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346 lines
13 KiB
346 lines
13 KiB
/*
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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 "pc/test/peer_connection_test_wrapper.h"
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#include <stddef.h>
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#include <memory>
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#include <string>
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#include <utility>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/audio/audio_mixer.h"
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#include "api/create_peerconnection_factory.h"
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#include "api/video_codecs/builtin_video_decoder_factory.h"
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#include "api/video_codecs/builtin_video_encoder_factory.h"
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#include "api/video_codecs/video_decoder_factory.h"
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#include "api/video_codecs/video_encoder_factory.h"
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#include "modules/audio_device/include/audio_device.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "p2p/base/fake_port_allocator.h"
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#include "p2p/base/port_allocator.h"
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#include "pc/test/fake_periodic_video_source.h"
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#include "pc/test/fake_periodic_video_track_source.h"
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#include "pc/test/fake_rtc_certificate_generator.h"
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#include "pc/test/mock_peer_connection_observers.h"
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#include "rtc_base/gunit.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/ref_counted_object.h"
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#include "rtc_base/rtc_certificate_generator.h"
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#include "rtc_base/string_encode.h"
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#include "rtc_base/thread_checker.h"
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#include "rtc_base/time_utils.h"
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#include "test/gtest.h"
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using webrtc::FakeVideoTrackRenderer;
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using webrtc::IceCandidateInterface;
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using webrtc::MediaStreamInterface;
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using webrtc::MediaStreamTrackInterface;
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using webrtc::MockSetSessionDescriptionObserver;
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using webrtc::PeerConnectionInterface;
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using webrtc::RtpReceiverInterface;
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using webrtc::SdpType;
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using webrtc::SessionDescriptionInterface;
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using webrtc::VideoTrackInterface;
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namespace {
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const char kStreamIdBase[] = "stream_id";
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const char kVideoTrackLabelBase[] = "video_track";
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const char kAudioTrackLabelBase[] = "audio_track";
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constexpr int kMaxWait = 10000;
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constexpr int kTestAudioFrameCount = 3;
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constexpr int kTestVideoFrameCount = 3;
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} // namespace
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void PeerConnectionTestWrapper::Connect(PeerConnectionTestWrapper* caller,
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PeerConnectionTestWrapper* callee) {
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caller->SignalOnIceCandidateReady.connect(
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callee, &PeerConnectionTestWrapper::AddIceCandidate);
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callee->SignalOnIceCandidateReady.connect(
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caller, &PeerConnectionTestWrapper::AddIceCandidate);
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caller->SignalOnSdpReady.connect(callee,
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&PeerConnectionTestWrapper::ReceiveOfferSdp);
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callee->SignalOnSdpReady.connect(
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caller, &PeerConnectionTestWrapper::ReceiveAnswerSdp);
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}
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PeerConnectionTestWrapper::PeerConnectionTestWrapper(
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const std::string& name,
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rtc::Thread* network_thread,
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rtc::Thread* worker_thread)
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: name_(name),
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network_thread_(network_thread),
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worker_thread_(worker_thread),
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pending_negotiation_(false) {
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pc_thread_checker_.Detach();
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}
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PeerConnectionTestWrapper::~PeerConnectionTestWrapper() {
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RTC_DCHECK_RUN_ON(&pc_thread_checker_);
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// Either network_thread or worker_thread might be active at this point.
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// Relying on ~PeerConnection to properly wait for them doesn't work,
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// as a vptr race might occur (before we enter the destruction body).
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// See: bugs.webrtc.org/9847
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if (pc()) {
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pc()->Close();
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}
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}
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bool PeerConnectionTestWrapper::CreatePc(
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const webrtc::PeerConnectionInterface::RTCConfiguration& config,
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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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std::unique_ptr<cricket::PortAllocator> port_allocator(
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new cricket::FakePortAllocator(network_thread_, nullptr));
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RTC_DCHECK_RUN_ON(&pc_thread_checker_);
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fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
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if (fake_audio_capture_module_ == NULL) {
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return false;
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}
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peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
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network_thread_, worker_thread_, rtc::Thread::Current(),
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rtc::scoped_refptr<webrtc::AudioDeviceModule>(fake_audio_capture_module_),
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audio_encoder_factory, audio_decoder_factory,
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webrtc::CreateBuiltinVideoEncoderFactory(),
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webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
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nullptr /* audio_processing */);
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if (!peer_connection_factory_) {
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return false;
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}
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std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator(
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new FakeRTCCertificateGenerator());
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peer_connection_ = peer_connection_factory_->CreatePeerConnection(
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config, std::move(port_allocator), std::move(cert_generator), this);
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return peer_connection_.get() != NULL;
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}
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rtc::scoped_refptr<webrtc::DataChannelInterface>
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PeerConnectionTestWrapper::CreateDataChannel(
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const std::string& label,
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const webrtc::DataChannelInit& init) {
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return peer_connection_->CreateDataChannel(label, &init);
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}
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void PeerConnectionTestWrapper::WaitForNegotiation() {
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EXPECT_TRUE_WAIT(!pending_negotiation_, kMaxWait);
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}
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void PeerConnectionTestWrapper::OnSignalingChange(
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webrtc::PeerConnectionInterface::SignalingState new_state) {
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if (new_state == webrtc::PeerConnectionInterface::SignalingState::kStable) {
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pending_negotiation_ = false;
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}
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}
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void PeerConnectionTestWrapper::OnAddTrack(
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rtc::scoped_refptr<RtpReceiverInterface> receiver,
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const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {
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RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ << ": OnAddTrack";
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if (receiver->track()->kind() == MediaStreamTrackInterface::kVideoKind) {
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auto* video_track =
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static_cast<VideoTrackInterface*>(receiver->track().get());
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renderer_ = std::make_unique<FakeVideoTrackRenderer>(video_track);
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}
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}
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void PeerConnectionTestWrapper::OnIceCandidate(
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const IceCandidateInterface* candidate) {
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std::string sdp;
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EXPECT_TRUE(candidate->ToString(&sdp));
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// Give the user a chance to modify sdp for testing.
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SignalOnIceCandidateCreated(&sdp);
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SignalOnIceCandidateReady(candidate->sdp_mid(), candidate->sdp_mline_index(),
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sdp);
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}
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void PeerConnectionTestWrapper::OnDataChannel(
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rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) {
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SignalOnDataChannel(data_channel);
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}
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void PeerConnectionTestWrapper::OnSuccess(SessionDescriptionInterface* desc) {
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// This callback should take the ownership of |desc|.
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std::unique_ptr<SessionDescriptionInterface> owned_desc(desc);
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std::string sdp;
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EXPECT_TRUE(desc->ToString(&sdp));
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RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ << ": "
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<< webrtc::SdpTypeToString(desc->GetType())
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<< " sdp created: " << sdp;
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// Give the user a chance to modify sdp for testing.
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SignalOnSdpCreated(&sdp);
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SetLocalDescription(desc->GetType(), sdp);
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SignalOnSdpReady(sdp);
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}
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void PeerConnectionTestWrapper::CreateOffer(
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const webrtc::PeerConnectionInterface::RTCOfferAnswerOptions& options) {
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RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ << ": CreateOffer.";
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pending_negotiation_ = true;
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peer_connection_->CreateOffer(this, options);
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}
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void PeerConnectionTestWrapper::CreateAnswer(
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const webrtc::PeerConnectionInterface::RTCOfferAnswerOptions& options) {
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RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
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<< ": CreateAnswer.";
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pending_negotiation_ = true;
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peer_connection_->CreateAnswer(this, options);
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}
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void PeerConnectionTestWrapper::ReceiveOfferSdp(const std::string& sdp) {
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SetRemoteDescription(SdpType::kOffer, sdp);
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CreateAnswer(webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());
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}
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void PeerConnectionTestWrapper::ReceiveAnswerSdp(const std::string& sdp) {
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SetRemoteDescription(SdpType::kAnswer, sdp);
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}
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void PeerConnectionTestWrapper::SetLocalDescription(SdpType type,
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const std::string& sdp) {
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RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
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<< ": SetLocalDescription " << webrtc::SdpTypeToString(type)
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<< " " << sdp;
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rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
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new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
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peer_connection_->SetLocalDescription(
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observer, webrtc::CreateSessionDescription(type, sdp).release());
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}
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void PeerConnectionTestWrapper::SetRemoteDescription(SdpType type,
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const std::string& sdp) {
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RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
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<< ": SetRemoteDescription " << webrtc::SdpTypeToString(type)
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<< " " << sdp;
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rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
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new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
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peer_connection_->SetRemoteDescription(
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observer, webrtc::CreateSessionDescription(type, sdp).release());
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}
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void PeerConnectionTestWrapper::AddIceCandidate(const std::string& sdp_mid,
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int sdp_mline_index,
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const std::string& candidate) {
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std::unique_ptr<webrtc::IceCandidateInterface> owned_candidate(
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webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, candidate, NULL));
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EXPECT_TRUE(peer_connection_->AddIceCandidate(owned_candidate.get()));
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}
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void PeerConnectionTestWrapper::WaitForCallEstablished() {
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WaitForConnection();
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WaitForAudio();
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WaitForVideo();
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}
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void PeerConnectionTestWrapper::WaitForConnection() {
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EXPECT_TRUE_WAIT(CheckForConnection(), kMaxWait);
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RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ << ": Connected.";
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}
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bool PeerConnectionTestWrapper::CheckForConnection() {
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return (peer_connection_->ice_connection_state() ==
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PeerConnectionInterface::kIceConnectionConnected) ||
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(peer_connection_->ice_connection_state() ==
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PeerConnectionInterface::kIceConnectionCompleted);
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}
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void PeerConnectionTestWrapper::WaitForAudio() {
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EXPECT_TRUE_WAIT(CheckForAudio(), kMaxWait);
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RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
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<< ": Got enough audio frames.";
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}
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bool PeerConnectionTestWrapper::CheckForAudio() {
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return (fake_audio_capture_module_->frames_received() >=
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kTestAudioFrameCount);
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}
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void PeerConnectionTestWrapper::WaitForVideo() {
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EXPECT_TRUE_WAIT(CheckForVideo(), kMaxWait);
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RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
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<< ": Got enough video frames.";
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}
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bool PeerConnectionTestWrapper::CheckForVideo() {
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if (!renderer_) {
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return false;
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}
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return (renderer_->num_rendered_frames() >= kTestVideoFrameCount);
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}
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void PeerConnectionTestWrapper::GetAndAddUserMedia(
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bool audio,
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const cricket::AudioOptions& audio_options,
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bool video) {
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rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
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GetUserMedia(audio, audio_options, video);
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for (const auto& audio_track : stream->GetAudioTracks()) {
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EXPECT_TRUE(peer_connection_->AddTrack(audio_track, {stream->id()}).ok());
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}
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for (const auto& video_track : stream->GetVideoTracks()) {
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EXPECT_TRUE(peer_connection_->AddTrack(video_track, {stream->id()}).ok());
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}
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}
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rtc::scoped_refptr<webrtc::MediaStreamInterface>
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PeerConnectionTestWrapper::GetUserMedia(
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bool audio,
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const cricket::AudioOptions& audio_options,
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bool video) {
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std::string stream_id =
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kStreamIdBase + rtc::ToString(num_get_user_media_calls_++);
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rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
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peer_connection_factory_->CreateLocalMediaStream(stream_id);
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if (audio) {
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cricket::AudioOptions options = audio_options;
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// Disable highpass filter so that we can get all the test audio frames.
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options.highpass_filter = false;
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rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
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peer_connection_factory_->CreateAudioSource(options);
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rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
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peer_connection_factory_->CreateAudioTrack(kAudioTrackLabelBase,
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source));
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stream->AddTrack(audio_track);
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}
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if (video) {
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// Set max frame rate to 10fps to reduce the risk of the tests to be flaky.
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webrtc::FakePeriodicVideoSource::Config config;
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config.frame_interval_ms = 100;
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config.timestamp_offset_ms = rtc::TimeMillis();
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rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source =
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new rtc::RefCountedObject<webrtc::FakePeriodicVideoTrackSource>(
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config, /* remote */ false);
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std::string videotrack_label = stream_id + kVideoTrackLabelBase;
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rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
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peer_connection_factory_->CreateVideoTrack(videotrack_label, source));
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stream->AddTrack(video_track);
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}
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return stream;
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}
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