761 lines
30 KiB
C++
761 lines
30 KiB
C++
/*
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* Copyright 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <memory>
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#include "absl/strings/match.h"
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#include "api/audio_codecs/L16/audio_decoder_L16.h"
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#include "api/audio_codecs/L16/audio_encoder_L16.h"
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#include "api/audio_codecs/audio_codec_pair_id.h"
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#include "api/audio_codecs/audio_decoder_factory_template.h"
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#include "api/audio_codecs/audio_encoder_factory_template.h"
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#include "api/audio_codecs/opus_audio_decoder_factory.h"
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#include "api/audio_codecs/opus_audio_encoder_factory.h"
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#include "media/sctp/sctp_transport_internal.h"
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#include "rtc_base/gunit.h"
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#include "rtc_base/logging.h"
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#ifdef WEBRTC_ANDROID
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#include "pc/test/android_test_initializer.h"
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#endif
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#include "pc/test/peer_connection_test_wrapper.h"
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// Notice that mockpeerconnectionobservers.h must be included after the above!
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#include "pc/test/mock_peer_connection_observers.h"
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#include "test/mock_audio_decoder.h"
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#include "test/mock_audio_decoder_factory.h"
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#include "test/mock_audio_encoder_factory.h"
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using ::testing::_;
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using ::testing::AtLeast;
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using ::testing::Invoke;
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using ::testing::StrictMock;
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using ::testing::Values;
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using webrtc::DataChannelInterface;
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using webrtc::MediaStreamInterface;
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using webrtc::PeerConnectionInterface;
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using webrtc::SdpSemantics;
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namespace {
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const int kMaxWait = 25000;
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} // namespace
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class PeerConnectionEndToEndBaseTest : public sigslot::has_slots<>,
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public ::testing::Test {
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public:
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typedef std::vector<rtc::scoped_refptr<DataChannelInterface>> DataChannelList;
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explicit PeerConnectionEndToEndBaseTest(SdpSemantics sdp_semantics) {
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network_thread_ = rtc::Thread::CreateWithSocketServer();
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worker_thread_ = rtc::Thread::Create();
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RTC_CHECK(network_thread_->Start());
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RTC_CHECK(worker_thread_->Start());
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caller_ = new rtc::RefCountedObject<PeerConnectionTestWrapper>(
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"caller", network_thread_.get(), worker_thread_.get());
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callee_ = new rtc::RefCountedObject<PeerConnectionTestWrapper>(
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"callee", network_thread_.get(), worker_thread_.get());
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webrtc::PeerConnectionInterface::IceServer ice_server;
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ice_server.uri = "stun:stun.l.google.com:19302";
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config_.servers.push_back(ice_server);
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config_.sdp_semantics = sdp_semantics;
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#ifdef WEBRTC_ANDROID
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webrtc::InitializeAndroidObjects();
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#endif
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}
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void CreatePcs(
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rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory1,
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rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory1,
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rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory2,
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rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory2) {
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EXPECT_TRUE(caller_->CreatePc(config_, audio_encoder_factory1,
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audio_decoder_factory1));
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EXPECT_TRUE(callee_->CreatePc(config_, audio_encoder_factory2,
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audio_decoder_factory2));
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PeerConnectionTestWrapper::Connect(caller_.get(), callee_.get());
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caller_->SignalOnDataChannel.connect(
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this, &PeerConnectionEndToEndBaseTest::OnCallerAddedDataChanel);
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callee_->SignalOnDataChannel.connect(
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this, &PeerConnectionEndToEndBaseTest::OnCalleeAddedDataChannel);
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}
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void CreatePcs(
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rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory,
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rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory) {
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CreatePcs(audio_encoder_factory, audio_decoder_factory,
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audio_encoder_factory, audio_decoder_factory);
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}
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void GetAndAddUserMedia() {
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cricket::AudioOptions audio_options;
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GetAndAddUserMedia(true, audio_options, true);
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}
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void GetAndAddUserMedia(bool audio,
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const cricket::AudioOptions& audio_options,
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bool video) {
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caller_->GetAndAddUserMedia(audio, audio_options, video);
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callee_->GetAndAddUserMedia(audio, audio_options, video);
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}
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void Negotiate() {
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caller_->CreateOffer(
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webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());
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}
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void WaitForCallEstablished() {
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caller_->WaitForCallEstablished();
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callee_->WaitForCallEstablished();
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}
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void WaitForConnection() {
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caller_->WaitForConnection();
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callee_->WaitForConnection();
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}
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void OnCallerAddedDataChanel(DataChannelInterface* dc) {
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caller_signaled_data_channels_.push_back(dc);
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}
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void OnCalleeAddedDataChannel(DataChannelInterface* dc) {
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callee_signaled_data_channels_.push_back(dc);
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}
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// Tests that |dc1| and |dc2| can send to and receive from each other.
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void TestDataChannelSendAndReceive(DataChannelInterface* dc1,
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DataChannelInterface* dc2,
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size_t size = 6) {
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std::unique_ptr<webrtc::MockDataChannelObserver> dc1_observer(
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new webrtc::MockDataChannelObserver(dc1));
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std::unique_ptr<webrtc::MockDataChannelObserver> dc2_observer(
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new webrtc::MockDataChannelObserver(dc2));
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static const std::string kDummyData =
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"ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789+/";
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webrtc::DataBuffer buffer("");
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size_t sizeLeft = size;
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while (sizeLeft > 0) {
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size_t chunkSize =
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sizeLeft > kDummyData.length() ? kDummyData.length() : sizeLeft;
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buffer.data.AppendData(kDummyData.data(), chunkSize);
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sizeLeft -= chunkSize;
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}
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EXPECT_TRUE(dc1->Send(buffer));
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EXPECT_EQ_WAIT(buffer.data,
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rtc::CopyOnWriteBuffer(dc2_observer->last_message()),
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kMaxWait);
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EXPECT_TRUE(dc2->Send(buffer));
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EXPECT_EQ_WAIT(buffer.data,
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rtc::CopyOnWriteBuffer(dc1_observer->last_message()),
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kMaxWait);
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EXPECT_EQ(1U, dc1_observer->received_message_count());
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EXPECT_EQ(size, dc1_observer->last_message().length());
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EXPECT_EQ(1U, dc2_observer->received_message_count());
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EXPECT_EQ(size, dc2_observer->last_message().length());
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}
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void WaitForDataChannelsToOpen(DataChannelInterface* local_dc,
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const DataChannelList& remote_dc_list,
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size_t remote_dc_index) {
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EXPECT_EQ_WAIT(DataChannelInterface::kOpen, local_dc->state(), kMaxWait);
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ASSERT_TRUE_WAIT(remote_dc_list.size() > remote_dc_index, kMaxWait);
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EXPECT_EQ_WAIT(DataChannelInterface::kOpen,
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remote_dc_list[remote_dc_index]->state(), kMaxWait);
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EXPECT_EQ(local_dc->id(), remote_dc_list[remote_dc_index]->id());
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}
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void CloseDataChannels(DataChannelInterface* local_dc,
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const DataChannelList& remote_dc_list,
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size_t remote_dc_index) {
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local_dc->Close();
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EXPECT_EQ_WAIT(DataChannelInterface::kClosed, local_dc->state(), kMaxWait);
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EXPECT_EQ_WAIT(DataChannelInterface::kClosed,
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remote_dc_list[remote_dc_index]->state(), kMaxWait);
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}
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protected:
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std::unique_ptr<rtc::Thread> network_thread_;
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std::unique_ptr<rtc::Thread> worker_thread_;
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rtc::scoped_refptr<PeerConnectionTestWrapper> caller_;
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rtc::scoped_refptr<PeerConnectionTestWrapper> callee_;
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DataChannelList caller_signaled_data_channels_;
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DataChannelList callee_signaled_data_channels_;
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webrtc::PeerConnectionInterface::RTCConfiguration config_;
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};
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class PeerConnectionEndToEndTest
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: public PeerConnectionEndToEndBaseTest,
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public ::testing::WithParamInterface<SdpSemantics> {
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protected:
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PeerConnectionEndToEndTest() : PeerConnectionEndToEndBaseTest(GetParam()) {}
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};
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namespace {
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std::unique_ptr<webrtc::AudioDecoder> CreateForwardingMockDecoder(
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std::unique_ptr<webrtc::AudioDecoder> real_decoder) {
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class ForwardingMockDecoder : public StrictMock<webrtc::MockAudioDecoder> {
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public:
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explicit ForwardingMockDecoder(std::unique_ptr<AudioDecoder> decoder)
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: decoder_(std::move(decoder)) {}
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private:
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std::unique_ptr<AudioDecoder> decoder_;
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};
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const auto dec = real_decoder.get(); // For lambda capturing.
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auto mock_decoder =
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std::make_unique<ForwardingMockDecoder>(std::move(real_decoder));
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EXPECT_CALL(*mock_decoder, Channels())
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.Times(AtLeast(1))
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.WillRepeatedly(Invoke([dec] { return dec->Channels(); }));
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EXPECT_CALL(*mock_decoder, DecodeInternal(_, _, _, _, _))
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.Times(AtLeast(1))
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.WillRepeatedly(
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Invoke([dec](const uint8_t* encoded, size_t encoded_len,
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int sample_rate_hz, int16_t* decoded,
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webrtc::AudioDecoder::SpeechType* speech_type) {
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return dec->Decode(encoded, encoded_len, sample_rate_hz,
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std::numeric_limits<size_t>::max(), decoded,
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speech_type);
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}));
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EXPECT_CALL(*mock_decoder, Die());
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EXPECT_CALL(*mock_decoder, HasDecodePlc()).WillRepeatedly(Invoke([dec] {
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return dec->HasDecodePlc();
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}));
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EXPECT_CALL(*mock_decoder, PacketDuration(_, _))
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.Times(AtLeast(1))
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.WillRepeatedly(Invoke([dec](const uint8_t* encoded, size_t encoded_len) {
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return dec->PacketDuration(encoded, encoded_len);
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}));
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EXPECT_CALL(*mock_decoder, SampleRateHz())
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.Times(AtLeast(1))
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.WillRepeatedly(Invoke([dec] { return dec->SampleRateHz(); }));
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return std::move(mock_decoder);
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}
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rtc::scoped_refptr<webrtc::AudioDecoderFactory>
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CreateForwardingMockDecoderFactory(
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webrtc::AudioDecoderFactory* real_decoder_factory) {
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rtc::scoped_refptr<webrtc::MockAudioDecoderFactory> mock_decoder_factory =
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new rtc::RefCountedObject<StrictMock<webrtc::MockAudioDecoderFactory>>;
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EXPECT_CALL(*mock_decoder_factory, GetSupportedDecoders())
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.Times(AtLeast(1))
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.WillRepeatedly(Invoke([real_decoder_factory] {
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return real_decoder_factory->GetSupportedDecoders();
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}));
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EXPECT_CALL(*mock_decoder_factory, IsSupportedDecoder(_))
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.Times(AtLeast(1))
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.WillRepeatedly(
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Invoke([real_decoder_factory](const webrtc::SdpAudioFormat& format) {
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return real_decoder_factory->IsSupportedDecoder(format);
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}));
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EXPECT_CALL(*mock_decoder_factory, MakeAudioDecoderMock(_, _, _))
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.Times(AtLeast(2))
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.WillRepeatedly(
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Invoke([real_decoder_factory](
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const webrtc::SdpAudioFormat& format,
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absl::optional<webrtc::AudioCodecPairId> codec_pair_id,
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std::unique_ptr<webrtc::AudioDecoder>* return_value) {
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auto real_decoder =
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real_decoder_factory->MakeAudioDecoder(format, codec_pair_id);
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*return_value =
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real_decoder
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? CreateForwardingMockDecoder(std::move(real_decoder))
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: nullptr;
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}));
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return mock_decoder_factory;
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}
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struct AudioEncoderUnicornSparklesRainbow {
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using Config = webrtc::AudioEncoderL16::Config;
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static absl::optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
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if (absl::EqualsIgnoreCase(format.name, "UnicornSparklesRainbow")) {
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const webrtc::SdpAudioFormat::Parameters expected_params = {
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{"num_horns", "1"}};
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EXPECT_EQ(expected_params, format.parameters);
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format.parameters.clear();
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format.name = "L16";
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return webrtc::AudioEncoderL16::SdpToConfig(format);
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} else {
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return absl::nullopt;
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}
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}
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static void AppendSupportedEncoders(
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std::vector<webrtc::AudioCodecSpec>* specs) {
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std::vector<webrtc::AudioCodecSpec> new_specs;
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webrtc::AudioEncoderL16::AppendSupportedEncoders(&new_specs);
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for (auto& spec : new_specs) {
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spec.format.name = "UnicornSparklesRainbow";
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EXPECT_TRUE(spec.format.parameters.empty());
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spec.format.parameters.emplace("num_horns", "1");
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specs->push_back(spec);
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}
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}
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static webrtc::AudioCodecInfo QueryAudioEncoder(const Config& config) {
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return webrtc::AudioEncoderL16::QueryAudioEncoder(config);
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}
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static std::unique_ptr<webrtc::AudioEncoder> MakeAudioEncoder(
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const Config& config,
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int payload_type,
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absl::optional<webrtc::AudioCodecPairId> codec_pair_id = absl::nullopt) {
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return webrtc::AudioEncoderL16::MakeAudioEncoder(config, payload_type,
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codec_pair_id);
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}
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};
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struct AudioDecoderUnicornSparklesRainbow {
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using Config = webrtc::AudioDecoderL16::Config;
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static absl::optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
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if (absl::EqualsIgnoreCase(format.name, "UnicornSparklesRainbow")) {
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const webrtc::SdpAudioFormat::Parameters expected_params = {
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{"num_horns", "1"}};
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EXPECT_EQ(expected_params, format.parameters);
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format.parameters.clear();
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format.name = "L16";
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return webrtc::AudioDecoderL16::SdpToConfig(format);
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} else {
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return absl::nullopt;
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}
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}
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static void AppendSupportedDecoders(
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std::vector<webrtc::AudioCodecSpec>* specs) {
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std::vector<webrtc::AudioCodecSpec> new_specs;
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webrtc::AudioDecoderL16::AppendSupportedDecoders(&new_specs);
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for (auto& spec : new_specs) {
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spec.format.name = "UnicornSparklesRainbow";
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EXPECT_TRUE(spec.format.parameters.empty());
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spec.format.parameters.emplace("num_horns", "1");
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specs->push_back(spec);
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}
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}
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static std::unique_ptr<webrtc::AudioDecoder> MakeAudioDecoder(
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const Config& config,
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absl::optional<webrtc::AudioCodecPairId> codec_pair_id = absl::nullopt) {
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return webrtc::AudioDecoderL16::MakeAudioDecoder(config, codec_pair_id);
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}
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};
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} // namespace
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TEST_P(PeerConnectionEndToEndTest, Call) {
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rtc::scoped_refptr<webrtc::AudioDecoderFactory> real_decoder_factory =
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webrtc::CreateOpusAudioDecoderFactory();
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CreatePcs(webrtc::CreateOpusAudioEncoderFactory(),
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CreateForwardingMockDecoderFactory(real_decoder_factory.get()));
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GetAndAddUserMedia();
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Negotiate();
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WaitForCallEstablished();
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}
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TEST_P(PeerConnectionEndToEndTest, CallWithSdesKeyNegotiation) {
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config_.enable_dtls_srtp = false;
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CreatePcs(webrtc::CreateOpusAudioEncoderFactory(),
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webrtc::CreateOpusAudioDecoderFactory());
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GetAndAddUserMedia();
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Negotiate();
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WaitForCallEstablished();
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}
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TEST_P(PeerConnectionEndToEndTest, CallWithCustomCodec) {
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class IdLoggingAudioEncoderFactory : public webrtc::AudioEncoderFactory {
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public:
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IdLoggingAudioEncoderFactory(
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rtc::scoped_refptr<AudioEncoderFactory> real_factory,
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std::vector<webrtc::AudioCodecPairId>* const codec_ids)
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: fact_(real_factory), codec_ids_(codec_ids) {}
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std::vector<webrtc::AudioCodecSpec> GetSupportedEncoders() override {
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return fact_->GetSupportedEncoders();
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}
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absl::optional<webrtc::AudioCodecInfo> QueryAudioEncoder(
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const webrtc::SdpAudioFormat& format) override {
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return fact_->QueryAudioEncoder(format);
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}
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std::unique_ptr<webrtc::AudioEncoder> MakeAudioEncoder(
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int payload_type,
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const webrtc::SdpAudioFormat& format,
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absl::optional<webrtc::AudioCodecPairId> codec_pair_id) override {
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EXPECT_TRUE(codec_pair_id.has_value());
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codec_ids_->push_back(*codec_pair_id);
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return fact_->MakeAudioEncoder(payload_type, format, codec_pair_id);
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}
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private:
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const rtc::scoped_refptr<webrtc::AudioEncoderFactory> fact_;
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std::vector<webrtc::AudioCodecPairId>* const codec_ids_;
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};
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class IdLoggingAudioDecoderFactory : public webrtc::AudioDecoderFactory {
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public:
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IdLoggingAudioDecoderFactory(
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rtc::scoped_refptr<AudioDecoderFactory> real_factory,
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std::vector<webrtc::AudioCodecPairId>* const codec_ids)
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: fact_(real_factory), codec_ids_(codec_ids) {}
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std::vector<webrtc::AudioCodecSpec> GetSupportedDecoders() override {
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return fact_->GetSupportedDecoders();
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}
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bool IsSupportedDecoder(const webrtc::SdpAudioFormat& format) override {
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return fact_->IsSupportedDecoder(format);
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}
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std::unique_ptr<webrtc::AudioDecoder> MakeAudioDecoder(
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const webrtc::SdpAudioFormat& format,
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absl::optional<webrtc::AudioCodecPairId> codec_pair_id) override {
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EXPECT_TRUE(codec_pair_id.has_value());
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codec_ids_->push_back(*codec_pair_id);
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return fact_->MakeAudioDecoder(format, codec_pair_id);
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}
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private:
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const rtc::scoped_refptr<webrtc::AudioDecoderFactory> fact_;
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std::vector<webrtc::AudioCodecPairId>* const codec_ids_;
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};
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std::vector<webrtc::AudioCodecPairId> encoder_id1, encoder_id2, decoder_id1,
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decoder_id2;
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CreatePcs(rtc::scoped_refptr<webrtc::AudioEncoderFactory>(
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new rtc::RefCountedObject<IdLoggingAudioEncoderFactory>(
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webrtc::CreateAudioEncoderFactory<
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AudioEncoderUnicornSparklesRainbow>(),
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&encoder_id1)),
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rtc::scoped_refptr<webrtc::AudioDecoderFactory>(
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new rtc::RefCountedObject<IdLoggingAudioDecoderFactory>(
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webrtc::CreateAudioDecoderFactory<
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AudioDecoderUnicornSparklesRainbow>(),
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&decoder_id1)),
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rtc::scoped_refptr<webrtc::AudioEncoderFactory>(
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new rtc::RefCountedObject<IdLoggingAudioEncoderFactory>(
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webrtc::CreateAudioEncoderFactory<
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AudioEncoderUnicornSparklesRainbow>(),
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&encoder_id2)),
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rtc::scoped_refptr<webrtc::AudioDecoderFactory>(
|
|
new rtc::RefCountedObject<IdLoggingAudioDecoderFactory>(
|
|
webrtc::CreateAudioDecoderFactory<
|
|
AudioDecoderUnicornSparklesRainbow>(),
|
|
&decoder_id2)));
|
|
GetAndAddUserMedia();
|
|
Negotiate();
|
|
WaitForCallEstablished();
|
|
|
|
// Each codec factory has been used to create one codec. The first pair got
|
|
// the same ID because they were passed to the same PeerConnectionFactory,
|
|
// and the second pair got the same ID---but these two IDs are not equal,
|
|
// because each PeerConnectionFactory has its own ID.
|
|
EXPECT_EQ(1U, encoder_id1.size());
|
|
EXPECT_EQ(1U, encoder_id2.size());
|
|
EXPECT_EQ(encoder_id1, decoder_id1);
|
|
EXPECT_EQ(encoder_id2, decoder_id2);
|
|
EXPECT_NE(encoder_id1, encoder_id2);
|
|
}
|
|
|
|
#ifdef HAVE_SCTP
|
|
// Verifies that a DataChannel created before the negotiation can transition to
|
|
// "OPEN" and transfer data.
|
|
TEST_P(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
|
|
CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
|
|
webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
|
|
|
|
webrtc::DataChannelInit init;
|
|
rtc::scoped_refptr<DataChannelInterface> caller_dc(
|
|
caller_->CreateDataChannel("data", init));
|
|
rtc::scoped_refptr<DataChannelInterface> callee_dc(
|
|
callee_->CreateDataChannel("data", init));
|
|
|
|
Negotiate();
|
|
WaitForConnection();
|
|
|
|
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
|
|
WaitForDataChannelsToOpen(callee_dc, caller_signaled_data_channels_, 0);
|
|
|
|
TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[0]);
|
|
TestDataChannelSendAndReceive(callee_dc, caller_signaled_data_channels_[0]);
|
|
|
|
CloseDataChannels(caller_dc, callee_signaled_data_channels_, 0);
|
|
CloseDataChannels(callee_dc, caller_signaled_data_channels_, 0);
|
|
}
|
|
|
|
// Verifies that a DataChannel created after the negotiation can transition to
|
|
// "OPEN" and transfer data.
|
|
TEST_P(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
|
|
CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
|
|
webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
|
|
|
|
webrtc::DataChannelInit init;
|
|
|
|
// This DataChannel is for creating the data content in the negotiation.
|
|
rtc::scoped_refptr<DataChannelInterface> dummy(
|
|
caller_->CreateDataChannel("data", init));
|
|
Negotiate();
|
|
WaitForConnection();
|
|
|
|
// Wait for the data channel created pre-negotiation to be opened.
|
|
WaitForDataChannelsToOpen(dummy, callee_signaled_data_channels_, 0);
|
|
|
|
// Create new DataChannels after the negotiation and verify their states.
|
|
rtc::scoped_refptr<DataChannelInterface> caller_dc(
|
|
caller_->CreateDataChannel("hello", init));
|
|
rtc::scoped_refptr<DataChannelInterface> callee_dc(
|
|
callee_->CreateDataChannel("hello", init));
|
|
|
|
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1);
|
|
WaitForDataChannelsToOpen(callee_dc, caller_signaled_data_channels_, 0);
|
|
|
|
TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1]);
|
|
TestDataChannelSendAndReceive(callee_dc, caller_signaled_data_channels_[0]);
|
|
|
|
CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1);
|
|
CloseDataChannels(callee_dc, caller_signaled_data_channels_, 0);
|
|
}
|
|
|
|
// Verifies that a DataChannel created can transfer large messages.
|
|
TEST_P(PeerConnectionEndToEndTest, CreateDataChannelLargeTransfer) {
|
|
CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
|
|
webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
|
|
|
|
webrtc::DataChannelInit init;
|
|
|
|
// This DataChannel is for creating the data content in the negotiation.
|
|
rtc::scoped_refptr<DataChannelInterface> dummy(
|
|
caller_->CreateDataChannel("data", init));
|
|
Negotiate();
|
|
WaitForConnection();
|
|
|
|
// Wait for the data channel created pre-negotiation to be opened.
|
|
WaitForDataChannelsToOpen(dummy, callee_signaled_data_channels_, 0);
|
|
|
|
// Create new DataChannels after the negotiation and verify their states.
|
|
rtc::scoped_refptr<DataChannelInterface> caller_dc(
|
|
caller_->CreateDataChannel("hello", init));
|
|
rtc::scoped_refptr<DataChannelInterface> callee_dc(
|
|
callee_->CreateDataChannel("hello", init));
|
|
|
|
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1);
|
|
WaitForDataChannelsToOpen(callee_dc, caller_signaled_data_channels_, 0);
|
|
|
|
TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1],
|
|
256 * 1024);
|
|
TestDataChannelSendAndReceive(callee_dc, caller_signaled_data_channels_[0],
|
|
256 * 1024);
|
|
|
|
CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1);
|
|
CloseDataChannels(callee_dc, caller_signaled_data_channels_, 0);
|
|
}
|
|
|
|
// Verifies that DataChannel IDs are even/odd based on the DTLS roles.
|
|
TEST_P(PeerConnectionEndToEndTest, DataChannelIdAssignment) {
|
|
CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
|
|
webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
|
|
|
|
webrtc::DataChannelInit init;
|
|
rtc::scoped_refptr<DataChannelInterface> caller_dc_1(
|
|
caller_->CreateDataChannel("data", init));
|
|
rtc::scoped_refptr<DataChannelInterface> callee_dc_1(
|
|
callee_->CreateDataChannel("data", init));
|
|
|
|
Negotiate();
|
|
WaitForConnection();
|
|
|
|
EXPECT_EQ(1, caller_dc_1->id() % 2);
|
|
EXPECT_EQ(0, callee_dc_1->id() % 2);
|
|
|
|
rtc::scoped_refptr<DataChannelInterface> caller_dc_2(
|
|
caller_->CreateDataChannel("data", init));
|
|
rtc::scoped_refptr<DataChannelInterface> callee_dc_2(
|
|
callee_->CreateDataChannel("data", init));
|
|
|
|
EXPECT_EQ(1, caller_dc_2->id() % 2);
|
|
EXPECT_EQ(0, callee_dc_2->id() % 2);
|
|
}
|
|
|
|
// Verifies that the message is received by the right remote DataChannel when
|
|
// there are multiple DataChannels.
|
|
TEST_P(PeerConnectionEndToEndTest,
|
|
MessageTransferBetweenTwoPairsOfDataChannels) {
|
|
CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
|
|
webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
|
|
|
|
webrtc::DataChannelInit init;
|
|
|
|
rtc::scoped_refptr<DataChannelInterface> caller_dc_1(
|
|
caller_->CreateDataChannel("data", init));
|
|
rtc::scoped_refptr<DataChannelInterface> caller_dc_2(
|
|
caller_->CreateDataChannel("data", init));
|
|
|
|
Negotiate();
|
|
WaitForConnection();
|
|
WaitForDataChannelsToOpen(caller_dc_1, callee_signaled_data_channels_, 0);
|
|
WaitForDataChannelsToOpen(caller_dc_2, callee_signaled_data_channels_, 1);
|
|
|
|
std::unique_ptr<webrtc::MockDataChannelObserver> dc_1_observer(
|
|
new webrtc::MockDataChannelObserver(callee_signaled_data_channels_[0]));
|
|
|
|
std::unique_ptr<webrtc::MockDataChannelObserver> dc_2_observer(
|
|
new webrtc::MockDataChannelObserver(callee_signaled_data_channels_[1]));
|
|
|
|
const std::string message_1 = "hello 1";
|
|
const std::string message_2 = "hello 2";
|
|
|
|
caller_dc_1->Send(webrtc::DataBuffer(message_1));
|
|
EXPECT_EQ_WAIT(message_1, dc_1_observer->last_message(), kMaxWait);
|
|
|
|
caller_dc_2->Send(webrtc::DataBuffer(message_2));
|
|
EXPECT_EQ_WAIT(message_2, dc_2_observer->last_message(), kMaxWait);
|
|
|
|
EXPECT_EQ(1U, dc_1_observer->received_message_count());
|
|
EXPECT_EQ(1U, dc_2_observer->received_message_count());
|
|
}
|
|
|
|
// Verifies that a DataChannel added from an OPEN message functions after
|
|
// a channel has been previously closed (webrtc issue 3778).
|
|
// This previously failed because the new channel re-used the ID of the closed
|
|
// channel, and the closed channel was incorrectly still assigned to the ID.
|
|
TEST_P(PeerConnectionEndToEndTest,
|
|
DataChannelFromOpenWorksAfterPreviousChannelClosed) {
|
|
CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
|
|
webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
|
|
|
|
webrtc::DataChannelInit init;
|
|
rtc::scoped_refptr<DataChannelInterface> caller_dc(
|
|
caller_->CreateDataChannel("data", init));
|
|
|
|
Negotiate();
|
|
WaitForConnection();
|
|
|
|
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
|
|
int first_channel_id = caller_dc->id();
|
|
// Wait for the local side to say it's closed, but not the remote side.
|
|
// Previously, the channel on which Close is called reported being closed
|
|
// prematurely, and this caused issues; see bugs.webrtc.org/4453.
|
|
caller_dc->Close();
|
|
EXPECT_EQ_WAIT(DataChannelInterface::kClosed, caller_dc->state(), kMaxWait);
|
|
|
|
// Create a new channel and ensure it works after closing the previous one.
|
|
caller_dc = caller_->CreateDataChannel("data2", init);
|
|
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1);
|
|
// Since the second channel was created after the first finished closing, it
|
|
// should be able to re-use the first one's ID.
|
|
EXPECT_EQ(first_channel_id, caller_dc->id());
|
|
TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1]);
|
|
|
|
CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1);
|
|
}
|
|
|
|
// Similar to the above test, but don't wait for the first channel to finish
|
|
// closing before creating the second one.
|
|
TEST_P(PeerConnectionEndToEndTest,
|
|
DataChannelFromOpenWorksWhilePreviousChannelClosing) {
|
|
CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
|
|
webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
|
|
|
|
webrtc::DataChannelInit init;
|
|
rtc::scoped_refptr<DataChannelInterface> caller_dc(
|
|
caller_->CreateDataChannel("data", init));
|
|
|
|
Negotiate();
|
|
WaitForConnection();
|
|
|
|
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
|
|
int first_channel_id = caller_dc->id();
|
|
caller_dc->Close();
|
|
|
|
// Immediately create a new channel, before waiting for the previous one to
|
|
// transition to "closed".
|
|
caller_dc = caller_->CreateDataChannel("data2", init);
|
|
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1);
|
|
// Since the second channel was created while the first was still closing,
|
|
// it should have been assigned a different ID.
|
|
EXPECT_NE(first_channel_id, caller_dc->id());
|
|
TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1]);
|
|
|
|
CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1);
|
|
}
|
|
|
|
// This tests that if a data channel is closed remotely while not referenced
|
|
// by the application (meaning only the PeerConnection contributes to its
|
|
// reference count), no memory access violation will occur.
|
|
// See: https://code.google.com/p/chromium/issues/detail?id=565048
|
|
TEST_P(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) {
|
|
CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
|
|
webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
|
|
|
|
webrtc::DataChannelInit init;
|
|
rtc::scoped_refptr<DataChannelInterface> caller_dc(
|
|
caller_->CreateDataChannel("data", init));
|
|
|
|
Negotiate();
|
|
WaitForConnection();
|
|
|
|
WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
|
|
// This removes the reference to the remote data channel that we hold.
|
|
callee_signaled_data_channels_.clear();
|
|
caller_dc->Close();
|
|
EXPECT_EQ_WAIT(DataChannelInterface::kClosed, caller_dc->state(), kMaxWait);
|
|
|
|
// Wait for a bit longer so the remote data channel will receive the
|
|
// close message and be destroyed.
|
|
rtc::Thread::Current()->ProcessMessages(100);
|
|
}
|
|
|
|
// Test behavior of creating too many datachannels.
|
|
TEST_P(PeerConnectionEndToEndTest, TooManyDataChannelsOpenedBeforeConnecting) {
|
|
CreatePcs(webrtc::MockAudioEncoderFactory::CreateEmptyFactory(),
|
|
webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
|
|
|
|
webrtc::DataChannelInit init;
|
|
std::vector<rtc::scoped_refptr<DataChannelInterface>> channels;
|
|
for (int i = 0; i <= cricket::kMaxSctpStreams / 2; i++) {
|
|
rtc::scoped_refptr<DataChannelInterface> caller_dc(
|
|
caller_->CreateDataChannel("data", init));
|
|
channels.push_back(std::move(caller_dc));
|
|
}
|
|
Negotiate();
|
|
WaitForConnection();
|
|
EXPECT_EQ_WAIT(callee_signaled_data_channels_.size(),
|
|
static_cast<size_t>(cricket::kMaxSctpStreams / 2), kMaxWait);
|
|
EXPECT_EQ(DataChannelInterface::kOpen,
|
|
channels[(cricket::kMaxSctpStreams / 2) - 1]->state());
|
|
EXPECT_EQ(DataChannelInterface::kClosed,
|
|
channels[cricket::kMaxSctpStreams / 2]->state());
|
|
}
|
|
|
|
#endif // HAVE_SCTP
|
|
|
|
TEST_P(PeerConnectionEndToEndTest, CanRestartIce) {
|
|
rtc::scoped_refptr<webrtc::AudioDecoderFactory> real_decoder_factory =
|
|
webrtc::CreateOpusAudioDecoderFactory();
|
|
CreatePcs(webrtc::CreateOpusAudioEncoderFactory(),
|
|
CreateForwardingMockDecoderFactory(real_decoder_factory.get()));
|
|
GetAndAddUserMedia();
|
|
Negotiate();
|
|
WaitForCallEstablished();
|
|
// Cause ICE restart to be requested.
|
|
auto config = caller_->pc()->GetConfiguration();
|
|
ASSERT_NE(PeerConnectionInterface::kRelay, config.type);
|
|
config.type = PeerConnectionInterface::kRelay;
|
|
ASSERT_TRUE(caller_->pc()->SetConfiguration(config).ok());
|
|
// When solving https://crbug.com/webrtc/10504, all we need to check
|
|
// is that we do not crash. We should also be testing that restart happens.
|
|
}
|
|
|
|
INSTANTIATE_TEST_SUITE_P(PeerConnectionEndToEndTest,
|
|
PeerConnectionEndToEndTest,
|
|
Values(SdpSemantics::kPlanB,
|
|
SdpSemantics::kUnifiedPlan));
|