sent_opus_parameters1).ok());
EXPECT_TRUE(video_receiver2->Receive(sent_vp8_parameters1).ok());
EXPECT_TRUE(audio_sender1->Send(sent_opus_parameters1).ok());
EXPECT_TRUE(video_sender1->Send(sent_vp8_parameters1).ok());
EXPECT_TRUE(audio_sender2->Send(sent_opus_parameters2).ok());
EXPECT_TRUE(video_sender2->Send(sent_vp8_parameters2).ok());
FakeVideoTrackRenderer fake_video_renderer1(
static_cast<VideoTrackInterface*>(video_receiver1->GetTrack().get()));
FakeVideoTrackRenderer fake_video_renderer2(
static_cast<VideoTrackInterface*>(video_receiver2->GetTrack().get()));
// Senders and receivers are connected and configured; audio and video frames
// should be able to flow at this point.
EXPECT_TRUE_WAIT(
fake_audio_capture_module1_->frames_received() > kDefaultNumFrames &&
fake_video_renderer1.num_rendered_frames() > kDefaultNumFrames &&
fake_audio_capture_module2_->frames_received() > kDefaultNumFrames &&
fake_video_renderer2.num_rendered_frames() > kDefaultNumFrames,
kDefaultTimeout);
}
// End-to-end test with two pairs of RTP senders and receivers, for audio and
// video. Unlike the test above, this attempts to make the parameters as
// complex as possible.
//
// Uses non-muxed RTCP, with separate audio/video transports, and a full set of
// parameters, as would normally be used in a PeerConnection.
//
// TODO(deadbeef): Update this test as more audio/video features become
// supported.
TEST_F(OrtcFactoryIntegrationTest, FullTwoWayAudioVideoRtpSendersAndReceivers) {
// We want four pairs of UDP transports for this test, for audio/video and
// RTP/RTCP.
auto audio_rtp_udp_transports = CreateAndConnectUdpTransportPair();
auto audio_rtcp_udp_transports = CreateAndConnectUdpTransportPair();
auto video_rtp_udp_transports = CreateAndConnectUdpTransportPair();
auto video_rtcp_udp_transports = CreateAndConnectUdpTransportPair();
// Since we have multiple RTP transports on each side, we need an RTP
// transport controller.
auto transport_controllers = CreateRtpTransportControllerPair();
RtcpParameters audio_rtcp_parameters;
audio_rtcp_parameters.mux = false;
auto audio_rtp_transports =
CreateRtpTransportPair(audio_rtcp_parameters, audio_rtp_udp_transports,
audio_rtcp_udp_transports, transport_controllers);
RtcpParameters video_rtcp_parameters;
video_rtcp_parameters.mux = false;
video_rtcp_parameters.reduced_size = true;
auto video_rtp_transports =
CreateRtpTransportPair(video_rtcp_parameters, video_rtp_udp_transports,
video_rtcp_udp_transports, transport_controllers);
// Create all the senders and receivers (four per endpoint).
auto audio_sender_result1 = ortc_factory1_->CreateRtpSender(
cricket::MEDIA_TYPE_AUDIO, audio_rtp_transports.first.get());
auto video_sender_result1 = ortc_factory1_->CreateRtpSender(
cricket::MEDIA_TYPE_VIDEO, video_rtp_transports.first.get());
auto audio_receiver_result1 = ortc_factory1_->CreateRtpReceiver(
cricket::MEDIA_TYPE_AUDIO, audio_rtp_transports.first.get());
auto video_receiver_result1 = ortc_factory1_->CreateRtpReceiver(
cricket::MEDIA_TYPE_VIDEO, video_rtp_transports.first.get());
ASSERT_TRUE(audio_sender_result1.ok());
ASSERT_TRUE(video_sender_result1.ok());
ASSERT_TRUE(audio_receiver_result1.ok());
ASSERT_TRUE(video_receiver_result1.ok());
auto audio_sender1 = audio_sender_result1.MoveValue();
auto video_sender1 = video_sender_result1.MoveValue();
auto audio_receiver1 = audio_receiver_result1.MoveValue();
auto video_receiver1 = video_receiver_result1.MoveValue();
auto audio_sender_result2 = ortc_factory2_->CreateRtpSender(
cricket::MEDIA_TYPE_AUDIO, audio_rtp_transports.second.get());
auto video_sender_result2 = ortc_factory2_->CreateRtpSender(
cricket::MEDIA_TYPE_VIDEO, video_rtp_transports.second.get());
auto audio_receiver_result2 = ortc_factory2_->CreateRtpReceiver(
cricket::MEDIA_TYPE_AUDIO, audio_rtp_transports.second.get());
auto video_receiver_result2 = ortc_factory2_->CreateRtpReceiver(
cricket::MEDIA_TYPE_VIDEO, video_rtp_transports.second.get());
ASSERT_TRUE(audio_sender_result2.ok());
ASSERT_TRUE(video_sender_result2.ok());
ASSERT_TRUE(audio_receiver_result2.ok());
ASSERT_TRUE(video_receiver_result2.ok());
auto audio_sender2 = audio_sender_result2.MoveValue();
auto video_sender2 = video_sender_result2.MoveValue();
auto audio_receiver2 = audio_receiver_result2.MoveValue();
auto video_receiver2 = video_receiver_result2.MoveValue();
RTCError error = audio_sender1->SetTrack(
CreateLocalAudioTrack("audio", ortc_factory1_.get()));
EXPECT_TRUE(error.ok());
error = video_sender1->SetTrack(
CreateLocalVideoTrackAndFakeCapturer("video", ortc_factory1_.get()));
EXPECT_TRUE(error.ok());
error = audio_sender2->SetTrack(
CreateLocalAudioTrack("audio", ortc_factory2_.get()));
EXPECT_TRUE(error.ok());
error = video_sender2->SetTrack(
CreateLocalVideoTrackAndFakeCapturer("video", ortc_factory2_.get()));
EXPECT_TRUE(error.ok());
// Use different codecs in different directions for extra challenge.
RtpParameters opus_send_parameters = MakeFullOpusParameters();
RtpParameters isac_send_parameters = MakeFullIsacParameters();
RtpParameters vp8_send_parameters = MakeFullVp8Parameters();
RtpParameters vp9_send_parameters = MakeFullVp9Parameters();
// Remove "payload_type" from receive parameters. Receiver will need to
// discern the payload type from packets received.
RtpParameters opus_receive_parameters = opus_send_parameters;
RtpParameters isac_receive_parameters = isac_send_parameters;
RtpParameters vp8_receive_parameters = vp8_send_parameters;
RtpParameters vp9_receive_parameters = vp9_send_parameters;
opus_receive_parameters.encodings[0].codec_payload_type.reset();
isac_receive_parameters.encodings[0].codec_payload_type.reset();
vp8_receive_parameters.encodings[0].codec_payload_type.reset();
vp9_receive_parameters.encodings[0].codec_payload_type.reset();
// Configure the senders' and receivers' parameters.
//
// Note: Intentionally, the top codec in the receive parameters does not
// match the codec sent by the other side. If "Receive" is called with a list
// of codecs, the receiver should be prepared to receive any of them, not
// just the one on top.
EXPECT_TRUE(audio_receiver1->Receive(opus_receive_parameters).ok());
EXPECT_TRUE(video_receiver1->Receive(vp8_receive_parameters).ok());
EXPECT_TRUE(audio_receiver2->Receive(isac_receive_parameters).ok());
EXPECT_TRUE(video_receiver2->Receive(vp9_receive_parameters).ok());
EXPECT_TRUE(audio_sender1->Send(opus_send_parameters).ok());
EXPECT_TRUE(video_sender1->Send(vp8_send_parameters).ok());
EXPECT_TRUE(audio_sender2->Send(isac_send_parameters).ok());
EXPECT_TRUE(video_sender2->Send(vp9_send_parameters).ok());
FakeVideoTrackRenderer fake_video_renderer1(
static_cast<VideoTrackInterface*>(video_receiver1->GetTrack().get()));
FakeVideoTrackRenderer fake_video_renderer2(
static_cast<VideoTrackInterface*>(video_receiver2->GetTrack().get()));
// Senders and receivers are connected and configured; audio and video frames
// should be able to flow at this point.
EXPECT_TRUE_WAIT(
fake_audio_capture_module1_->frames_received() > kDefaultNumFrames &&
fake_video_renderer1.num_rendered_frames() > kDefaultNumFrames &&
fake_audio_capture_module2_->frames_received() > kDefaultNumFrames &&
fake_video_renderer2.num_rendered_frames() > kDefaultNumFrames,
kDefaultTimeout);
}
// TODO(deadbeef): End-to-end test for multiple senders/receivers of the same
// media type, once that's supported. Currently, it is not because the
// BaseChannel model relies on there being a single VoiceChannel and
// VideoChannel, and these only support a single set of codecs/etc. per
// send/receive direction.
// TODO(deadbeef): End-to-end test for simulcast, once that's supported by this
// API.
} // namespace webrtc
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