blob: c3ffcb0893729fb721a62d35325b5d11be6e0154 [file] [log] [blame]
/*
* Copyright 2013 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 "pc/test/peer_connection_test_wrapper.h"
#include <stddef.h>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/types/optional.h"
#include "api/audio/audio_mixer.h"
#include "api/create_peerconnection_factory.h"
#include "api/sequence_checker.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "p2p/base/fake_port_allocator.h"
#include "p2p/base/port_allocator.h"
#include "pc/test/fake_periodic_video_source.h"
#include "pc/test/fake_periodic_video_track_source.h"
#include "pc/test/fake_rtc_certificate_generator.h"
#include "pc/test/mock_peer_connection_observers.h"
#include "rtc_base/gunit.h"
#include "rtc_base/logging.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/rtc_certificate_generator.h"
#include "rtc_base/string_encode.h"
#include "rtc_base/time_utils.h"
#include "test/gtest.h"
using webrtc::FakeVideoTrackRenderer;
using webrtc::IceCandidateInterface;
using webrtc::MediaStreamInterface;
using webrtc::MediaStreamTrackInterface;
using webrtc::MockSetSessionDescriptionObserver;
using webrtc::PeerConnectionInterface;
using webrtc::RtpReceiverInterface;
using webrtc::SdpType;
using webrtc::SessionDescriptionInterface;
using webrtc::VideoTrackInterface;
namespace {
const char kStreamIdBase[] = "stream_id";
const char kVideoTrackLabelBase[] = "video_track";
const char kAudioTrackLabelBase[] = "audio_track";
constexpr int kMaxWait = 10000;
constexpr int kTestAudioFrameCount = 3;
constexpr int kTestVideoFrameCount = 3;
} // namespace
void PeerConnectionTestWrapper::Connect(PeerConnectionTestWrapper* caller,
PeerConnectionTestWrapper* callee) {
caller->SignalOnIceCandidateReady.connect(
callee, &PeerConnectionTestWrapper::AddIceCandidate);
callee->SignalOnIceCandidateReady.connect(
caller, &PeerConnectionTestWrapper::AddIceCandidate);
caller->SignalOnSdpReady.connect(callee,
&PeerConnectionTestWrapper::ReceiveOfferSdp);
callee->SignalOnSdpReady.connect(
caller, &PeerConnectionTestWrapper::ReceiveAnswerSdp);
}
PeerConnectionTestWrapper::PeerConnectionTestWrapper(
const std::string& name,
rtc::Thread* network_thread,
rtc::Thread* worker_thread)
: name_(name),
network_thread_(network_thread),
worker_thread_(worker_thread),
pending_negotiation_(false) {
pc_thread_checker_.Detach();
}
PeerConnectionTestWrapper::~PeerConnectionTestWrapper() {
RTC_DCHECK_RUN_ON(&pc_thread_checker_);
// Either network_thread or worker_thread might be active at this point.
// Relying on ~PeerConnection to properly wait for them doesn't work,
// as a vptr race might occur (before we enter the destruction body).
// See: bugs.webrtc.org/9847
if (pc()) {
pc()->Close();
}
}
bool PeerConnectionTestWrapper::CreatePc(
const webrtc::PeerConnectionInterface::RTCConfiguration& config,
rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory,
rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory) {
std::unique_ptr<cricket::PortAllocator> port_allocator(
new cricket::FakePortAllocator(network_thread_, nullptr));
RTC_DCHECK_RUN_ON(&pc_thread_checker_);
fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
if (fake_audio_capture_module_ == NULL) {
return false;
}
peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
network_thread_, worker_thread_, rtc::Thread::Current(),
rtc::scoped_refptr<webrtc::AudioDeviceModule>(fake_audio_capture_module_),
audio_encoder_factory, audio_decoder_factory,
webrtc::CreateBuiltinVideoEncoderFactory(),
webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
nullptr /* audio_processing */);
if (!peer_connection_factory_) {
return false;
}
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator(
new FakeRTCCertificateGenerator());
peer_connection_ = peer_connection_factory_->CreatePeerConnection(
config, std::move(port_allocator), std::move(cert_generator), this);
return peer_connection_.get() != NULL;
}
rtc::scoped_refptr<webrtc::DataChannelInterface>
PeerConnectionTestWrapper::CreateDataChannel(
const std::string& label,
const webrtc::DataChannelInit& init) {
return peer_connection_->CreateDataChannel(label, &init);
}
void PeerConnectionTestWrapper::WaitForNegotiation() {
EXPECT_TRUE_WAIT(!pending_negotiation_, kMaxWait);
}
void PeerConnectionTestWrapper::OnSignalingChange(
webrtc::PeerConnectionInterface::SignalingState new_state) {
if (new_state == webrtc::PeerConnectionInterface::SignalingState::kStable) {
pending_negotiation_ = false;
}
}
void PeerConnectionTestWrapper::OnAddTrack(
rtc::scoped_refptr<RtpReceiverInterface> receiver,
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {
RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ << ": OnAddTrack";
if (receiver->track()->kind() == MediaStreamTrackInterface::kVideoKind) {
auto* video_track =
static_cast<VideoTrackInterface*>(receiver->track().get());
renderer_ = std::make_unique<FakeVideoTrackRenderer>(video_track);
}
}
void PeerConnectionTestWrapper::OnIceCandidate(
const IceCandidateInterface* candidate) {
std::string sdp;
EXPECT_TRUE(candidate->ToString(&sdp));
// Give the user a chance to modify sdp for testing.
SignalOnIceCandidateCreated(&sdp);
SignalOnIceCandidateReady(candidate->sdp_mid(), candidate->sdp_mline_index(),
sdp);
}
void PeerConnectionTestWrapper::OnDataChannel(
rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) {
SignalOnDataChannel(data_channel);
}
void PeerConnectionTestWrapper::OnSuccess(SessionDescriptionInterface* desc) {
// This callback should take the ownership of |desc|.
std::unique_ptr<SessionDescriptionInterface> owned_desc(desc);
std::string sdp;
EXPECT_TRUE(desc->ToString(&sdp));
RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ << ": "
<< webrtc::SdpTypeToString(desc->GetType())
<< " sdp created: " << sdp;
// Give the user a chance to modify sdp for testing.
SignalOnSdpCreated(&sdp);
SetLocalDescription(desc->GetType(), sdp);
SignalOnSdpReady(sdp);
}
void PeerConnectionTestWrapper::CreateOffer(
const webrtc::PeerConnectionInterface::RTCOfferAnswerOptions& options) {
RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ << ": CreateOffer.";
pending_negotiation_ = true;
peer_connection_->CreateOffer(this, options);
}
void PeerConnectionTestWrapper::CreateAnswer(
const webrtc::PeerConnectionInterface::RTCOfferAnswerOptions& options) {
RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
<< ": CreateAnswer.";
pending_negotiation_ = true;
peer_connection_->CreateAnswer(this, options);
}
void PeerConnectionTestWrapper::ReceiveOfferSdp(const std::string& sdp) {
SetRemoteDescription(SdpType::kOffer, sdp);
CreateAnswer(webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());
}
void PeerConnectionTestWrapper::ReceiveAnswerSdp(const std::string& sdp) {
SetRemoteDescription(SdpType::kAnswer, sdp);
}
void PeerConnectionTestWrapper::SetLocalDescription(SdpType type,
const std::string& sdp) {
RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
<< ": SetLocalDescription " << webrtc::SdpTypeToString(type)
<< " " << sdp;
rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
peer_connection_->SetLocalDescription(
observer, webrtc::CreateSessionDescription(type, sdp).release());
}
void PeerConnectionTestWrapper::SetRemoteDescription(SdpType type,
const std::string& sdp) {
RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
<< ": SetRemoteDescription " << webrtc::SdpTypeToString(type)
<< " " << sdp;
rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
peer_connection_->SetRemoteDescription(
observer, webrtc::CreateSessionDescription(type, sdp).release());
}
void PeerConnectionTestWrapper::AddIceCandidate(const std::string& sdp_mid,
int sdp_mline_index,
const std::string& candidate) {
std::unique_ptr<webrtc::IceCandidateInterface> owned_candidate(
webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, candidate, NULL));
EXPECT_TRUE(peer_connection_->AddIceCandidate(owned_candidate.get()));
}
void PeerConnectionTestWrapper::WaitForCallEstablished() {
WaitForConnection();
WaitForAudio();
WaitForVideo();
}
void PeerConnectionTestWrapper::WaitForConnection() {
EXPECT_TRUE_WAIT(CheckForConnection(), kMaxWait);
RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ << ": Connected.";
}
bool PeerConnectionTestWrapper::CheckForConnection() {
return (peer_connection_->ice_connection_state() ==
PeerConnectionInterface::kIceConnectionConnected) ||
(peer_connection_->ice_connection_state() ==
PeerConnectionInterface::kIceConnectionCompleted);
}
void PeerConnectionTestWrapper::WaitForAudio() {
EXPECT_TRUE_WAIT(CheckForAudio(), kMaxWait);
RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
<< ": Got enough audio frames.";
}
bool PeerConnectionTestWrapper::CheckForAudio() {
return (fake_audio_capture_module_->frames_received() >=
kTestAudioFrameCount);
}
void PeerConnectionTestWrapper::WaitForVideo() {
EXPECT_TRUE_WAIT(CheckForVideo(), kMaxWait);
RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_
<< ": Got enough video frames.";
}
bool PeerConnectionTestWrapper::CheckForVideo() {
if (!renderer_) {
return false;
}
return (renderer_->num_rendered_frames() >= kTestVideoFrameCount);
}
void PeerConnectionTestWrapper::GetAndAddUserMedia(
bool audio,
const cricket::AudioOptions& audio_options,
bool video) {
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
GetUserMedia(audio, audio_options, video);
for (const auto& audio_track : stream->GetAudioTracks()) {
EXPECT_TRUE(peer_connection_->AddTrack(audio_track, {stream->id()}).ok());
}
for (const auto& video_track : stream->GetVideoTracks()) {
EXPECT_TRUE(peer_connection_->AddTrack(video_track, {stream->id()}).ok());
}
}
rtc::scoped_refptr<webrtc::MediaStreamInterface>
PeerConnectionTestWrapper::GetUserMedia(
bool audio,
const cricket::AudioOptions& audio_options,
bool video) {
std::string stream_id =
kStreamIdBase + rtc::ToString(num_get_user_media_calls_++);
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
peer_connection_factory_->CreateLocalMediaStream(stream_id);
if (audio) {
cricket::AudioOptions options = audio_options;
// Disable highpass filter so that we can get all the test audio frames.
options.highpass_filter = false;
rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
peer_connection_factory_->CreateAudioSource(options);
rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
peer_connection_factory_->CreateAudioTrack(kAudioTrackLabelBase,
source));
stream->AddTrack(audio_track);
}
if (video) {
// Set max frame rate to 10fps to reduce the risk of the tests to be flaky.
webrtc::FakePeriodicVideoSource::Config config;
config.frame_interval_ms = 100;
config.timestamp_offset_ms = rtc::TimeMillis();
rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source =
new rtc::RefCountedObject<webrtc::FakePeriodicVideoTrackSource>(
config, /* remote */ false);
std::string videotrack_label = stream_id + kVideoTrackLabelBase;
rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
peer_connection_factory_->CreateVideoTrack(videotrack_label, source));
stream->AddTrack(video_track);
}
return stream;
}