| Index: webrtc/api/peerconnection_unittest.cc
|
| diff --git a/webrtc/api/peerconnection_unittest.cc b/webrtc/api/peerconnection_unittest.cc
|
| deleted file mode 100644
|
| index 301fed55dbce0659d4fcd45eae8c5ad48d2308ff..0000000000000000000000000000000000000000
|
| --- a/webrtc/api/peerconnection_unittest.cc
|
| +++ /dev/null
|
| @@ -1,2803 +0,0 @@
|
| -/*
|
| - * Copyright 2012 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 <stdio.h>
|
| -
|
| -#include <algorithm>
|
| -#include <list>
|
| -#include <map>
|
| -#include <memory>
|
| -#include <utility>
|
| -#include <vector>
|
| -
|
| -#include "webrtc/api/dtmfsender.h"
|
| -#include "webrtc/api/fakemetricsobserver.h"
|
| -#include "webrtc/api/localaudiosource.h"
|
| -#include "webrtc/api/mediastreaminterface.h"
|
| -#include "webrtc/api/peerconnection.h"
|
| -#include "webrtc/api/peerconnectionfactory.h"
|
| -#include "webrtc/api/peerconnectioninterface.h"
|
| -#include "webrtc/api/test/fakeaudiocapturemodule.h"
|
| -#include "webrtc/api/test/fakeconstraints.h"
|
| -#include "webrtc/api/test/fakeperiodicvideocapturer.h"
|
| -#include "webrtc/api/test/fakertccertificategenerator.h"
|
| -#include "webrtc/api/test/fakevideotrackrenderer.h"
|
| -#include "webrtc/api/test/mockpeerconnectionobservers.h"
|
| -#include "webrtc/base/fakenetwork.h"
|
| -#include "webrtc/base/gunit.h"
|
| -#include "webrtc/base/helpers.h"
|
| -#include "webrtc/base/physicalsocketserver.h"
|
| -#include "webrtc/base/ssladapter.h"
|
| -#include "webrtc/base/sslstreamadapter.h"
|
| -#include "webrtc/base/thread.h"
|
| -#include "webrtc/base/virtualsocketserver.h"
|
| -#include "webrtc/media/engine/fakewebrtcvideoengine.h"
|
| -#include "webrtc/p2p/base/p2pconstants.h"
|
| -#include "webrtc/p2p/base/sessiondescription.h"
|
| -#include "webrtc/p2p/base/testturnserver.h"
|
| -#include "webrtc/p2p/client/basicportallocator.h"
|
| -#include "webrtc/pc/mediasession.h"
|
| -
|
| -#define MAYBE_SKIP_TEST(feature) \
|
| - if (!(feature())) { \
|
| - LOG(LS_INFO) << "Feature disabled... skipping"; \
|
| - return; \
|
| - }
|
| -
|
| -using cricket::ContentInfo;
|
| -using cricket::FakeWebRtcVideoDecoder;
|
| -using cricket::FakeWebRtcVideoDecoderFactory;
|
| -using cricket::FakeWebRtcVideoEncoder;
|
| -using cricket::FakeWebRtcVideoEncoderFactory;
|
| -using cricket::MediaContentDescription;
|
| -using webrtc::DataBuffer;
|
| -using webrtc::DataChannelInterface;
|
| -using webrtc::DtmfSender;
|
| -using webrtc::DtmfSenderInterface;
|
| -using webrtc::DtmfSenderObserverInterface;
|
| -using webrtc::FakeConstraints;
|
| -using webrtc::MediaConstraintsInterface;
|
| -using webrtc::MediaStreamInterface;
|
| -using webrtc::MediaStreamTrackInterface;
|
| -using webrtc::MockCreateSessionDescriptionObserver;
|
| -using webrtc::MockDataChannelObserver;
|
| -using webrtc::MockSetSessionDescriptionObserver;
|
| -using webrtc::MockStatsObserver;
|
| -using webrtc::ObserverInterface;
|
| -using webrtc::PeerConnectionInterface;
|
| -using webrtc::PeerConnectionFactory;
|
| -using webrtc::SessionDescriptionInterface;
|
| -using webrtc::StreamCollectionInterface;
|
| -
|
| -namespace {
|
| -
|
| -static const int kMaxWaitMs = 10000;
|
| -// Disable for TSan v2, see
|
| -// https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
|
| -// This declaration is also #ifdef'd as it causes uninitialized-variable
|
| -// warnings.
|
| -#if !defined(THREAD_SANITIZER)
|
| -static const int kMaxWaitForStatsMs = 3000;
|
| -#endif
|
| -static const int kMaxWaitForActivationMs = 5000;
|
| -static const int kMaxWaitForFramesMs = 10000;
|
| -static const int kEndAudioFrameCount = 3;
|
| -static const int kEndVideoFrameCount = 3;
|
| -
|
| -static const char kStreamLabelBase[] = "stream_label";
|
| -static const char kVideoTrackLabelBase[] = "video_track";
|
| -static const char kAudioTrackLabelBase[] = "audio_track";
|
| -static const char kDataChannelLabel[] = "data_channel";
|
| -
|
| -// Disable for TSan v2, see
|
| -// https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
|
| -// This declaration is also #ifdef'd as it causes unused-variable errors.
|
| -#if !defined(THREAD_SANITIZER)
|
| -// SRTP cipher name negotiated by the tests. This must be updated if the
|
| -// default changes.
|
| -static const int kDefaultSrtpCryptoSuite = rtc::SRTP_AES128_CM_SHA1_32;
|
| -static const int kDefaultSrtpCryptoSuiteGcm = rtc::SRTP_AEAD_AES_256_GCM;
|
| -#endif
|
| -
|
| -// Used to simulate signaling ICE/SDP between two PeerConnections.
|
| -enum Message { MSG_SDP_MESSAGE, MSG_ICE_MESSAGE };
|
| -
|
| -struct SdpMessage {
|
| - std::string type;
|
| - std::string msg;
|
| -};
|
| -
|
| -struct IceMessage {
|
| - std::string sdp_mid;
|
| - int sdp_mline_index;
|
| - std::string msg;
|
| -};
|
| -
|
| -static void RemoveLinesFromSdp(const std::string& line_start,
|
| - std::string* sdp) {
|
| - const char kSdpLineEnd[] = "\r\n";
|
| - size_t ssrc_pos = 0;
|
| - while ((ssrc_pos = sdp->find(line_start, ssrc_pos)) !=
|
| - std::string::npos) {
|
| - size_t end_ssrc = sdp->find(kSdpLineEnd, ssrc_pos);
|
| - sdp->erase(ssrc_pos, end_ssrc - ssrc_pos + strlen(kSdpLineEnd));
|
| - }
|
| -}
|
| -
|
| -bool StreamsHaveAudioTrack(StreamCollectionInterface* streams) {
|
| - for (size_t idx = 0; idx < streams->count(); idx++) {
|
| - auto stream = streams->at(idx);
|
| - if (stream->GetAudioTracks().size() > 0) {
|
| - return true;
|
| - }
|
| - }
|
| - return false;
|
| -}
|
| -
|
| -bool StreamsHaveVideoTrack(StreamCollectionInterface* streams) {
|
| - for (size_t idx = 0; idx < streams->count(); idx++) {
|
| - auto stream = streams->at(idx);
|
| - if (stream->GetVideoTracks().size() > 0) {
|
| - return true;
|
| - }
|
| - }
|
| - return false;
|
| -}
|
| -
|
| -class SignalingMessageReceiver {
|
| - public:
|
| - virtual void ReceiveSdpMessage(const std::string& type,
|
| - std::string& msg) = 0;
|
| - virtual void ReceiveIceMessage(const std::string& sdp_mid,
|
| - int sdp_mline_index,
|
| - const std::string& msg) = 0;
|
| -
|
| - protected:
|
| - SignalingMessageReceiver() {}
|
| - virtual ~SignalingMessageReceiver() {}
|
| -};
|
| -
|
| -class MockRtpReceiverObserver : public webrtc::RtpReceiverObserverInterface {
|
| - public:
|
| - MockRtpReceiverObserver(cricket::MediaType media_type)
|
| - : expected_media_type_(media_type) {}
|
| -
|
| - void OnFirstPacketReceived(cricket::MediaType media_type) override {
|
| - ASSERT_EQ(expected_media_type_, media_type);
|
| - first_packet_received_ = true;
|
| - }
|
| -
|
| - bool first_packet_received() { return first_packet_received_; }
|
| -
|
| - virtual ~MockRtpReceiverObserver() {}
|
| -
|
| - private:
|
| - bool first_packet_received_ = false;
|
| - cricket::MediaType expected_media_type_;
|
| -};
|
| -
|
| -class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
|
| - public SignalingMessageReceiver,
|
| - public ObserverInterface,
|
| - public rtc::MessageHandler {
|
| - public:
|
| - // We need these using declarations because there are two versions of each of
|
| - // the below methods and we only override one of them.
|
| - // TODO(deadbeef): Remove once there's only one version of the methods.
|
| - using PeerConnectionObserver::OnAddStream;
|
| - using PeerConnectionObserver::OnRemoveStream;
|
| - using PeerConnectionObserver::OnDataChannel;
|
| -
|
| - // If |config| is not provided, uses a default constructed RTCConfiguration.
|
| - static PeerConnectionTestClient* CreateClientWithDtlsIdentityStore(
|
| - const std::string& id,
|
| - const MediaConstraintsInterface* constraints,
|
| - const PeerConnectionFactory::Options* options,
|
| - const PeerConnectionInterface::RTCConfiguration* config,
|
| - std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
|
| - bool prefer_constraint_apis,
|
| - rtc::Thread* network_thread,
|
| - rtc::Thread* worker_thread) {
|
| - PeerConnectionTestClient* client(new PeerConnectionTestClient(id));
|
| - if (!client->Init(constraints, options, config, std::move(cert_generator),
|
| - prefer_constraint_apis, network_thread, worker_thread)) {
|
| - delete client;
|
| - return nullptr;
|
| - }
|
| - return client;
|
| - }
|
| -
|
| - static PeerConnectionTestClient* CreateClient(
|
| - const std::string& id,
|
| - const MediaConstraintsInterface* constraints,
|
| - const PeerConnectionFactory::Options* options,
|
| - const PeerConnectionInterface::RTCConfiguration* config,
|
| - rtc::Thread* network_thread,
|
| - rtc::Thread* worker_thread) {
|
| - std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
|
| - rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
|
| - new FakeRTCCertificateGenerator() : nullptr);
|
| -
|
| - return CreateClientWithDtlsIdentityStore(id, constraints, options, config,
|
| - std::move(cert_generator), true,
|
| - network_thread, worker_thread);
|
| - }
|
| -
|
| - static PeerConnectionTestClient* CreateClientPreferNoConstraints(
|
| - const std::string& id,
|
| - const PeerConnectionFactory::Options* options,
|
| - rtc::Thread* network_thread,
|
| - rtc::Thread* worker_thread) {
|
| - std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
|
| - rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
|
| - new FakeRTCCertificateGenerator() : nullptr);
|
| -
|
| - return CreateClientWithDtlsIdentityStore(id, nullptr, options, nullptr,
|
| - std::move(cert_generator), false,
|
| - network_thread, worker_thread);
|
| - }
|
| -
|
| - ~PeerConnectionTestClient() {
|
| - }
|
| -
|
| - void Negotiate() { Negotiate(true, true); }
|
| -
|
| - void Negotiate(bool audio, bool video) {
|
| - std::unique_ptr<SessionDescriptionInterface> offer;
|
| - ASSERT_TRUE(DoCreateOffer(&offer));
|
| -
|
| - if (offer->description()->GetContentByName("audio")) {
|
| - offer->description()->GetContentByName("audio")->rejected = !audio;
|
| - }
|
| - if (offer->description()->GetContentByName("video")) {
|
| - offer->description()->GetContentByName("video")->rejected = !video;
|
| - }
|
| -
|
| - std::string sdp;
|
| - EXPECT_TRUE(offer->ToString(&sdp));
|
| - EXPECT_TRUE(DoSetLocalDescription(offer.release()));
|
| - SendSdpMessage(webrtc::SessionDescriptionInterface::kOffer, sdp);
|
| - }
|
| -
|
| - void SendSdpMessage(const std::string& type, std::string& msg) {
|
| - if (signaling_delay_ms_ == 0) {
|
| - if (signaling_message_receiver_) {
|
| - signaling_message_receiver_->ReceiveSdpMessage(type, msg);
|
| - }
|
| - } else {
|
| - rtc::Thread::Current()->PostDelayed(
|
| - RTC_FROM_HERE, signaling_delay_ms_, this, MSG_SDP_MESSAGE,
|
| - new rtc::TypedMessageData<SdpMessage>({type, msg}));
|
| - }
|
| - }
|
| -
|
| - void SendIceMessage(const std::string& sdp_mid,
|
| - int sdp_mline_index,
|
| - const std::string& msg) {
|
| - if (signaling_delay_ms_ == 0) {
|
| - if (signaling_message_receiver_) {
|
| - signaling_message_receiver_->ReceiveIceMessage(sdp_mid, sdp_mline_index,
|
| - msg);
|
| - }
|
| - } else {
|
| - rtc::Thread::Current()->PostDelayed(RTC_FROM_HERE, signaling_delay_ms_,
|
| - this, MSG_ICE_MESSAGE,
|
| - new rtc::TypedMessageData<IceMessage>(
|
| - {sdp_mid, sdp_mline_index, msg}));
|
| - }
|
| - }
|
| -
|
| - // MessageHandler callback.
|
| - void OnMessage(rtc::Message* msg) override {
|
| - switch (msg->message_id) {
|
| - case MSG_SDP_MESSAGE: {
|
| - auto sdp_message =
|
| - static_cast<rtc::TypedMessageData<SdpMessage>*>(msg->pdata);
|
| - if (signaling_message_receiver_) {
|
| - signaling_message_receiver_->ReceiveSdpMessage(
|
| - sdp_message->data().type, sdp_message->data().msg);
|
| - }
|
| - delete sdp_message;
|
| - break;
|
| - }
|
| - case MSG_ICE_MESSAGE: {
|
| - auto ice_message =
|
| - static_cast<rtc::TypedMessageData<IceMessage>*>(msg->pdata);
|
| - if (signaling_message_receiver_) {
|
| - signaling_message_receiver_->ReceiveIceMessage(
|
| - ice_message->data().sdp_mid, ice_message->data().sdp_mline_index,
|
| - ice_message->data().msg);
|
| - }
|
| - delete ice_message;
|
| - break;
|
| - }
|
| - default:
|
| - RTC_CHECK(false);
|
| - }
|
| - }
|
| -
|
| - // SignalingMessageReceiver callback.
|
| - void ReceiveSdpMessage(const std::string& type, std::string& msg) override {
|
| - FilterIncomingSdpMessage(&msg);
|
| - if (type == webrtc::SessionDescriptionInterface::kOffer) {
|
| - HandleIncomingOffer(msg);
|
| - } else {
|
| - HandleIncomingAnswer(msg);
|
| - }
|
| - }
|
| -
|
| - // SignalingMessageReceiver callback.
|
| - void ReceiveIceMessage(const std::string& sdp_mid,
|
| - int sdp_mline_index,
|
| - const std::string& msg) override {
|
| - LOG(INFO) << id_ << "ReceiveIceMessage";
|
| - std::unique_ptr<webrtc::IceCandidateInterface> candidate(
|
| - webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, msg, nullptr));
|
| - EXPECT_TRUE(pc()->AddIceCandidate(candidate.get()));
|
| - }
|
| -
|
| - // PeerConnectionObserver callbacks.
|
| - void OnSignalingChange(
|
| - webrtc::PeerConnectionInterface::SignalingState new_state) override {
|
| - EXPECT_EQ(pc()->signaling_state(), new_state);
|
| - }
|
| - void OnAddStream(
|
| - rtc::scoped_refptr<MediaStreamInterface> media_stream) override {
|
| - media_stream->RegisterObserver(this);
|
| - for (size_t i = 0; i < media_stream->GetVideoTracks().size(); ++i) {
|
| - const std::string id = media_stream->GetVideoTracks()[i]->id();
|
| - ASSERT_TRUE(fake_video_renderers_.find(id) ==
|
| - fake_video_renderers_.end());
|
| - fake_video_renderers_[id].reset(new webrtc::FakeVideoTrackRenderer(
|
| - media_stream->GetVideoTracks()[i]));
|
| - }
|
| - }
|
| - void OnRemoveStream(
|
| - rtc::scoped_refptr<MediaStreamInterface> media_stream) override {}
|
| - void OnRenegotiationNeeded() override {}
|
| - void OnIceConnectionChange(
|
| - webrtc::PeerConnectionInterface::IceConnectionState new_state) override {
|
| - EXPECT_EQ(pc()->ice_connection_state(), new_state);
|
| - }
|
| - void OnIceGatheringChange(
|
| - webrtc::PeerConnectionInterface::IceGatheringState new_state) override {
|
| - EXPECT_EQ(pc()->ice_gathering_state(), new_state);
|
| - }
|
| - void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override {
|
| - LOG(INFO) << id_ << "OnIceCandidate";
|
| -
|
| - std::string ice_sdp;
|
| - EXPECT_TRUE(candidate->ToString(&ice_sdp));
|
| - if (signaling_message_receiver_ == nullptr) {
|
| - // Remote party may be deleted.
|
| - return;
|
| - }
|
| - SendIceMessage(candidate->sdp_mid(), candidate->sdp_mline_index(), ice_sdp);
|
| - }
|
| -
|
| - // MediaStreamInterface callback
|
| - void OnChanged() override {
|
| - // Track added or removed from MediaStream, so update our renderers.
|
| - rtc::scoped_refptr<StreamCollectionInterface> remote_streams =
|
| - pc()->remote_streams();
|
| - // Remove renderers for tracks that were removed.
|
| - for (auto it = fake_video_renderers_.begin();
|
| - it != fake_video_renderers_.end();) {
|
| - if (remote_streams->FindVideoTrack(it->first) == nullptr) {
|
| - auto to_remove = it++;
|
| - removed_fake_video_renderers_.push_back(std::move(to_remove->second));
|
| - fake_video_renderers_.erase(to_remove);
|
| - } else {
|
| - ++it;
|
| - }
|
| - }
|
| - // Create renderers for new video tracks.
|
| - for (size_t stream_index = 0; stream_index < remote_streams->count();
|
| - ++stream_index) {
|
| - MediaStreamInterface* remote_stream = remote_streams->at(stream_index);
|
| - for (size_t track_index = 0;
|
| - track_index < remote_stream->GetVideoTracks().size();
|
| - ++track_index) {
|
| - const std::string id =
|
| - remote_stream->GetVideoTracks()[track_index]->id();
|
| - if (fake_video_renderers_.find(id) != fake_video_renderers_.end()) {
|
| - continue;
|
| - }
|
| - fake_video_renderers_[id].reset(new webrtc::FakeVideoTrackRenderer(
|
| - remote_stream->GetVideoTracks()[track_index]));
|
| - }
|
| - }
|
| - }
|
| -
|
| - void SetVideoConstraints(const webrtc::FakeConstraints& video_constraint) {
|
| - video_constraints_ = video_constraint;
|
| - }
|
| -
|
| - void AddMediaStream(bool audio, bool video) {
|
| - std::string stream_label =
|
| - kStreamLabelBase +
|
| - rtc::ToString<int>(static_cast<int>(pc()->local_streams()->count()));
|
| - rtc::scoped_refptr<MediaStreamInterface> stream =
|
| - peer_connection_factory_->CreateLocalMediaStream(stream_label);
|
| -
|
| - if (audio && can_receive_audio()) {
|
| - stream->AddTrack(CreateLocalAudioTrack(stream_label));
|
| - }
|
| - if (video && can_receive_video()) {
|
| - stream->AddTrack(CreateLocalVideoTrack(stream_label));
|
| - }
|
| -
|
| - EXPECT_TRUE(pc()->AddStream(stream));
|
| - }
|
| -
|
| - size_t NumberOfLocalMediaStreams() { return pc()->local_streams()->count(); }
|
| -
|
| - bool SessionActive() {
|
| - return pc()->signaling_state() == webrtc::PeerConnectionInterface::kStable;
|
| - }
|
| -
|
| - // Automatically add a stream when receiving an offer, if we don't have one.
|
| - // Defaults to true.
|
| - void set_auto_add_stream(bool auto_add_stream) {
|
| - auto_add_stream_ = auto_add_stream;
|
| - }
|
| -
|
| - void set_signaling_message_receiver(
|
| - SignalingMessageReceiver* signaling_message_receiver) {
|
| - signaling_message_receiver_ = signaling_message_receiver;
|
| - }
|
| -
|
| - void set_signaling_delay_ms(int delay_ms) { signaling_delay_ms_ = delay_ms; }
|
| -
|
| - void EnableVideoDecoderFactory() {
|
| - video_decoder_factory_enabled_ = true;
|
| - fake_video_decoder_factory_->AddSupportedVideoCodecType(
|
| - webrtc::kVideoCodecVP8);
|
| - }
|
| -
|
| - void IceRestart() {
|
| - offer_answer_constraints_.SetMandatoryIceRestart(true);
|
| - offer_answer_options_.ice_restart = true;
|
| - SetExpectIceRestart(true);
|
| - }
|
| -
|
| - void SetExpectIceRestart(bool expect_restart) {
|
| - expect_ice_restart_ = expect_restart;
|
| - }
|
| -
|
| - bool ExpectIceRestart() const { return expect_ice_restart_; }
|
| -
|
| - void SetExpectIceRenomination(bool expect_renomination) {
|
| - expect_ice_renomination_ = expect_renomination;
|
| - }
|
| - void SetExpectRemoteIceRenomination(bool expect_renomination) {
|
| - expect_remote_ice_renomination_ = expect_renomination;
|
| - }
|
| - bool ExpectIceRenomination() { return expect_ice_renomination_; }
|
| - bool ExpectRemoteIceRenomination() { return expect_remote_ice_renomination_; }
|
| -
|
| - // The below 3 methods assume streams will be offered.
|
| - // Thus they'll only set the "offer to receive" flag to true if it's
|
| - // currently false, not if it's just unset.
|
| - void SetReceiveAudioVideo(bool audio, bool video) {
|
| - SetReceiveAudio(audio);
|
| - SetReceiveVideo(video);
|
| - ASSERT_EQ(audio, can_receive_audio());
|
| - ASSERT_EQ(video, can_receive_video());
|
| - }
|
| -
|
| - void SetReceiveAudio(bool audio) {
|
| - if (audio && can_receive_audio()) {
|
| - return;
|
| - }
|
| - offer_answer_constraints_.SetMandatoryReceiveAudio(audio);
|
| - offer_answer_options_.offer_to_receive_audio = audio ? 1 : 0;
|
| - }
|
| -
|
| - void SetReceiveVideo(bool video) {
|
| - if (video && can_receive_video()) {
|
| - return;
|
| - }
|
| - offer_answer_constraints_.SetMandatoryReceiveVideo(video);
|
| - offer_answer_options_.offer_to_receive_video = video ? 1 : 0;
|
| - }
|
| -
|
| - void SetOfferToReceiveAudioVideo(bool audio, bool video) {
|
| - offer_answer_constraints_.SetMandatoryReceiveAudio(audio);
|
| - offer_answer_options_.offer_to_receive_audio = audio ? 1 : 0;
|
| - offer_answer_constraints_.SetMandatoryReceiveVideo(video);
|
| - offer_answer_options_.offer_to_receive_video = video ? 1 : 0;
|
| - }
|
| -
|
| - void RemoveMsidFromReceivedSdp(bool remove) { remove_msid_ = remove; }
|
| -
|
| - void RemoveSdesCryptoFromReceivedSdp(bool remove) { remove_sdes_ = remove; }
|
| -
|
| - void RemoveBundleFromReceivedSdp(bool remove) { remove_bundle_ = remove; }
|
| -
|
| - void RemoveCvoFromReceivedSdp(bool remove) { remove_cvo_ = remove; }
|
| -
|
| - bool can_receive_audio() {
|
| - bool value;
|
| - if (prefer_constraint_apis_) {
|
| - if (webrtc::FindConstraint(
|
| - &offer_answer_constraints_,
|
| - MediaConstraintsInterface::kOfferToReceiveAudio, &value,
|
| - nullptr)) {
|
| - return value;
|
| - }
|
| - return true;
|
| - }
|
| - return offer_answer_options_.offer_to_receive_audio > 0 ||
|
| - offer_answer_options_.offer_to_receive_audio ==
|
| - PeerConnectionInterface::RTCOfferAnswerOptions::kUndefined;
|
| - }
|
| -
|
| - bool can_receive_video() {
|
| - bool value;
|
| - if (prefer_constraint_apis_) {
|
| - if (webrtc::FindConstraint(
|
| - &offer_answer_constraints_,
|
| - MediaConstraintsInterface::kOfferToReceiveVideo, &value,
|
| - nullptr)) {
|
| - return value;
|
| - }
|
| - return true;
|
| - }
|
| - return offer_answer_options_.offer_to_receive_video > 0 ||
|
| - offer_answer_options_.offer_to_receive_video ==
|
| - PeerConnectionInterface::RTCOfferAnswerOptions::kUndefined;
|
| - }
|
| -
|
| - void OnDataChannel(
|
| - rtc::scoped_refptr<DataChannelInterface> data_channel) override {
|
| - LOG(INFO) << id_ << "OnDataChannel";
|
| - data_channel_ = data_channel;
|
| - data_observer_.reset(new MockDataChannelObserver(data_channel));
|
| - }
|
| -
|
| - void CreateDataChannel() { CreateDataChannel(nullptr); }
|
| -
|
| - void CreateDataChannel(const webrtc::DataChannelInit* init) {
|
| - data_channel_ = pc()->CreateDataChannel(kDataChannelLabel, init);
|
| - ASSERT_TRUE(data_channel_.get() != nullptr);
|
| - data_observer_.reset(new MockDataChannelObserver(data_channel_));
|
| - }
|
| -
|
| - rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack(
|
| - const std::string& stream_label) {
|
| - FakeConstraints constraints;
|
| - // Disable highpass filter so that we can get all the test audio frames.
|
| - constraints.AddMandatory(MediaConstraintsInterface::kHighpassFilter, false);
|
| - rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
|
| - peer_connection_factory_->CreateAudioSource(&constraints);
|
| - // TODO(perkj): Test audio source when it is implemented. Currently audio
|
| - // always use the default input.
|
| - std::string label = stream_label + kAudioTrackLabelBase;
|
| - return peer_connection_factory_->CreateAudioTrack(label, source);
|
| - }
|
| -
|
| - rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack(
|
| - const std::string& stream_label) {
|
| - // Set max frame rate to 10fps to reduce the risk of the tests to be flaky.
|
| - FakeConstraints source_constraints = video_constraints_;
|
| - source_constraints.SetMandatoryMaxFrameRate(10);
|
| -
|
| - cricket::FakeVideoCapturer* fake_capturer =
|
| - new webrtc::FakePeriodicVideoCapturer();
|
| - fake_capturer->SetRotation(capture_rotation_);
|
| - video_capturers_.push_back(fake_capturer);
|
| - rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source =
|
| - peer_connection_factory_->CreateVideoSource(fake_capturer,
|
| - &source_constraints);
|
| - std::string label = stream_label + kVideoTrackLabelBase;
|
| -
|
| - rtc::scoped_refptr<webrtc::VideoTrackInterface> track(
|
| - peer_connection_factory_->CreateVideoTrack(label, source));
|
| - if (!local_video_renderer_) {
|
| - local_video_renderer_.reset(new webrtc::FakeVideoTrackRenderer(track));
|
| - }
|
| - return track;
|
| - }
|
| -
|
| - DataChannelInterface* data_channel() { return data_channel_; }
|
| - const MockDataChannelObserver* data_observer() const {
|
| - return data_observer_.get();
|
| - }
|
| -
|
| - webrtc::PeerConnectionInterface* pc() const { return peer_connection_.get(); }
|
| -
|
| - void StopVideoCapturers() {
|
| - for (auto* capturer : video_capturers_) {
|
| - capturer->Stop();
|
| - }
|
| - }
|
| -
|
| - void SetCaptureRotation(webrtc::VideoRotation rotation) {
|
| - ASSERT_TRUE(video_capturers_.empty());
|
| - capture_rotation_ = rotation;
|
| - }
|
| -
|
| - bool AudioFramesReceivedCheck(int number_of_frames) const {
|
| - return number_of_frames <= fake_audio_capture_module_->frames_received();
|
| - }
|
| -
|
| - int audio_frames_received() const {
|
| - return fake_audio_capture_module_->frames_received();
|
| - }
|
| -
|
| - bool VideoFramesReceivedCheck(int number_of_frames) {
|
| - if (video_decoder_factory_enabled_) {
|
| - const std::vector<FakeWebRtcVideoDecoder*>& decoders
|
| - = fake_video_decoder_factory_->decoders();
|
| - if (decoders.empty()) {
|
| - return number_of_frames <= 0;
|
| - }
|
| - // Note - this checks that EACH decoder has the requisite number
|
| - // of frames. The video_frames_received() function sums them.
|
| - for (FakeWebRtcVideoDecoder* decoder : decoders) {
|
| - if (number_of_frames > decoder->GetNumFramesReceived()) {
|
| - return false;
|
| - }
|
| - }
|
| - return true;
|
| - } else {
|
| - if (fake_video_renderers_.empty()) {
|
| - return number_of_frames <= 0;
|
| - }
|
| -
|
| - for (const auto& pair : fake_video_renderers_) {
|
| - if (number_of_frames > pair.second->num_rendered_frames()) {
|
| - return false;
|
| - }
|
| - }
|
| - return true;
|
| - }
|
| - }
|
| -
|
| - int video_frames_received() const {
|
| - int total = 0;
|
| - if (video_decoder_factory_enabled_) {
|
| - const std::vector<FakeWebRtcVideoDecoder*>& decoders =
|
| - fake_video_decoder_factory_->decoders();
|
| - for (const FakeWebRtcVideoDecoder* decoder : decoders) {
|
| - total += decoder->GetNumFramesReceived();
|
| - }
|
| - } else {
|
| - for (const auto& pair : fake_video_renderers_) {
|
| - total += pair.second->num_rendered_frames();
|
| - }
|
| - for (const auto& renderer : removed_fake_video_renderers_) {
|
| - total += renderer->num_rendered_frames();
|
| - }
|
| - }
|
| - return total;
|
| - }
|
| -
|
| - // Verify the CreateDtmfSender interface
|
| - void VerifyDtmf() {
|
| - std::unique_ptr<DummyDtmfObserver> observer(new DummyDtmfObserver());
|
| - rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender;
|
| -
|
| - // We can't create a DTMF sender with an invalid audio track or a non local
|
| - // track.
|
| - EXPECT_TRUE(peer_connection_->CreateDtmfSender(nullptr) == nullptr);
|
| - rtc::scoped_refptr<webrtc::AudioTrackInterface> non_localtrack(
|
| - peer_connection_factory_->CreateAudioTrack("dummy_track", nullptr));
|
| - EXPECT_TRUE(peer_connection_->CreateDtmfSender(non_localtrack) == nullptr);
|
| -
|
| - // We should be able to create a DTMF sender from a local track.
|
| - webrtc::AudioTrackInterface* localtrack =
|
| - peer_connection_->local_streams()->at(0)->GetAudioTracks()[0];
|
| - dtmf_sender = peer_connection_->CreateDtmfSender(localtrack);
|
| - EXPECT_TRUE(dtmf_sender.get() != nullptr);
|
| - dtmf_sender->RegisterObserver(observer.get());
|
| -
|
| - // Test the DtmfSender object just created.
|
| - EXPECT_TRUE(dtmf_sender->CanInsertDtmf());
|
| - EXPECT_TRUE(dtmf_sender->InsertDtmf("1a", 100, 50));
|
| -
|
| - // We don't need to verify that the DTMF tones are actually sent out because
|
| - // that is already covered by the tests of the lower level components.
|
| -
|
| - EXPECT_TRUE_WAIT(observer->completed(), kMaxWaitMs);
|
| - std::vector<std::string> tones;
|
| - tones.push_back("1");
|
| - tones.push_back("a");
|
| - tones.push_back("");
|
| - observer->Verify(tones);
|
| -
|
| - dtmf_sender->UnregisterObserver();
|
| - }
|
| -
|
| - // Verifies that the SessionDescription have rejected the appropriate media
|
| - // content.
|
| - void VerifyRejectedMediaInSessionDescription() {
|
| - ASSERT_TRUE(peer_connection_->remote_description() != nullptr);
|
| - ASSERT_TRUE(peer_connection_->local_description() != nullptr);
|
| - const cricket::SessionDescription* remote_desc =
|
| - peer_connection_->remote_description()->description();
|
| - const cricket::SessionDescription* local_desc =
|
| - peer_connection_->local_description()->description();
|
| -
|
| - const ContentInfo* remote_audio_content = GetFirstAudioContent(remote_desc);
|
| - if (remote_audio_content) {
|
| - const ContentInfo* audio_content =
|
| - GetFirstAudioContent(local_desc);
|
| - EXPECT_EQ(can_receive_audio(), !audio_content->rejected);
|
| - }
|
| -
|
| - const ContentInfo* remote_video_content = GetFirstVideoContent(remote_desc);
|
| - if (remote_video_content) {
|
| - const ContentInfo* video_content =
|
| - GetFirstVideoContent(local_desc);
|
| - EXPECT_EQ(can_receive_video(), !video_content->rejected);
|
| - }
|
| - }
|
| -
|
| - void VerifyLocalIceUfragAndPassword() {
|
| - ASSERT_TRUE(peer_connection_->local_description() != nullptr);
|
| - const cricket::SessionDescription* desc =
|
| - peer_connection_->local_description()->description();
|
| - const cricket::ContentInfos& contents = desc->contents();
|
| -
|
| - for (size_t index = 0; index < contents.size(); ++index) {
|
| - if (contents[index].rejected)
|
| - continue;
|
| - const cricket::TransportDescription* transport_desc =
|
| - desc->GetTransportDescriptionByName(contents[index].name);
|
| -
|
| - std::map<int, IceUfragPwdPair>::const_iterator ufragpair_it =
|
| - ice_ufrag_pwd_.find(static_cast<int>(index));
|
| - if (ufragpair_it == ice_ufrag_pwd_.end()) {
|
| - ASSERT_FALSE(ExpectIceRestart());
|
| - ice_ufrag_pwd_[static_cast<int>(index)] =
|
| - IceUfragPwdPair(transport_desc->ice_ufrag, transport_desc->ice_pwd);
|
| - } else if (ExpectIceRestart()) {
|
| - const IceUfragPwdPair& ufrag_pwd = ufragpair_it->second;
|
| - EXPECT_NE(ufrag_pwd.first, transport_desc->ice_ufrag);
|
| - EXPECT_NE(ufrag_pwd.second, transport_desc->ice_pwd);
|
| - } else {
|
| - const IceUfragPwdPair& ufrag_pwd = ufragpair_it->second;
|
| - EXPECT_EQ(ufrag_pwd.first, transport_desc->ice_ufrag);
|
| - EXPECT_EQ(ufrag_pwd.second, transport_desc->ice_pwd);
|
| - }
|
| - }
|
| - }
|
| -
|
| - void VerifyLocalIceRenomination() {
|
| - ASSERT_TRUE(peer_connection_->local_description() != nullptr);
|
| - const cricket::SessionDescription* desc =
|
| - peer_connection_->local_description()->description();
|
| - const cricket::ContentInfos& contents = desc->contents();
|
| -
|
| - for (auto content : contents) {
|
| - if (content.rejected)
|
| - continue;
|
| - const cricket::TransportDescription* transport_desc =
|
| - desc->GetTransportDescriptionByName(content.name);
|
| - const auto& options = transport_desc->transport_options;
|
| - auto iter = std::find(options.begin(), options.end(),
|
| - cricket::ICE_RENOMINATION_STR);
|
| - EXPECT_EQ(ExpectIceRenomination(), iter != options.end());
|
| - }
|
| - }
|
| -
|
| - void VerifyRemoteIceRenomination() {
|
| - ASSERT_TRUE(peer_connection_->remote_description() != nullptr);
|
| - const cricket::SessionDescription* desc =
|
| - peer_connection_->remote_description()->description();
|
| - const cricket::ContentInfos& contents = desc->contents();
|
| -
|
| - for (auto content : contents) {
|
| - if (content.rejected)
|
| - continue;
|
| - const cricket::TransportDescription* transport_desc =
|
| - desc->GetTransportDescriptionByName(content.name);
|
| - const auto& options = transport_desc->transport_options;
|
| - auto iter = std::find(options.begin(), options.end(),
|
| - cricket::ICE_RENOMINATION_STR);
|
| - EXPECT_EQ(ExpectRemoteIceRenomination(), iter != options.end());
|
| - }
|
| - }
|
| -
|
| - int GetAudioOutputLevelStats(webrtc::MediaStreamTrackInterface* track) {
|
| - rtc::scoped_refptr<MockStatsObserver>
|
| - observer(new rtc::RefCountedObject<MockStatsObserver>());
|
| - EXPECT_TRUE(peer_connection_->GetStats(
|
| - observer, track, PeerConnectionInterface::kStatsOutputLevelStandard));
|
| - EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
|
| - EXPECT_NE(0, observer->timestamp());
|
| - return observer->AudioOutputLevel();
|
| - }
|
| -
|
| - int GetAudioInputLevelStats() {
|
| - rtc::scoped_refptr<MockStatsObserver>
|
| - observer(new rtc::RefCountedObject<MockStatsObserver>());
|
| - EXPECT_TRUE(peer_connection_->GetStats(
|
| - observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard));
|
| - EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
|
| - EXPECT_NE(0, observer->timestamp());
|
| - return observer->AudioInputLevel();
|
| - }
|
| -
|
| - int GetBytesReceivedStats(webrtc::MediaStreamTrackInterface* track) {
|
| - rtc::scoped_refptr<MockStatsObserver>
|
| - observer(new rtc::RefCountedObject<MockStatsObserver>());
|
| - EXPECT_TRUE(peer_connection_->GetStats(
|
| - observer, track, PeerConnectionInterface::kStatsOutputLevelStandard));
|
| - EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
|
| - EXPECT_NE(0, observer->timestamp());
|
| - return observer->BytesReceived();
|
| - }
|
| -
|
| - int GetBytesSentStats(webrtc::MediaStreamTrackInterface* track) {
|
| - rtc::scoped_refptr<MockStatsObserver>
|
| - observer(new rtc::RefCountedObject<MockStatsObserver>());
|
| - EXPECT_TRUE(peer_connection_->GetStats(
|
| - observer, track, PeerConnectionInterface::kStatsOutputLevelStandard));
|
| - EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
|
| - EXPECT_NE(0, observer->timestamp());
|
| - return observer->BytesSent();
|
| - }
|
| -
|
| - int GetAvailableReceivedBandwidthStats() {
|
| - rtc::scoped_refptr<MockStatsObserver>
|
| - observer(new rtc::RefCountedObject<MockStatsObserver>());
|
| - EXPECT_TRUE(peer_connection_->GetStats(
|
| - observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard));
|
| - EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
|
| - EXPECT_NE(0, observer->timestamp());
|
| - int bw = observer->AvailableReceiveBandwidth();
|
| - return bw;
|
| - }
|
| -
|
| - std::string GetDtlsCipherStats() {
|
| - rtc::scoped_refptr<MockStatsObserver>
|
| - observer(new rtc::RefCountedObject<MockStatsObserver>());
|
| - EXPECT_TRUE(peer_connection_->GetStats(
|
| - observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard));
|
| - EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
|
| - EXPECT_NE(0, observer->timestamp());
|
| - return observer->DtlsCipher();
|
| - }
|
| -
|
| - std::string GetSrtpCipherStats() {
|
| - rtc::scoped_refptr<MockStatsObserver>
|
| - observer(new rtc::RefCountedObject<MockStatsObserver>());
|
| - EXPECT_TRUE(peer_connection_->GetStats(
|
| - observer, nullptr, PeerConnectionInterface::kStatsOutputLevelStandard));
|
| - EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
|
| - EXPECT_NE(0, observer->timestamp());
|
| - return observer->SrtpCipher();
|
| - }
|
| -
|
| - int rendered_width() {
|
| - EXPECT_FALSE(fake_video_renderers_.empty());
|
| - return fake_video_renderers_.empty() ? 1 :
|
| - fake_video_renderers_.begin()->second->width();
|
| - }
|
| -
|
| - int rendered_height() {
|
| - EXPECT_FALSE(fake_video_renderers_.empty());
|
| - return fake_video_renderers_.empty() ? 1 :
|
| - fake_video_renderers_.begin()->second->height();
|
| - }
|
| -
|
| - webrtc::VideoRotation rendered_rotation() {
|
| - EXPECT_FALSE(fake_video_renderers_.empty());
|
| - return fake_video_renderers_.empty()
|
| - ? webrtc::kVideoRotation_0
|
| - : fake_video_renderers_.begin()->second->rotation();
|
| - }
|
| -
|
| - int local_rendered_width() {
|
| - return local_video_renderer_ ? local_video_renderer_->width() : 1;
|
| - }
|
| -
|
| - int local_rendered_height() {
|
| - return local_video_renderer_ ? local_video_renderer_->height() : 1;
|
| - }
|
| -
|
| - size_t number_of_remote_streams() {
|
| - if (!pc())
|
| - return 0;
|
| - return pc()->remote_streams()->count();
|
| - }
|
| -
|
| - StreamCollectionInterface* remote_streams() const {
|
| - if (!pc()) {
|
| - ADD_FAILURE();
|
| - return nullptr;
|
| - }
|
| - return pc()->remote_streams();
|
| - }
|
| -
|
| - StreamCollectionInterface* local_streams() {
|
| - if (!pc()) {
|
| - ADD_FAILURE();
|
| - return nullptr;
|
| - }
|
| - return pc()->local_streams();
|
| - }
|
| -
|
| - bool HasLocalAudioTrack() { return StreamsHaveAudioTrack(local_streams()); }
|
| -
|
| - bool HasLocalVideoTrack() { return StreamsHaveVideoTrack(local_streams()); }
|
| -
|
| - webrtc::PeerConnectionInterface::SignalingState signaling_state() {
|
| - return pc()->signaling_state();
|
| - }
|
| -
|
| - webrtc::PeerConnectionInterface::IceConnectionState ice_connection_state() {
|
| - return pc()->ice_connection_state();
|
| - }
|
| -
|
| - webrtc::PeerConnectionInterface::IceGatheringState ice_gathering_state() {
|
| - return pc()->ice_gathering_state();
|
| - }
|
| -
|
| - std::vector<std::unique_ptr<MockRtpReceiverObserver>> const&
|
| - rtp_receiver_observers() {
|
| - return rtp_receiver_observers_;
|
| - }
|
| -
|
| - void SetRtpReceiverObservers() {
|
| - rtp_receiver_observers_.clear();
|
| - for (auto receiver : pc()->GetReceivers()) {
|
| - std::unique_ptr<MockRtpReceiverObserver> observer(
|
| - new MockRtpReceiverObserver(receiver->media_type()));
|
| - receiver->SetObserver(observer.get());
|
| - rtp_receiver_observers_.push_back(std::move(observer));
|
| - }
|
| - }
|
| -
|
| - private:
|
| - class DummyDtmfObserver : public DtmfSenderObserverInterface {
|
| - public:
|
| - DummyDtmfObserver() : completed_(false) {}
|
| -
|
| - // Implements DtmfSenderObserverInterface.
|
| - void OnToneChange(const std::string& tone) override {
|
| - tones_.push_back(tone);
|
| - if (tone.empty()) {
|
| - completed_ = true;
|
| - }
|
| - }
|
| -
|
| - void Verify(const std::vector<std::string>& tones) const {
|
| - ASSERT_TRUE(tones_.size() == tones.size());
|
| - EXPECT_TRUE(std::equal(tones.begin(), tones.end(), tones_.begin()));
|
| - }
|
| -
|
| - bool completed() const { return completed_; }
|
| -
|
| - private:
|
| - bool completed_;
|
| - std::vector<std::string> tones_;
|
| - };
|
| -
|
| - explicit PeerConnectionTestClient(const std::string& id) : id_(id) {}
|
| -
|
| - bool Init(
|
| - const MediaConstraintsInterface* constraints,
|
| - const PeerConnectionFactory::Options* options,
|
| - const PeerConnectionInterface::RTCConfiguration* config,
|
| - std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
|
| - bool prefer_constraint_apis,
|
| - rtc::Thread* network_thread,
|
| - rtc::Thread* worker_thread) {
|
| - EXPECT_TRUE(!peer_connection_);
|
| - EXPECT_TRUE(!peer_connection_factory_);
|
| - if (!prefer_constraint_apis) {
|
| - EXPECT_TRUE(!constraints);
|
| - }
|
| - prefer_constraint_apis_ = prefer_constraint_apis;
|
| -
|
| - fake_network_manager_.reset(new rtc::FakeNetworkManager());
|
| - fake_network_manager_->AddInterface(rtc::SocketAddress("192.168.1.1", 0));
|
| -
|
| - std::unique_ptr<cricket::PortAllocator> port_allocator(
|
| - new cricket::BasicPortAllocator(fake_network_manager_.get()));
|
| - fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
|
| -
|
| - if (fake_audio_capture_module_ == nullptr) {
|
| - return false;
|
| - }
|
| - fake_video_decoder_factory_ = new FakeWebRtcVideoDecoderFactory();
|
| - fake_video_encoder_factory_ = new FakeWebRtcVideoEncoderFactory();
|
| - rtc::Thread* const signaling_thread = rtc::Thread::Current();
|
| - peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
|
| - network_thread, worker_thread, signaling_thread,
|
| - fake_audio_capture_module_, fake_video_encoder_factory_,
|
| - fake_video_decoder_factory_);
|
| - if (!peer_connection_factory_) {
|
| - return false;
|
| - }
|
| - if (options) {
|
| - peer_connection_factory_->SetOptions(*options);
|
| - }
|
| - peer_connection_ =
|
| - CreatePeerConnection(std::move(port_allocator), constraints, config,
|
| - std::move(cert_generator));
|
| - return peer_connection_.get() != nullptr;
|
| - }
|
| -
|
| - rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(
|
| - std::unique_ptr<cricket::PortAllocator> port_allocator,
|
| - const MediaConstraintsInterface* constraints,
|
| - const PeerConnectionInterface::RTCConfiguration* config,
|
| - std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator) {
|
| - // CreatePeerConnection with RTCConfiguration.
|
| - PeerConnectionInterface::RTCConfiguration default_config;
|
| -
|
| - if (!config) {
|
| - config = &default_config;
|
| - }
|
| -
|
| - return peer_connection_factory_->CreatePeerConnection(
|
| - *config, constraints, std::move(port_allocator),
|
| - std::move(cert_generator), this);
|
| - }
|
| -
|
| - void HandleIncomingOffer(const std::string& msg) {
|
| - LOG(INFO) << id_ << "HandleIncomingOffer ";
|
| - if (NumberOfLocalMediaStreams() == 0 && auto_add_stream_) {
|
| - // If we are not sending any streams ourselves it is time to add some.
|
| - AddMediaStream(true, true);
|
| - }
|
| - std::unique_ptr<SessionDescriptionInterface> desc(
|
| - webrtc::CreateSessionDescription("offer", msg, nullptr));
|
| - EXPECT_TRUE(DoSetRemoteDescription(desc.release()));
|
| - // Set the RtpReceiverObserver after receivers are created.
|
| - SetRtpReceiverObservers();
|
| - std::unique_ptr<SessionDescriptionInterface> answer;
|
| - EXPECT_TRUE(DoCreateAnswer(&answer));
|
| - std::string sdp;
|
| - EXPECT_TRUE(answer->ToString(&sdp));
|
| - EXPECT_TRUE(DoSetLocalDescription(answer.release()));
|
| - SendSdpMessage(webrtc::SessionDescriptionInterface::kAnswer, sdp);
|
| - }
|
| -
|
| - void HandleIncomingAnswer(const std::string& msg) {
|
| - LOG(INFO) << id_ << "HandleIncomingAnswer";
|
| - std::unique_ptr<SessionDescriptionInterface> desc(
|
| - webrtc::CreateSessionDescription("answer", msg, nullptr));
|
| - EXPECT_TRUE(DoSetRemoteDescription(desc.release()));
|
| - // Set the RtpReceiverObserver after receivers are created.
|
| - SetRtpReceiverObservers();
|
| - }
|
| -
|
| - bool DoCreateOfferAnswer(std::unique_ptr<SessionDescriptionInterface>* desc,
|
| - bool offer) {
|
| - rtc::scoped_refptr<MockCreateSessionDescriptionObserver>
|
| - observer(new rtc::RefCountedObject<
|
| - MockCreateSessionDescriptionObserver>());
|
| - if (prefer_constraint_apis_) {
|
| - if (offer) {
|
| - pc()->CreateOffer(observer, &offer_answer_constraints_);
|
| - } else {
|
| - pc()->CreateAnswer(observer, &offer_answer_constraints_);
|
| - }
|
| - } else {
|
| - if (offer) {
|
| - pc()->CreateOffer(observer, offer_answer_options_);
|
| - } else {
|
| - pc()->CreateAnswer(observer, offer_answer_options_);
|
| - }
|
| - }
|
| - EXPECT_EQ_WAIT(true, observer->called(), kMaxWaitMs);
|
| - desc->reset(observer->release_desc());
|
| - if (observer->result() && ExpectIceRestart()) {
|
| - EXPECT_EQ(0u, (*desc)->candidates(0)->count());
|
| - }
|
| - return observer->result();
|
| - }
|
| -
|
| - bool DoCreateOffer(std::unique_ptr<SessionDescriptionInterface>* desc) {
|
| - return DoCreateOfferAnswer(desc, true);
|
| - }
|
| -
|
| - bool DoCreateAnswer(std::unique_ptr<SessionDescriptionInterface>* desc) {
|
| - return DoCreateOfferAnswer(desc, false);
|
| - }
|
| -
|
| - bool DoSetLocalDescription(SessionDescriptionInterface* desc) {
|
| - rtc::scoped_refptr<MockSetSessionDescriptionObserver>
|
| - observer(new rtc::RefCountedObject<
|
| - MockSetSessionDescriptionObserver>());
|
| - LOG(INFO) << id_ << "SetLocalDescription ";
|
| - pc()->SetLocalDescription(observer, desc);
|
| - // Ignore the observer result. If we wait for the result with
|
| - // EXPECT_TRUE_WAIT, local ice candidates might be sent to the remote peer
|
| - // before the offer which is an error.
|
| - // The reason is that EXPECT_TRUE_WAIT uses
|
| - // rtc::Thread::Current()->ProcessMessages(1);
|
| - // ProcessMessages waits at least 1ms but processes all messages before
|
| - // returning. Since this test is synchronous and send messages to the remote
|
| - // peer whenever a callback is invoked, this can lead to messages being
|
| - // sent to the remote peer in the wrong order.
|
| - // TODO(perkj): Find a way to check the result without risking that the
|
| - // order of sent messages are changed. Ex- by posting all messages that are
|
| - // sent to the remote peer.
|
| - return true;
|
| - }
|
| -
|
| - bool DoSetRemoteDescription(SessionDescriptionInterface* desc) {
|
| - rtc::scoped_refptr<MockSetSessionDescriptionObserver>
|
| - observer(new rtc::RefCountedObject<
|
| - MockSetSessionDescriptionObserver>());
|
| - LOG(INFO) << id_ << "SetRemoteDescription ";
|
| - pc()->SetRemoteDescription(observer, desc);
|
| - EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
|
| - return observer->result();
|
| - }
|
| -
|
| - // This modifies all received SDP messages before they are processed.
|
| - void FilterIncomingSdpMessage(std::string* sdp) {
|
| - if (remove_msid_) {
|
| - const char kSdpSsrcAttribute[] = "a=ssrc:";
|
| - RemoveLinesFromSdp(kSdpSsrcAttribute, sdp);
|
| - const char kSdpMsidSupportedAttribute[] = "a=msid-semantic:";
|
| - RemoveLinesFromSdp(kSdpMsidSupportedAttribute, sdp);
|
| - }
|
| - if (remove_bundle_) {
|
| - const char kSdpBundleAttribute[] = "a=group:BUNDLE";
|
| - RemoveLinesFromSdp(kSdpBundleAttribute, sdp);
|
| - }
|
| - if (remove_sdes_) {
|
| - const char kSdpSdesCryptoAttribute[] = "a=crypto";
|
| - RemoveLinesFromSdp(kSdpSdesCryptoAttribute, sdp);
|
| - }
|
| - if (remove_cvo_) {
|
| - const char kSdpCvoExtenstion[] = "urn:3gpp:video-orientation";
|
| - RemoveLinesFromSdp(kSdpCvoExtenstion, sdp);
|
| - }
|
| - }
|
| -
|
| - std::string id_;
|
| -
|
| - std::unique_ptr<rtc::FakeNetworkManager> fake_network_manager_;
|
| -
|
| - rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
|
| - rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
|
| - peer_connection_factory_;
|
| -
|
| - bool prefer_constraint_apis_ = true;
|
| - bool auto_add_stream_ = true;
|
| -
|
| - typedef std::pair<std::string, std::string> IceUfragPwdPair;
|
| - std::map<int, IceUfragPwdPair> ice_ufrag_pwd_;
|
| - bool expect_ice_restart_ = false;
|
| - bool expect_ice_renomination_ = false;
|
| - bool expect_remote_ice_renomination_ = false;
|
| -
|
| - // Needed to keep track of number of frames sent.
|
| - rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
|
| - // Needed to keep track of number of frames received.
|
| - std::map<std::string, std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
|
| - fake_video_renderers_;
|
| - // Needed to ensure frames aren't received for removed tracks.
|
| - std::vector<std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
|
| - removed_fake_video_renderers_;
|
| - // Needed to keep track of number of frames received when external decoder
|
| - // used.
|
| - FakeWebRtcVideoDecoderFactory* fake_video_decoder_factory_ = nullptr;
|
| - FakeWebRtcVideoEncoderFactory* fake_video_encoder_factory_ = nullptr;
|
| - bool video_decoder_factory_enabled_ = false;
|
| - webrtc::FakeConstraints video_constraints_;
|
| -
|
| - // For remote peer communication.
|
| - SignalingMessageReceiver* signaling_message_receiver_ = nullptr;
|
| - int signaling_delay_ms_ = 0;
|
| -
|
| - // Store references to the video capturers we've created, so that we can stop
|
| - // them, if required.
|
| - std::vector<cricket::FakeVideoCapturer*> video_capturers_;
|
| - webrtc::VideoRotation capture_rotation_ = webrtc::kVideoRotation_0;
|
| - // |local_video_renderer_| attached to the first created local video track.
|
| - std::unique_ptr<webrtc::FakeVideoTrackRenderer> local_video_renderer_;
|
| -
|
| - webrtc::FakeConstraints offer_answer_constraints_;
|
| - PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options_;
|
| - bool remove_msid_ = false; // True if MSID should be removed in received SDP.
|
| - bool remove_bundle_ =
|
| - false; // True if bundle should be removed in received SDP.
|
| - bool remove_sdes_ =
|
| - false; // True if a=crypto should be removed in received SDP.
|
| - // |remove_cvo_| is true if extension urn:3gpp:video-orientation should be
|
| - // removed in the received SDP.
|
| - bool remove_cvo_ = false;
|
| -
|
| - rtc::scoped_refptr<DataChannelInterface> data_channel_;
|
| - std::unique_ptr<MockDataChannelObserver> data_observer_;
|
| -
|
| - std::vector<std::unique_ptr<MockRtpReceiverObserver>> rtp_receiver_observers_;
|
| -};
|
| -
|
| -class P2PTestConductor : public testing::Test {
|
| - public:
|
| - P2PTestConductor()
|
| - : pss_(new rtc::PhysicalSocketServer),
|
| - ss_(new rtc::VirtualSocketServer(pss_.get())),
|
| - network_thread_(new rtc::Thread(ss_.get())),
|
| - worker_thread_(rtc::Thread::Create()) {
|
| - RTC_CHECK(network_thread_->Start());
|
| - RTC_CHECK(worker_thread_->Start());
|
| - }
|
| -
|
| - bool SessionActive() {
|
| - return initiating_client_->SessionActive() &&
|
| - receiving_client_->SessionActive();
|
| - }
|
| -
|
| - // Return true if the number of frames provided have been received
|
| - // on the video and audio tracks provided.
|
| - bool FramesHaveArrived(int audio_frames_to_receive,
|
| - int video_frames_to_receive) {
|
| - bool all_good = true;
|
| - if (initiating_client_->HasLocalAudioTrack() &&
|
| - receiving_client_->can_receive_audio()) {
|
| - all_good &=
|
| - receiving_client_->AudioFramesReceivedCheck(audio_frames_to_receive);
|
| - }
|
| - if (initiating_client_->HasLocalVideoTrack() &&
|
| - receiving_client_->can_receive_video()) {
|
| - all_good &=
|
| - receiving_client_->VideoFramesReceivedCheck(video_frames_to_receive);
|
| - }
|
| - if (receiving_client_->HasLocalAudioTrack() &&
|
| - initiating_client_->can_receive_audio()) {
|
| - all_good &=
|
| - initiating_client_->AudioFramesReceivedCheck(audio_frames_to_receive);
|
| - }
|
| - if (receiving_client_->HasLocalVideoTrack() &&
|
| - initiating_client_->can_receive_video()) {
|
| - all_good &=
|
| - initiating_client_->VideoFramesReceivedCheck(video_frames_to_receive);
|
| - }
|
| - return all_good;
|
| - }
|
| -
|
| - void VerifyDtmf() {
|
| - initiating_client_->VerifyDtmf();
|
| - receiving_client_->VerifyDtmf();
|
| - }
|
| -
|
| - void TestUpdateOfferWithRejectedContent() {
|
| - // Renegotiate, rejecting the video m-line.
|
| - initiating_client_->Negotiate(true, false);
|
| - ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
|
| -
|
| - int pc1_audio_received = initiating_client_->audio_frames_received();
|
| - int pc1_video_received = initiating_client_->video_frames_received();
|
| - int pc2_audio_received = receiving_client_->audio_frames_received();
|
| - int pc2_video_received = receiving_client_->video_frames_received();
|
| -
|
| - // Wait for some additional audio frames to be received.
|
| - EXPECT_TRUE_WAIT(initiating_client_->AudioFramesReceivedCheck(
|
| - pc1_audio_received + kEndAudioFrameCount) &&
|
| - receiving_client_->AudioFramesReceivedCheck(
|
| - pc2_audio_received + kEndAudioFrameCount),
|
| - kMaxWaitForFramesMs);
|
| -
|
| - // During this time, we shouldn't have received any additional video frames
|
| - // for the rejected video tracks.
|
| - EXPECT_EQ(pc1_video_received, initiating_client_->video_frames_received());
|
| - EXPECT_EQ(pc2_video_received, receiving_client_->video_frames_received());
|
| - }
|
| -
|
| - void VerifyRenderedAspectRatio(int width, int height) {
|
| - VerifyRenderedAspectRatio(width, height, webrtc::kVideoRotation_0);
|
| - }
|
| -
|
| - void VerifyRenderedAspectRatio(int width,
|
| - int height,
|
| - webrtc::VideoRotation rotation) {
|
| - double expected_aspect_ratio = static_cast<double>(width) / height;
|
| - double receiving_client_rendered_aspect_ratio =
|
| - static_cast<double>(receiving_client()->rendered_width()) /
|
| - receiving_client()->rendered_height();
|
| - double initializing_client_rendered_aspect_ratio =
|
| - static_cast<double>(initializing_client()->rendered_width()) /
|
| - initializing_client()->rendered_height();
|
| - double initializing_client_local_rendered_aspect_ratio =
|
| - static_cast<double>(initializing_client()->local_rendered_width()) /
|
| - initializing_client()->local_rendered_height();
|
| - // Verify end-to-end rendered aspect ratio.
|
| - EXPECT_EQ(expected_aspect_ratio, receiving_client_rendered_aspect_ratio);
|
| - EXPECT_EQ(expected_aspect_ratio, initializing_client_rendered_aspect_ratio);
|
| - // Verify aspect ratio of the local preview.
|
| - EXPECT_EQ(expected_aspect_ratio,
|
| - initializing_client_local_rendered_aspect_ratio);
|
| -
|
| - // Verify rotation.
|
| - EXPECT_EQ(rotation, receiving_client()->rendered_rotation());
|
| - EXPECT_EQ(rotation, initializing_client()->rendered_rotation());
|
| - }
|
| -
|
| - void VerifySessionDescriptions() {
|
| - initiating_client_->VerifyRejectedMediaInSessionDescription();
|
| - receiving_client_->VerifyRejectedMediaInSessionDescription();
|
| - initiating_client_->VerifyLocalIceUfragAndPassword();
|
| - receiving_client_->VerifyLocalIceUfragAndPassword();
|
| - }
|
| -
|
| - ~P2PTestConductor() {
|
| - if (initiating_client_) {
|
| - initiating_client_->set_signaling_message_receiver(nullptr);
|
| - }
|
| - if (receiving_client_) {
|
| - receiving_client_->set_signaling_message_receiver(nullptr);
|
| - }
|
| - }
|
| -
|
| - bool CreateTestClients() { return CreateTestClients(nullptr, nullptr); }
|
| -
|
| - bool CreateTestClients(MediaConstraintsInterface* init_constraints,
|
| - MediaConstraintsInterface* recv_constraints) {
|
| - return CreateTestClients(init_constraints, nullptr, nullptr,
|
| - recv_constraints, nullptr, nullptr);
|
| - }
|
| -
|
| - bool CreateTestClients(
|
| - const PeerConnectionInterface::RTCConfiguration& init_config,
|
| - const PeerConnectionInterface::RTCConfiguration& recv_config) {
|
| - return CreateTestClients(nullptr, nullptr, &init_config, nullptr, nullptr,
|
| - &recv_config);
|
| - }
|
| -
|
| - bool CreateTestClientsThatPreferNoConstraints() {
|
| - initiating_client_.reset(
|
| - PeerConnectionTestClient::CreateClientPreferNoConstraints(
|
| - "Caller: ", nullptr, network_thread_.get(), worker_thread_.get()));
|
| - receiving_client_.reset(
|
| - PeerConnectionTestClient::CreateClientPreferNoConstraints(
|
| - "Callee: ", nullptr, network_thread_.get(), worker_thread_.get()));
|
| - if (!initiating_client_ || !receiving_client_) {
|
| - return false;
|
| - }
|
| - // Remember the choice for possible later resets of the clients.
|
| - prefer_constraint_apis_ = false;
|
| - SetSignalingReceivers();
|
| - return true;
|
| - }
|
| -
|
| - bool CreateTestClients(
|
| - MediaConstraintsInterface* init_constraints,
|
| - PeerConnectionFactory::Options* init_options,
|
| - const PeerConnectionInterface::RTCConfiguration* init_config,
|
| - MediaConstraintsInterface* recv_constraints,
|
| - PeerConnectionFactory::Options* recv_options,
|
| - const PeerConnectionInterface::RTCConfiguration* recv_config) {
|
| - initiating_client_.reset(PeerConnectionTestClient::CreateClient(
|
| - "Caller: ", init_constraints, init_options, init_config,
|
| - network_thread_.get(), worker_thread_.get()));
|
| - receiving_client_.reset(PeerConnectionTestClient::CreateClient(
|
| - "Callee: ", recv_constraints, recv_options, recv_config,
|
| - network_thread_.get(), worker_thread_.get()));
|
| - if (!initiating_client_ || !receiving_client_) {
|
| - return false;
|
| - }
|
| - SetSignalingReceivers();
|
| - return true;
|
| - }
|
| -
|
| - void SetSignalingReceivers() {
|
| - initiating_client_->set_signaling_message_receiver(receiving_client_.get());
|
| - receiving_client_->set_signaling_message_receiver(initiating_client_.get());
|
| - }
|
| -
|
| - void SetSignalingDelayMs(int delay_ms) {
|
| - initiating_client_->set_signaling_delay_ms(delay_ms);
|
| - receiving_client_->set_signaling_delay_ms(delay_ms);
|
| - }
|
| -
|
| - void SetVideoConstraints(const webrtc::FakeConstraints& init_constraints,
|
| - const webrtc::FakeConstraints& recv_constraints) {
|
| - initiating_client_->SetVideoConstraints(init_constraints);
|
| - receiving_client_->SetVideoConstraints(recv_constraints);
|
| - }
|
| -
|
| - void SetCaptureRotation(webrtc::VideoRotation rotation) {
|
| - initiating_client_->SetCaptureRotation(rotation);
|
| - receiving_client_->SetCaptureRotation(rotation);
|
| - }
|
| -
|
| - void EnableVideoDecoderFactory() {
|
| - initiating_client_->EnableVideoDecoderFactory();
|
| - receiving_client_->EnableVideoDecoderFactory();
|
| - }
|
| -
|
| - // This test sets up a call between two parties. Both parties send static
|
| - // frames to each other. Once the test is finished the number of sent frames
|
| - // is compared to the number of received frames.
|
| - void LocalP2PTest() {
|
| - if (initiating_client_->NumberOfLocalMediaStreams() == 0) {
|
| - initiating_client_->AddMediaStream(true, true);
|
| - }
|
| - initiating_client_->Negotiate();
|
| - // Assert true is used here since next tests are guaranteed to fail and
|
| - // would eat up 5 seconds.
|
| - ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
|
| - VerifySessionDescriptions();
|
| -
|
| - int audio_frame_count = kEndAudioFrameCount;
|
| - int video_frame_count = kEndVideoFrameCount;
|
| - // TODO(ronghuawu): Add test to cover the case of sendonly and recvonly.
|
| -
|
| - if ((!initiating_client_->can_receive_audio() &&
|
| - !initiating_client_->can_receive_video()) ||
|
| - (!receiving_client_->can_receive_audio() &&
|
| - !receiving_client_->can_receive_video())) {
|
| - // Neither audio nor video will flow, so connections won't be
|
| - // established. There's nothing more to check.
|
| - // TODO(hta): Check connection if there's a data channel.
|
| - return;
|
| - }
|
| -
|
| - // Audio or video is expected to flow, so both clients should reach the
|
| - // Connected state, and the offerer (ICE controller) should proceed to
|
| - // Completed.
|
| - // Note: These tests have been observed to fail under heavy load at
|
| - // shorter timeouts, so they may be flaky.
|
| - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
|
| - initiating_client_->ice_connection_state(),
|
| - kMaxWaitForFramesMs);
|
| - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
|
| - receiving_client_->ice_connection_state(),
|
| - kMaxWaitForFramesMs);
|
| -
|
| - // The ICE gathering state should end up in kIceGatheringComplete,
|
| - // but there's a bug that prevents this at the moment, and the state
|
| - // machine is being updated by the WEBRTC WG.
|
| - // TODO(hta): Update this check when spec revisions finish.
|
| - EXPECT_NE(webrtc::PeerConnectionInterface::kIceGatheringNew,
|
| - initiating_client_->ice_gathering_state());
|
| - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
|
| - receiving_client_->ice_gathering_state(),
|
| - kMaxWaitForFramesMs);
|
| -
|
| - // Check that the expected number of frames have arrived.
|
| - EXPECT_TRUE_WAIT(FramesHaveArrived(audio_frame_count, video_frame_count),
|
| - kMaxWaitForFramesMs);
|
| - }
|
| -
|
| - void SetupAndVerifyDtlsCall() {
|
| - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
|
| - FakeConstraints setup_constraints;
|
| - setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
|
| - true);
|
| - // Disable resolution adaptation, we don't want it interfering with the
|
| - // test results.
|
| - webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
|
| - rtc_config.set_cpu_adaptation(false);
|
| -
|
| - ASSERT_TRUE(CreateTestClients(&setup_constraints, nullptr, &rtc_config,
|
| - &setup_constraints, nullptr, &rtc_config));
|
| - LocalP2PTest();
|
| - VerifyRenderedAspectRatio(640, 480);
|
| - }
|
| -
|
| - PeerConnectionTestClient* CreateDtlsClientWithAlternateKey() {
|
| - FakeConstraints setup_constraints;
|
| - setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
|
| - true);
|
| - // Disable resolution adaptation, we don't want it interfering with the
|
| - // test results.
|
| - webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
|
| - rtc_config.set_cpu_adaptation(false);
|
| -
|
| - std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
|
| - rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
|
| - new FakeRTCCertificateGenerator() : nullptr);
|
| - cert_generator->use_alternate_key();
|
| -
|
| - // Make sure the new client is using a different certificate.
|
| - return PeerConnectionTestClient::CreateClientWithDtlsIdentityStore(
|
| - "New Peer: ", &setup_constraints, nullptr, &rtc_config,
|
| - std::move(cert_generator), prefer_constraint_apis_,
|
| - network_thread_.get(), worker_thread_.get());
|
| - }
|
| -
|
| - void SendRtpData(webrtc::DataChannelInterface* dc, const std::string& data) {
|
| - // Messages may get lost on the unreliable DataChannel, so we send multiple
|
| - // times to avoid test flakiness.
|
| - static const size_t kSendAttempts = 5;
|
| -
|
| - for (size_t i = 0; i < kSendAttempts; ++i) {
|
| - dc->Send(DataBuffer(data));
|
| - }
|
| - }
|
| -
|
| - rtc::Thread* network_thread() { return network_thread_.get(); }
|
| -
|
| - rtc::VirtualSocketServer* virtual_socket_server() { return ss_.get(); }
|
| -
|
| - PeerConnectionTestClient* initializing_client() {
|
| - return initiating_client_.get();
|
| - }
|
| -
|
| - // Set the |initiating_client_| to the |client| passed in and return the
|
| - // original |initiating_client_|.
|
| - PeerConnectionTestClient* set_initializing_client(
|
| - PeerConnectionTestClient* client) {
|
| - PeerConnectionTestClient* old = initiating_client_.release();
|
| - initiating_client_.reset(client);
|
| - return old;
|
| - }
|
| -
|
| - PeerConnectionTestClient* receiving_client() {
|
| - return receiving_client_.get();
|
| - }
|
| -
|
| - // Set the |receiving_client_| to the |client| passed in and return the
|
| - // original |receiving_client_|.
|
| - PeerConnectionTestClient* set_receiving_client(
|
| - PeerConnectionTestClient* client) {
|
| - PeerConnectionTestClient* old = receiving_client_.release();
|
| - receiving_client_.reset(client);
|
| - return old;
|
| - }
|
| -
|
| - bool AllObserversReceived(
|
| - const std::vector<std::unique_ptr<MockRtpReceiverObserver>>& observers) {
|
| - for (auto& observer : observers) {
|
| - if (!observer->first_packet_received()) {
|
| - return false;
|
| - }
|
| - }
|
| - return true;
|
| - }
|
| -
|
| - void TestGcmNegotiation(bool local_gcm_enabled, bool remote_gcm_enabled,
|
| - int expected_cipher_suite) {
|
| - PeerConnectionFactory::Options init_options;
|
| - init_options.crypto_options.enable_gcm_crypto_suites = local_gcm_enabled;
|
| - PeerConnectionFactory::Options recv_options;
|
| - recv_options.crypto_options.enable_gcm_crypto_suites = remote_gcm_enabled;
|
| - ASSERT_TRUE(CreateTestClients(nullptr, &init_options, nullptr, nullptr,
|
| - &recv_options, nullptr));
|
| - rtc::scoped_refptr<webrtc::FakeMetricsObserver>
|
| - init_observer =
|
| - new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
|
| - initializing_client()->pc()->RegisterUMAObserver(init_observer);
|
| - LocalP2PTest();
|
| -
|
| - EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(expected_cipher_suite),
|
| - initializing_client()->GetSrtpCipherStats(),
|
| - kMaxWaitMs);
|
| - EXPECT_EQ(1,
|
| - init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
|
| - expected_cipher_suite));
|
| - }
|
| -
|
| - private:
|
| - // |ss_| is used by |network_thread_| so it must be destroyed later.
|
| - std::unique_ptr<rtc::PhysicalSocketServer> pss_;
|
| - std::unique_ptr<rtc::VirtualSocketServer> ss_;
|
| - // |network_thread_| and |worker_thread_| are used by both
|
| - // |initiating_client_| and |receiving_client_| so they must be destroyed
|
| - // later.
|
| - std::unique_ptr<rtc::Thread> network_thread_;
|
| - std::unique_ptr<rtc::Thread> worker_thread_;
|
| - std::unique_ptr<PeerConnectionTestClient> initiating_client_;
|
| - std::unique_ptr<PeerConnectionTestClient> receiving_client_;
|
| - bool prefer_constraint_apis_ = true;
|
| -};
|
| -
|
| -// Disable for TSan v2, see
|
| -// https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
|
| -#if !defined(THREAD_SANITIZER)
|
| -
|
| -TEST_F(P2PTestConductor, TestRtpReceiverObserverCallbackFunction) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - LocalP2PTest();
|
| - EXPECT_TRUE_WAIT(
|
| - AllObserversReceived(initializing_client()->rtp_receiver_observers()),
|
| - kMaxWaitForFramesMs);
|
| - EXPECT_TRUE_WAIT(
|
| - AllObserversReceived(receiving_client()->rtp_receiver_observers()),
|
| - kMaxWaitForFramesMs);
|
| -}
|
| -
|
| -// The observers are expected to fire the signal even if they are set after the
|
| -// first packet is received.
|
| -TEST_F(P2PTestConductor, TestSetRtpReceiverObserverAfterFirstPacketIsReceived) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - LocalP2PTest();
|
| - // Reset the RtpReceiverObservers.
|
| - initializing_client()->SetRtpReceiverObservers();
|
| - receiving_client()->SetRtpReceiverObservers();
|
| - EXPECT_TRUE_WAIT(
|
| - AllObserversReceived(initializing_client()->rtp_receiver_observers()),
|
| - kMaxWaitForFramesMs);
|
| - EXPECT_TRUE_WAIT(
|
| - AllObserversReceived(receiving_client()->rtp_receiver_observers()),
|
| - kMaxWaitForFramesMs);
|
| -}
|
| -
|
| -// This test sets up a Jsep call between two parties and test Dtmf.
|
| -// TODO(holmer): Disabled due to sometimes crashing on buildbots.
|
| -// See issue webrtc/2378.
|
| -TEST_F(P2PTestConductor, DISABLED_LocalP2PTestDtmf) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - LocalP2PTest();
|
| - VerifyDtmf();
|
| -}
|
| -
|
| -// This test sets up a Jsep call between two parties and test that we can get a
|
| -// video aspect ratio of 16:9.
|
| -TEST_F(P2PTestConductor, LocalP2PTest16To9) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - FakeConstraints constraint;
|
| - double requested_ratio = 640.0/360;
|
| - constraint.SetMandatoryMinAspectRatio(requested_ratio);
|
| - SetVideoConstraints(constraint, constraint);
|
| - LocalP2PTest();
|
| -
|
| - ASSERT_LE(0, initializing_client()->rendered_height());
|
| - double initiating_video_ratio =
|
| - static_cast<double>(initializing_client()->rendered_width()) /
|
| - initializing_client()->rendered_height();
|
| - EXPECT_LE(requested_ratio, initiating_video_ratio);
|
| -
|
| - ASSERT_LE(0, receiving_client()->rendered_height());
|
| - double receiving_video_ratio =
|
| - static_cast<double>(receiving_client()->rendered_width()) /
|
| - receiving_client()->rendered_height();
|
| - EXPECT_LE(requested_ratio, receiving_video_ratio);
|
| -}
|
| -
|
| -// This test sets up a Jsep call between two parties and test that the
|
| -// received video has a resolution of 1280*720.
|
| -// TODO(mallinath): Enable when
|
| -// http://code.google.com/p/webrtc/issues/detail?id=981 is fixed.
|
| -TEST_F(P2PTestConductor, DISABLED_LocalP2PTest1280By720) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - FakeConstraints constraint;
|
| - constraint.SetMandatoryMinWidth(1280);
|
| - constraint.SetMandatoryMinHeight(720);
|
| - SetVideoConstraints(constraint, constraint);
|
| - LocalP2PTest();
|
| - VerifyRenderedAspectRatio(1280, 720);
|
| -}
|
| -
|
| -// This test sets up a call between two endpoints that are configured to use
|
| -// DTLS key agreement. As a result, DTLS is negotiated and used for transport.
|
| -TEST_F(P2PTestConductor, LocalP2PTestDtls) {
|
| - SetupAndVerifyDtlsCall();
|
| -}
|
| -
|
| -// This test sets up an one-way call, with media only from initiator to
|
| -// responder.
|
| -TEST_F(P2PTestConductor, OneWayMediaCall) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - receiving_client()->set_auto_add_stream(false);
|
| - LocalP2PTest();
|
| -}
|
| -
|
| -TEST_F(P2PTestConductor, OneWayMediaCallWithoutConstraints) {
|
| - ASSERT_TRUE(CreateTestClientsThatPreferNoConstraints());
|
| - receiving_client()->set_auto_add_stream(false);
|
| - LocalP2PTest();
|
| -}
|
| -
|
| -// This test sets up a audio call initially and then upgrades to audio/video,
|
| -// using DTLS.
|
| -TEST_F(P2PTestConductor, LocalP2PTestDtlsRenegotiate) {
|
| - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
|
| - FakeConstraints setup_constraints;
|
| - setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
|
| - true);
|
| - ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
|
| - receiving_client()->SetReceiveAudioVideo(true, false);
|
| - LocalP2PTest();
|
| - receiving_client()->SetReceiveAudioVideo(true, true);
|
| - receiving_client()->Negotiate();
|
| -}
|
| -
|
| -// This test sets up a call transfer to a new caller with a different DTLS
|
| -// fingerprint.
|
| -TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCallee) {
|
| - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
|
| - SetupAndVerifyDtlsCall();
|
| -
|
| - // Keeping the original peer around which will still send packets to the
|
| - // receiving client. These SRTP packets will be dropped.
|
| - std::unique_ptr<PeerConnectionTestClient> original_peer(
|
| - set_initializing_client(CreateDtlsClientWithAlternateKey()));
|
| - original_peer->pc()->Close();
|
| -
|
| - SetSignalingReceivers();
|
| - receiving_client()->SetExpectIceRestart(true);
|
| - LocalP2PTest();
|
| - VerifyRenderedAspectRatio(640, 480);
|
| -}
|
| -
|
| -// This test sets up a non-bundle call and apply bundle during ICE restart. When
|
| -// bundle is in effect in the restart, the channel can successfully reset its
|
| -// DTLS-SRTP context.
|
| -TEST_F(P2PTestConductor, LocalP2PTestDtlsBundleInIceRestart) {
|
| - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
|
| - FakeConstraints setup_constraints;
|
| - setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
|
| - true);
|
| - ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
|
| - receiving_client()->RemoveBundleFromReceivedSdp(true);
|
| - LocalP2PTest();
|
| - VerifyRenderedAspectRatio(640, 480);
|
| -
|
| - initializing_client()->IceRestart();
|
| - receiving_client()->SetExpectIceRestart(true);
|
| - receiving_client()->RemoveBundleFromReceivedSdp(false);
|
| - LocalP2PTest();
|
| - VerifyRenderedAspectRatio(640, 480);
|
| -}
|
| -
|
| -// This test sets up a call transfer to a new callee with a different DTLS
|
| -// fingerprint.
|
| -TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCaller) {
|
| - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
|
| - SetupAndVerifyDtlsCall();
|
| -
|
| - // Keeping the original peer around which will still send packets to the
|
| - // receiving client. These SRTP packets will be dropped.
|
| - std::unique_ptr<PeerConnectionTestClient> original_peer(
|
| - set_receiving_client(CreateDtlsClientWithAlternateKey()));
|
| - original_peer->pc()->Close();
|
| -
|
| - SetSignalingReceivers();
|
| - initializing_client()->IceRestart();
|
| - LocalP2PTest();
|
| - VerifyRenderedAspectRatio(640, 480);
|
| -}
|
| -
|
| -TEST_F(P2PTestConductor, LocalP2PTestCVO) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - SetCaptureRotation(webrtc::kVideoRotation_90);
|
| - LocalP2PTest();
|
| - VerifyRenderedAspectRatio(640, 480, webrtc::kVideoRotation_90);
|
| -}
|
| -
|
| -TEST_F(P2PTestConductor, LocalP2PTestReceiverDoesntSupportCVO) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - SetCaptureRotation(webrtc::kVideoRotation_90);
|
| - receiving_client()->RemoveCvoFromReceivedSdp(true);
|
| - LocalP2PTest();
|
| - VerifyRenderedAspectRatio(480, 640, webrtc::kVideoRotation_0);
|
| -}
|
| -
|
| -// This test sets up a call between two endpoints that are configured to use
|
| -// DTLS key agreement. The offerer don't support SDES. As a result, DTLS is
|
| -// negotiated and used for transport.
|
| -TEST_F(P2PTestConductor, LocalP2PTestOfferDtlsButNotSdes) {
|
| - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
|
| - FakeConstraints setup_constraints;
|
| - setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
|
| - true);
|
| - ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
|
| - receiving_client()->RemoveSdesCryptoFromReceivedSdp(true);
|
| - LocalP2PTest();
|
| - VerifyRenderedAspectRatio(640, 480);
|
| -}
|
| -
|
| -// This test verifies that the negotiation will succeed with data channel only
|
| -// in max-bundle mode.
|
| -TEST_F(P2PTestConductor, LocalP2PTestOfferDataChannelOnly) {
|
| - webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
|
| - rtc_config.bundle_policy =
|
| - webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle;
|
| - ASSERT_TRUE(CreateTestClients(rtc_config, rtc_config));
|
| - initializing_client()->CreateDataChannel();
|
| - initializing_client()->Negotiate();
|
| -}
|
| -
|
| -// This test sets up a Jsep call between two parties, and the callee only
|
| -// accept to receive video.
|
| -TEST_F(P2PTestConductor, LocalP2PTestAnswerVideo) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - receiving_client()->SetReceiveAudioVideo(false, true);
|
| - LocalP2PTest();
|
| -}
|
| -
|
| -// This test sets up a Jsep call between two parties, and the callee only
|
| -// accept to receive audio.
|
| -TEST_F(P2PTestConductor, LocalP2PTestAnswerAudio) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - receiving_client()->SetReceiveAudioVideo(true, false);
|
| - LocalP2PTest();
|
| -}
|
| -
|
| -// This test sets up a Jsep call between two parties, and the callee reject both
|
| -// audio and video.
|
| -TEST_F(P2PTestConductor, LocalP2PTestAnswerNone) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - receiving_client()->SetReceiveAudioVideo(false, false);
|
| - LocalP2PTest();
|
| -}
|
| -
|
| -// This test sets up an audio and video call between two parties. After the call
|
| -// runs for a while (10 frames), the caller sends an update offer with video
|
| -// being rejected. Once the re-negotiation is done, the video flow should stop
|
| -// and the audio flow should continue.
|
| -TEST_F(P2PTestConductor, UpdateOfferWithRejectedContent) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - LocalP2PTest();
|
| - TestUpdateOfferWithRejectedContent();
|
| -}
|
| -
|
| -// This test sets up a Jsep call between two parties. The MSID is removed from
|
| -// the SDP strings from the caller.
|
| -TEST_F(P2PTestConductor, LocalP2PTestWithoutMsid) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - receiving_client()->RemoveMsidFromReceivedSdp(true);
|
| - // TODO(perkj): Currently there is a bug that cause audio to stop playing if
|
| - // audio and video is muxed when MSID is disabled. Remove
|
| - // SetRemoveBundleFromSdp once
|
| - // https://code.google.com/p/webrtc/issues/detail?id=1193 is fixed.
|
| - receiving_client()->RemoveBundleFromReceivedSdp(true);
|
| - LocalP2PTest();
|
| -}
|
| -
|
| -TEST_F(P2PTestConductor, LocalP2PTestTwoStreams) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - // Set optional video constraint to max 320pixels to decrease CPU usage.
|
| - FakeConstraints constraint;
|
| - constraint.SetOptionalMaxWidth(320);
|
| - SetVideoConstraints(constraint, constraint);
|
| - initializing_client()->AddMediaStream(true, true);
|
| - initializing_client()->AddMediaStream(false, true);
|
| - ASSERT_EQ(2u, initializing_client()->NumberOfLocalMediaStreams());
|
| - LocalP2PTest();
|
| - EXPECT_EQ(2u, receiving_client()->number_of_remote_streams());
|
| -}
|
| -
|
| -// Test that we can receive the audio output level from a remote audio track.
|
| -TEST_F(P2PTestConductor, GetAudioOutputLevelStats) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - LocalP2PTest();
|
| -
|
| - StreamCollectionInterface* remote_streams =
|
| - initializing_client()->remote_streams();
|
| - ASSERT_GT(remote_streams->count(), 0u);
|
| - ASSERT_GT(remote_streams->at(0)->GetAudioTracks().size(), 0u);
|
| - MediaStreamTrackInterface* remote_audio_track =
|
| - remote_streams->at(0)->GetAudioTracks()[0];
|
| -
|
| - // Get the audio output level stats. Note that the level is not available
|
| - // until a RTCP packet has been received.
|
| - EXPECT_TRUE_WAIT(
|
| - initializing_client()->GetAudioOutputLevelStats(remote_audio_track) > 0,
|
| - kMaxWaitForStatsMs);
|
| -}
|
| -
|
| -// Test that an audio input level is reported.
|
| -TEST_F(P2PTestConductor, GetAudioInputLevelStats) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - LocalP2PTest();
|
| -
|
| - // Get the audio input level stats. The level should be available very
|
| - // soon after the test starts.
|
| - EXPECT_TRUE_WAIT(initializing_client()->GetAudioInputLevelStats() > 0,
|
| - kMaxWaitForStatsMs);
|
| -}
|
| -
|
| -// Test that we can get incoming byte counts from both audio and video tracks.
|
| -TEST_F(P2PTestConductor, GetBytesReceivedStats) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - LocalP2PTest();
|
| -
|
| - StreamCollectionInterface* remote_streams =
|
| - initializing_client()->remote_streams();
|
| - ASSERT_GT(remote_streams->count(), 0u);
|
| - ASSERT_GT(remote_streams->at(0)->GetAudioTracks().size(), 0u);
|
| - MediaStreamTrackInterface* remote_audio_track =
|
| - remote_streams->at(0)->GetAudioTracks()[0];
|
| - EXPECT_TRUE_WAIT(
|
| - initializing_client()->GetBytesReceivedStats(remote_audio_track) > 0,
|
| - kMaxWaitForStatsMs);
|
| -
|
| - MediaStreamTrackInterface* remote_video_track =
|
| - remote_streams->at(0)->GetVideoTracks()[0];
|
| - EXPECT_TRUE_WAIT(
|
| - initializing_client()->GetBytesReceivedStats(remote_video_track) > 0,
|
| - kMaxWaitForStatsMs);
|
| -}
|
| -
|
| -// Test that we can get outgoing byte counts from both audio and video tracks.
|
| -TEST_F(P2PTestConductor, GetBytesSentStats) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - LocalP2PTest();
|
| -
|
| - StreamCollectionInterface* local_streams =
|
| - initializing_client()->local_streams();
|
| - ASSERT_GT(local_streams->count(), 0u);
|
| - ASSERT_GT(local_streams->at(0)->GetAudioTracks().size(), 0u);
|
| - MediaStreamTrackInterface* local_audio_track =
|
| - local_streams->at(0)->GetAudioTracks()[0];
|
| - EXPECT_TRUE_WAIT(
|
| - initializing_client()->GetBytesSentStats(local_audio_track) > 0,
|
| - kMaxWaitForStatsMs);
|
| -
|
| - MediaStreamTrackInterface* local_video_track =
|
| - local_streams->at(0)->GetVideoTracks()[0];
|
| - EXPECT_TRUE_WAIT(
|
| - initializing_client()->GetBytesSentStats(local_video_track) > 0,
|
| - kMaxWaitForStatsMs);
|
| -}
|
| -
|
| -// Test that DTLS 1.0 is used if both sides only support DTLS 1.0.
|
| -TEST_F(P2PTestConductor, GetDtls12None) {
|
| - PeerConnectionFactory::Options init_options;
|
| - init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
|
| - PeerConnectionFactory::Options recv_options;
|
| - recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
|
| - ASSERT_TRUE(CreateTestClients(nullptr, &init_options, nullptr, nullptr,
|
| - &recv_options, nullptr));
|
| - rtc::scoped_refptr<webrtc::FakeMetricsObserver>
|
| - init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
|
| - initializing_client()->pc()->RegisterUMAObserver(init_observer);
|
| - LocalP2PTest();
|
| -
|
| - EXPECT_TRUE_WAIT(
|
| - rtc::SSLStreamAdapter::IsAcceptableCipher(
|
| - initializing_client()->GetDtlsCipherStats(), rtc::KT_DEFAULT),
|
| - kMaxWaitForStatsMs);
|
| - EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
|
| - initializing_client()->GetSrtpCipherStats(),
|
| - kMaxWaitForStatsMs);
|
| - EXPECT_EQ(1,
|
| - init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
|
| - kDefaultSrtpCryptoSuite));
|
| -}
|
| -
|
| -// Test that DTLS 1.2 is used if both ends support it.
|
| -TEST_F(P2PTestConductor, GetDtls12Both) {
|
| - PeerConnectionFactory::Options init_options;
|
| - init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
|
| - PeerConnectionFactory::Options recv_options;
|
| - recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
|
| - ASSERT_TRUE(CreateTestClients(nullptr, &init_options, nullptr, nullptr,
|
| - &recv_options, nullptr));
|
| - rtc::scoped_refptr<webrtc::FakeMetricsObserver>
|
| - init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
|
| - initializing_client()->pc()->RegisterUMAObserver(init_observer);
|
| - LocalP2PTest();
|
| -
|
| - EXPECT_TRUE_WAIT(
|
| - rtc::SSLStreamAdapter::IsAcceptableCipher(
|
| - initializing_client()->GetDtlsCipherStats(), rtc::KT_DEFAULT),
|
| - kMaxWaitForStatsMs);
|
| - EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
|
| - initializing_client()->GetSrtpCipherStats(),
|
| - kMaxWaitForStatsMs);
|
| - EXPECT_EQ(1,
|
| - init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
|
| - kDefaultSrtpCryptoSuite));
|
| -}
|
| -
|
| -// Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the
|
| -// received supports 1.0.
|
| -TEST_F(P2PTestConductor, GetDtls12Init) {
|
| - PeerConnectionFactory::Options init_options;
|
| - init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
|
| - PeerConnectionFactory::Options recv_options;
|
| - recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
|
| - ASSERT_TRUE(CreateTestClients(nullptr, &init_options, nullptr, nullptr,
|
| - &recv_options, nullptr));
|
| - rtc::scoped_refptr<webrtc::FakeMetricsObserver>
|
| - init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
|
| - initializing_client()->pc()->RegisterUMAObserver(init_observer);
|
| - LocalP2PTest();
|
| -
|
| - EXPECT_TRUE_WAIT(
|
| - rtc::SSLStreamAdapter::IsAcceptableCipher(
|
| - initializing_client()->GetDtlsCipherStats(), rtc::KT_DEFAULT),
|
| - kMaxWaitForStatsMs);
|
| - EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
|
| - initializing_client()->GetSrtpCipherStats(),
|
| - kMaxWaitForStatsMs);
|
| - EXPECT_EQ(1,
|
| - init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
|
| - kDefaultSrtpCryptoSuite));
|
| -}
|
| -
|
| -// Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the
|
| -// received supports 1.2.
|
| -TEST_F(P2PTestConductor, GetDtls12Recv) {
|
| - PeerConnectionFactory::Options init_options;
|
| - init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
|
| - PeerConnectionFactory::Options recv_options;
|
| - recv_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
|
| - ASSERT_TRUE(CreateTestClients(nullptr, &init_options, nullptr, nullptr,
|
| - &recv_options, nullptr));
|
| - rtc::scoped_refptr<webrtc::FakeMetricsObserver>
|
| - init_observer = new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
|
| - initializing_client()->pc()->RegisterUMAObserver(init_observer);
|
| - LocalP2PTest();
|
| -
|
| - EXPECT_TRUE_WAIT(
|
| - rtc::SSLStreamAdapter::IsAcceptableCipher(
|
| - initializing_client()->GetDtlsCipherStats(), rtc::KT_DEFAULT),
|
| - kMaxWaitForStatsMs);
|
| - EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
|
| - initializing_client()->GetSrtpCipherStats(),
|
| - kMaxWaitForStatsMs);
|
| - EXPECT_EQ(1,
|
| - init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
|
| - kDefaultSrtpCryptoSuite));
|
| -}
|
| -
|
| -// Test that a non-GCM cipher is used if both sides only support non-GCM.
|
| -TEST_F(P2PTestConductor, GetGcmNone) {
|
| - TestGcmNegotiation(false, false, kDefaultSrtpCryptoSuite);
|
| -}
|
| -
|
| -// Test that a GCM cipher is used if both ends support it.
|
| -TEST_F(P2PTestConductor, GetGcmBoth) {
|
| - TestGcmNegotiation(true, true, kDefaultSrtpCryptoSuiteGcm);
|
| -}
|
| -
|
| -// Test that GCM isn't used if only the initiator supports it.
|
| -TEST_F(P2PTestConductor, GetGcmInit) {
|
| - TestGcmNegotiation(true, false, kDefaultSrtpCryptoSuite);
|
| -}
|
| -
|
| -// Test that GCM isn't used if only the receiver supports it.
|
| -TEST_F(P2PTestConductor, GetGcmRecv) {
|
| - TestGcmNegotiation(false, true, kDefaultSrtpCryptoSuite);
|
| -}
|
| -
|
| -// This test sets up a call between two parties with audio, video and an RTP
|
| -// data channel.
|
| -TEST_F(P2PTestConductor, LocalP2PTestRtpDataChannel) {
|
| - FakeConstraints setup_constraints;
|
| - setup_constraints.SetAllowRtpDataChannels();
|
| - ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
|
| - initializing_client()->CreateDataChannel();
|
| - LocalP2PTest();
|
| - ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
|
| - ASSERT_TRUE(receiving_client()->data_channel() != nullptr);
|
| - EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
|
| - kMaxWaitMs);
|
| - EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(),
|
| - kMaxWaitMs);
|
| -
|
| - std::string data = "hello world";
|
| -
|
| - SendRtpData(initializing_client()->data_channel(), data);
|
| - EXPECT_EQ_WAIT(data, receiving_client()->data_observer()->last_message(),
|
| - kMaxWaitMs);
|
| -
|
| - SendRtpData(receiving_client()->data_channel(), data);
|
| - EXPECT_EQ_WAIT(data, initializing_client()->data_observer()->last_message(),
|
| - kMaxWaitMs);
|
| -
|
| - receiving_client()->data_channel()->Close();
|
| - // Send new offer and answer.
|
| - receiving_client()->Negotiate();
|
| - EXPECT_FALSE(initializing_client()->data_observer()->IsOpen());
|
| - EXPECT_FALSE(receiving_client()->data_observer()->IsOpen());
|
| -}
|
| -
|
| -// This test sets up a call between two parties with audio, video and an SCTP
|
| -// data channel.
|
| -TEST_F(P2PTestConductor, LocalP2PTestSctpDataChannel) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - initializing_client()->CreateDataChannel();
|
| - LocalP2PTest();
|
| - ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
|
| - EXPECT_TRUE_WAIT(receiving_client()->data_channel() != nullptr, kMaxWaitMs);
|
| - EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
|
| - kMaxWaitMs);
|
| - EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
|
| -
|
| - std::string data = "hello world";
|
| -
|
| - initializing_client()->data_channel()->Send(DataBuffer(data));
|
| - EXPECT_EQ_WAIT(data, receiving_client()->data_observer()->last_message(),
|
| - kMaxWaitMs);
|
| -
|
| - receiving_client()->data_channel()->Send(DataBuffer(data));
|
| - EXPECT_EQ_WAIT(data, initializing_client()->data_observer()->last_message(),
|
| - kMaxWaitMs);
|
| -
|
| - receiving_client()->data_channel()->Close();
|
| - EXPECT_TRUE_WAIT(!initializing_client()->data_observer()->IsOpen(),
|
| - kMaxWaitMs);
|
| - EXPECT_TRUE_WAIT(!receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
|
| -}
|
| -
|
| -TEST_F(P2PTestConductor, UnorderedSctpDataChannel) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - webrtc::DataChannelInit init;
|
| - init.ordered = false;
|
| - initializing_client()->CreateDataChannel(&init);
|
| -
|
| - // Introduce random network delays.
|
| - // Otherwise it's not a true "unordered" test.
|
| - virtual_socket_server()->set_delay_mean(20);
|
| - virtual_socket_server()->set_delay_stddev(5);
|
| - virtual_socket_server()->UpdateDelayDistribution();
|
| -
|
| - initializing_client()->Negotiate();
|
| - ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
|
| - EXPECT_TRUE_WAIT(receiving_client()->data_channel() != nullptr, kMaxWaitMs);
|
| - EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
|
| - kMaxWaitMs);
|
| - EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
|
| -
|
| - static constexpr int kNumMessages = 100;
|
| - // Deliberately chosen to be larger than the MTU so messages get fragmented.
|
| - static constexpr size_t kMaxMessageSize = 4096;
|
| - // Create and send random messages.
|
| - std::vector<std::string> sent_messages;
|
| - for (int i = 0; i < kNumMessages; ++i) {
|
| - size_t length = (rand() % kMaxMessageSize) + 1;
|
| - std::string message;
|
| - ASSERT_TRUE(rtc::CreateRandomString(length, &message));
|
| - initializing_client()->data_channel()->Send(DataBuffer(message));
|
| - receiving_client()->data_channel()->Send(DataBuffer(message));
|
| - sent_messages.push_back(message);
|
| - }
|
| -
|
| - EXPECT_EQ_WAIT(
|
| - kNumMessages,
|
| - initializing_client()->data_observer()->received_message_count(),
|
| - kMaxWaitMs);
|
| - EXPECT_EQ_WAIT(kNumMessages,
|
| - receiving_client()->data_observer()->received_message_count(),
|
| - kMaxWaitMs);
|
| -
|
| - // Sort and compare to make sure none of the messages were corrupted.
|
| - std::vector<std::string> initializing_client_received_messages =
|
| - initializing_client()->data_observer()->messages();
|
| - std::vector<std::string> receiving_client_received_messages =
|
| - receiving_client()->data_observer()->messages();
|
| - std::sort(sent_messages.begin(), sent_messages.end());
|
| - std::sort(initializing_client_received_messages.begin(),
|
| - initializing_client_received_messages.end());
|
| - std::sort(receiving_client_received_messages.begin(),
|
| - receiving_client_received_messages.end());
|
| - EXPECT_EQ(sent_messages, initializing_client_received_messages);
|
| - EXPECT_EQ(sent_messages, receiving_client_received_messages);
|
| -
|
| - receiving_client()->data_channel()->Close();
|
| - EXPECT_TRUE_WAIT(!initializing_client()->data_observer()->IsOpen(),
|
| - kMaxWaitMs);
|
| - EXPECT_TRUE_WAIT(!receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
|
| -}
|
| -
|
| -// This test sets up a call between two parties and creates a data channel.
|
| -// The test tests that received data is buffered unless an observer has been
|
| -// registered.
|
| -// Rtp data channels can receive data before the underlying
|
| -// transport has detected that a channel is writable and thus data can be
|
| -// received before the data channel state changes to open. That is hard to test
|
| -// but the same buffering is used in that case.
|
| -TEST_F(P2PTestConductor, RegisterDataChannelObserver) {
|
| - FakeConstraints setup_constraints;
|
| - setup_constraints.SetAllowRtpDataChannels();
|
| - ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
|
| - initializing_client()->CreateDataChannel();
|
| - initializing_client()->Negotiate();
|
| -
|
| - ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
|
| - ASSERT_TRUE(receiving_client()->data_channel() != nullptr);
|
| - EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
|
| - kMaxWaitMs);
|
| - EXPECT_EQ_WAIT(DataChannelInterface::kOpen,
|
| - receiving_client()->data_channel()->state(), kMaxWaitMs);
|
| -
|
| - // Unregister the existing observer.
|
| - receiving_client()->data_channel()->UnregisterObserver();
|
| -
|
| - std::string data = "hello world";
|
| - SendRtpData(initializing_client()->data_channel(), data);
|
| -
|
| - // Wait a while to allow the sent data to arrive before an observer is
|
| - // registered..
|
| - rtc::Thread::Current()->ProcessMessages(100);
|
| -
|
| - MockDataChannelObserver new_observer(receiving_client()->data_channel());
|
| - EXPECT_EQ_WAIT(data, new_observer.last_message(), kMaxWaitMs);
|
| -}
|
| -
|
| -// This test sets up a call between two parties with audio, video and but only
|
| -// the initiating client support data.
|
| -TEST_F(P2PTestConductor, LocalP2PTestReceiverDoesntSupportData) {
|
| - FakeConstraints setup_constraints_1;
|
| - setup_constraints_1.SetAllowRtpDataChannels();
|
| - // Must disable DTLS to make negotiation succeed.
|
| - setup_constraints_1.SetMandatory(
|
| - MediaConstraintsInterface::kEnableDtlsSrtp, false);
|
| - FakeConstraints setup_constraints_2;
|
| - setup_constraints_2.SetMandatory(
|
| - MediaConstraintsInterface::kEnableDtlsSrtp, false);
|
| - ASSERT_TRUE(CreateTestClients(&setup_constraints_1, &setup_constraints_2));
|
| - initializing_client()->CreateDataChannel();
|
| - LocalP2PTest();
|
| - EXPECT_TRUE(initializing_client()->data_channel() != nullptr);
|
| - EXPECT_FALSE(receiving_client()->data_channel());
|
| - EXPECT_FALSE(initializing_client()->data_observer()->IsOpen());
|
| -}
|
| -
|
| -// This test sets up a call between two parties with audio, video. When audio
|
| -// and video is setup and flowing and data channel is negotiated.
|
| -TEST_F(P2PTestConductor, AddDataChannelAfterRenegotiation) {
|
| - FakeConstraints setup_constraints;
|
| - setup_constraints.SetAllowRtpDataChannels();
|
| - ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
|
| - LocalP2PTest();
|
| - initializing_client()->CreateDataChannel();
|
| - // Send new offer and answer.
|
| - initializing_client()->Negotiate();
|
| - ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
|
| - ASSERT_TRUE(receiving_client()->data_channel() != nullptr);
|
| - EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
|
| - kMaxWaitMs);
|
| - EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(),
|
| - kMaxWaitMs);
|
| -}
|
| -
|
| -// This test sets up a Jsep call with SCTP DataChannel and verifies the
|
| -// negotiation is completed without error.
|
| -#ifdef HAVE_SCTP
|
| -TEST_F(P2PTestConductor, CreateOfferWithSctpDataChannel) {
|
| - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
|
| - FakeConstraints constraints;
|
| - constraints.SetMandatory(
|
| - MediaConstraintsInterface::kEnableDtlsSrtp, true);
|
| - ASSERT_TRUE(CreateTestClients(&constraints, &constraints));
|
| - initializing_client()->CreateDataChannel();
|
| - initializing_client()->Negotiate(false, false);
|
| -}
|
| -#endif
|
| -
|
| -// This test sets up a call between two parties with audio, and video.
|
| -// During the call, the initializing side restart ice and the test verifies that
|
| -// new ice candidates are generated and audio and video still can flow.
|
| -TEST_F(P2PTestConductor, IceRestart) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| -
|
| - // Negotiate and wait for ice completion and make sure audio and video plays.
|
| - LocalP2PTest();
|
| -
|
| - // Create a SDP string of the first audio candidate for both clients.
|
| - const webrtc::IceCandidateCollection* audio_candidates_initiator =
|
| - initializing_client()->pc()->local_description()->candidates(0);
|
| - const webrtc::IceCandidateCollection* audio_candidates_receiver =
|
| - receiving_client()->pc()->local_description()->candidates(0);
|
| - ASSERT_GT(audio_candidates_initiator->count(), 0u);
|
| - ASSERT_GT(audio_candidates_receiver->count(), 0u);
|
| - std::string initiator_candidate;
|
| - EXPECT_TRUE(
|
| - audio_candidates_initiator->at(0)->ToString(&initiator_candidate));
|
| - std::string receiver_candidate;
|
| - EXPECT_TRUE(audio_candidates_receiver->at(0)->ToString(&receiver_candidate));
|
| -
|
| - // Restart ice on the initializing client.
|
| - receiving_client()->SetExpectIceRestart(true);
|
| - initializing_client()->IceRestart();
|
| -
|
| - // Negotiate and wait for ice completion again and make sure audio and video
|
| - // plays.
|
| - LocalP2PTest();
|
| -
|
| - // Create a SDP string of the first audio candidate for both clients again.
|
| - const webrtc::IceCandidateCollection* audio_candidates_initiator_restart =
|
| - initializing_client()->pc()->local_description()->candidates(0);
|
| - const webrtc::IceCandidateCollection* audio_candidates_reciever_restart =
|
| - receiving_client()->pc()->local_description()->candidates(0);
|
| - ASSERT_GT(audio_candidates_initiator_restart->count(), 0u);
|
| - ASSERT_GT(audio_candidates_reciever_restart->count(), 0u);
|
| - std::string initiator_candidate_restart;
|
| - EXPECT_TRUE(audio_candidates_initiator_restart->at(0)->ToString(
|
| - &initiator_candidate_restart));
|
| - std::string receiver_candidate_restart;
|
| - EXPECT_TRUE(audio_candidates_reciever_restart->at(0)->ToString(
|
| - &receiver_candidate_restart));
|
| -
|
| - // Verify that the first candidates in the local session descriptions has
|
| - // changed.
|
| - EXPECT_NE(initiator_candidate, initiator_candidate_restart);
|
| - EXPECT_NE(receiver_candidate, receiver_candidate_restart);
|
| -}
|
| -
|
| -TEST_F(P2PTestConductor, IceRenominationDisabled) {
|
| - PeerConnectionInterface::RTCConfiguration config;
|
| - config.enable_ice_renomination = false;
|
| - ASSERT_TRUE(CreateTestClients(config, config));
|
| - LocalP2PTest();
|
| -
|
| - initializing_client()->VerifyLocalIceRenomination();
|
| - receiving_client()->VerifyLocalIceRenomination();
|
| - initializing_client()->VerifyRemoteIceRenomination();
|
| - receiving_client()->VerifyRemoteIceRenomination();
|
| -}
|
| -
|
| -TEST_F(P2PTestConductor, IceRenominationEnabled) {
|
| - PeerConnectionInterface::RTCConfiguration config;
|
| - config.enable_ice_renomination = true;
|
| - ASSERT_TRUE(CreateTestClients(config, config));
|
| - initializing_client()->SetExpectIceRenomination(true);
|
| - initializing_client()->SetExpectRemoteIceRenomination(true);
|
| - receiving_client()->SetExpectIceRenomination(true);
|
| - receiving_client()->SetExpectRemoteIceRenomination(true);
|
| - LocalP2PTest();
|
| -
|
| - initializing_client()->VerifyLocalIceRenomination();
|
| - receiving_client()->VerifyLocalIceRenomination();
|
| - initializing_client()->VerifyRemoteIceRenomination();
|
| - receiving_client()->VerifyRemoteIceRenomination();
|
| -}
|
| -
|
| -// This test sets up a call between two parties with audio, and video.
|
| -// It then renegotiates setting the video m-line to "port 0", then later
|
| -// renegotiates again, enabling video.
|
| -TEST_F(P2PTestConductor, LocalP2PTestVideoDisableEnable) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| -
|
| - // Do initial negotiation. Will result in video and audio sendonly m-lines.
|
| - receiving_client()->set_auto_add_stream(false);
|
| - initializing_client()->AddMediaStream(true, true);
|
| - initializing_client()->Negotiate();
|
| -
|
| - // Negotiate again, disabling the video m-line (receiving client will
|
| - // set port to 0 due to mandatory "OfferToReceiveVideo: false" constraint).
|
| - receiving_client()->SetReceiveVideo(false);
|
| - initializing_client()->Negotiate();
|
| -
|
| - // Enable video and do negotiation again, making sure video is received
|
| - // end-to-end.
|
| - receiving_client()->SetReceiveVideo(true);
|
| - receiving_client()->AddMediaStream(true, true);
|
| - LocalP2PTest();
|
| -}
|
| -
|
| -// This test sets up a Jsep call between two parties with external
|
| -// VideoDecoderFactory.
|
| -// TODO(holmer): Disabled due to sometimes crashing on buildbots.
|
| -// See issue webrtc/2378.
|
| -TEST_F(P2PTestConductor, DISABLED_LocalP2PTestWithVideoDecoderFactory) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - EnableVideoDecoderFactory();
|
| - LocalP2PTest();
|
| -}
|
| -
|
| -// This tests that if we negotiate after calling CreateSender but before we
|
| -// have a track, then set a track later, frames from the newly-set track are
|
| -// received end-to-end.
|
| -TEST_F(P2PTestConductor, EarlyWarmupTest) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - auto audio_sender =
|
| - initializing_client()->pc()->CreateSender("audio", "stream_id");
|
| - auto video_sender =
|
| - initializing_client()->pc()->CreateSender("video", "stream_id");
|
| - initializing_client()->Negotiate();
|
| - // Wait for ICE connection to complete, without any tracks.
|
| - // Note that the receiving client WILL (in HandleIncomingOffer) create
|
| - // tracks, so it's only the initiator here that's doing early warmup.
|
| - ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
|
| - VerifySessionDescriptions();
|
| - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
|
| - initializing_client()->ice_connection_state(),
|
| - kMaxWaitForFramesMs);
|
| - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
|
| - receiving_client()->ice_connection_state(),
|
| - kMaxWaitForFramesMs);
|
| - // Now set the tracks, and expect frames to immediately start flowing.
|
| - EXPECT_TRUE(
|
| - audio_sender->SetTrack(initializing_client()->CreateLocalAudioTrack("")));
|
| - EXPECT_TRUE(
|
| - video_sender->SetTrack(initializing_client()->CreateLocalVideoTrack("")));
|
| - EXPECT_TRUE_WAIT(FramesHaveArrived(kEndAudioFrameCount, kEndVideoFrameCount),
|
| - kMaxWaitForFramesMs);
|
| -}
|
| -
|
| -#ifdef HAVE_QUIC
|
| -// This test sets up a call between two parties using QUIC instead of DTLS for
|
| -// audio and video, and a QUIC data channel.
|
| -TEST_F(P2PTestConductor, LocalP2PTestQuicDataChannel) {
|
| - PeerConnectionInterface::RTCConfiguration quic_config;
|
| - quic_config.enable_quic = true;
|
| - ASSERT_TRUE(CreateTestClients(quic_config, quic_config));
|
| - webrtc::DataChannelInit init;
|
| - init.ordered = false;
|
| - init.reliable = true;
|
| - init.id = 1;
|
| - initializing_client()->CreateDataChannel(&init);
|
| - receiving_client()->CreateDataChannel(&init);
|
| - LocalP2PTest();
|
| - ASSERT_NE(nullptr, initializing_client()->data_channel());
|
| - ASSERT_NE(nullptr, receiving_client()->data_channel());
|
| - EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
|
| - kMaxWaitMs);
|
| - EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
|
| -
|
| - std::string data = "hello world";
|
| -
|
| - initializing_client()->data_channel()->Send(DataBuffer(data));
|
| - EXPECT_EQ_WAIT(data, receiving_client()->data_observer()->last_message(),
|
| - kMaxWaitMs);
|
| -
|
| - receiving_client()->data_channel()->Send(DataBuffer(data));
|
| - EXPECT_EQ_WAIT(data, initializing_client()->data_observer()->last_message(),
|
| - kMaxWaitMs);
|
| -}
|
| -
|
| -// Tests that negotiation of QUIC data channels is completed without error.
|
| -TEST_F(P2PTestConductor, NegotiateQuicDataChannel) {
|
| - PeerConnectionInterface::RTCConfiguration quic_config;
|
| - quic_config.enable_quic = true;
|
| - ASSERT_TRUE(CreateTestClients(quic_config, quic_config));
|
| - FakeConstraints constraints;
|
| - constraints.SetMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, true);
|
| - ASSERT_TRUE(CreateTestClients(&constraints, &constraints));
|
| - webrtc::DataChannelInit init;
|
| - init.ordered = false;
|
| - init.reliable = true;
|
| - init.id = 1;
|
| - initializing_client()->CreateDataChannel(&init);
|
| - initializing_client()->Negotiate(false, false);
|
| -}
|
| -
|
| -// This test sets up a JSEP call using QUIC. The callee only receives video.
|
| -TEST_F(P2PTestConductor, LocalP2PTestVideoOnlyWithQuic) {
|
| - PeerConnectionInterface::RTCConfiguration quic_config;
|
| - quic_config.enable_quic = true;
|
| - ASSERT_TRUE(CreateTestClients(quic_config, quic_config));
|
| - receiving_client()->SetReceiveAudioVideo(false, true);
|
| - LocalP2PTest();
|
| -}
|
| -
|
| -// This test sets up a JSEP call using QUIC. The callee only receives audio.
|
| -TEST_F(P2PTestConductor, LocalP2PTestAudioOnlyWithQuic) {
|
| - PeerConnectionInterface::RTCConfiguration quic_config;
|
| - quic_config.enable_quic = true;
|
| - ASSERT_TRUE(CreateTestClients(quic_config, quic_config));
|
| - receiving_client()->SetReceiveAudioVideo(true, false);
|
| - LocalP2PTest();
|
| -}
|
| -
|
| -// This test sets up a JSEP call using QUIC. The callee rejects both audio and
|
| -// video.
|
| -TEST_F(P2PTestConductor, LocalP2PTestNoVideoAudioWithQuic) {
|
| - PeerConnectionInterface::RTCConfiguration quic_config;
|
| - quic_config.enable_quic = true;
|
| - ASSERT_TRUE(CreateTestClients(quic_config, quic_config));
|
| - receiving_client()->SetReceiveAudioVideo(false, false);
|
| - LocalP2PTest();
|
| -}
|
| -
|
| -#endif // HAVE_QUIC
|
| -
|
| -TEST_F(P2PTestConductor, ForwardVideoOnlyStream) {
|
| - ASSERT_TRUE(CreateTestClients());
|
| - // One-way stream
|
| - receiving_client()->set_auto_add_stream(false);
|
| - // Video only, audio forwarding not expected to work.
|
| - initializing_client()->AddMediaStream(false, true);
|
| - initializing_client()->Negotiate();
|
| -
|
| - ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
|
| - VerifySessionDescriptions();
|
| -
|
| - ASSERT_TRUE(initializing_client()->can_receive_video());
|
| - ASSERT_TRUE(receiving_client()->can_receive_video());
|
| -
|
| - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
|
| - initializing_client()->ice_connection_state(),
|
| - kMaxWaitForFramesMs);
|
| - EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
|
| - receiving_client()->ice_connection_state(),
|
| - kMaxWaitForFramesMs);
|
| -
|
| - ASSERT_TRUE(receiving_client()->remote_streams()->count() == 1);
|
| -
|
| - // Echo the stream back.
|
| - receiving_client()->pc()->AddStream(
|
| - receiving_client()->remote_streams()->at(0));
|
| - receiving_client()->Negotiate();
|
| -
|
| - EXPECT_TRUE_WAIT(
|
| - initializing_client()->VideoFramesReceivedCheck(kEndVideoFrameCount),
|
| - kMaxWaitForFramesMs);
|
| -}
|
| -
|
| -// Test that we achieve the expected end-to-end connection time, using a
|
| -// fake clock and simulated latency on the media and signaling paths.
|
| -// We use a TURN<->TURN connection because this is usually the quickest to
|
| -// set up initially, especially when we're confident the connection will work
|
| -// and can start sending media before we get a STUN response.
|
| -//
|
| -// With various optimizations enabled, here are the network delays we expect to
|
| -// be on the critical path:
|
| -// 1. 2 signaling trips: Signaling offer and offerer's TURN candidate, then
|
| -// signaling answer (with DTLS fingerprint).
|
| -// 2. 9 media hops: Rest of the DTLS handshake. 3 hops in each direction when
|
| -// using TURN<->TURN pair, and DTLS exchange is 4 packets,
|
| -// the first of which should have arrived before the answer.
|
| -TEST_F(P2PTestConductor, EndToEndConnectionTimeWithTurnTurnPair) {
|
| - rtc::ScopedFakeClock fake_clock;
|
| - // Some things use a time of "0" as a special value, so we need to start out
|
| - // the fake clock at a nonzero time.
|
| - // TODO(deadbeef): Fix this.
|
| - fake_clock.AdvanceTime(rtc::TimeDelta::FromSeconds(1));
|
| -
|
| - static constexpr int media_hop_delay_ms = 50;
|
| - static constexpr int signaling_trip_delay_ms = 500;
|
| - // For explanation of these values, see comment above.
|
| - static constexpr int required_media_hops = 9;
|
| - static constexpr int required_signaling_trips = 2;
|
| - // For internal delays (such as posting an event asychronously).
|
| - static constexpr int allowed_internal_delay_ms = 20;
|
| - static constexpr int total_connection_time_ms =
|
| - media_hop_delay_ms * required_media_hops +
|
| - signaling_trip_delay_ms * required_signaling_trips +
|
| - allowed_internal_delay_ms;
|
| -
|
| - static const rtc::SocketAddress turn_server_1_internal_address{"88.88.88.0",
|
| - 3478};
|
| - static const rtc::SocketAddress turn_server_1_external_address{"88.88.88.1",
|
| - 0};
|
| - static const rtc::SocketAddress turn_server_2_internal_address{"99.99.99.0",
|
| - 3478};
|
| - static const rtc::SocketAddress turn_server_2_external_address{"99.99.99.1",
|
| - 0};
|
| - cricket::TestTurnServer turn_server_1(network_thread(),
|
| - turn_server_1_internal_address,
|
| - turn_server_1_external_address);
|
| - cricket::TestTurnServer turn_server_2(network_thread(),
|
| - turn_server_2_internal_address,
|
| - turn_server_2_external_address);
|
| - // Bypass permission check on received packets so media can be sent before
|
| - // the candidate is signaled.
|
| - turn_server_1.set_enable_permission_checks(false);
|
| - turn_server_2.set_enable_permission_checks(false);
|
| -
|
| - PeerConnectionInterface::RTCConfiguration client_1_config;
|
| - webrtc::PeerConnectionInterface::IceServer ice_server_1;
|
| - ice_server_1.urls.push_back("turn:88.88.88.0:3478");
|
| - ice_server_1.username = "test";
|
| - ice_server_1.password = "test";
|
| - client_1_config.servers.push_back(ice_server_1);
|
| - client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
|
| - client_1_config.presume_writable_when_fully_relayed = true;
|
| -
|
| - PeerConnectionInterface::RTCConfiguration client_2_config;
|
| - webrtc::PeerConnectionInterface::IceServer ice_server_2;
|
| - ice_server_2.urls.push_back("turn:99.99.99.0:3478");
|
| - ice_server_2.username = "test";
|
| - ice_server_2.password = "test";
|
| - client_2_config.servers.push_back(ice_server_2);
|
| - client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
|
| - client_2_config.presume_writable_when_fully_relayed = true;
|
| -
|
| - ASSERT_TRUE(CreateTestClients(client_1_config, client_2_config));
|
| - // Set up the simulated delays.
|
| - SetSignalingDelayMs(signaling_trip_delay_ms);
|
| - virtual_socket_server()->set_delay_mean(media_hop_delay_ms);
|
| - virtual_socket_server()->UpdateDelayDistribution();
|
| -
|
| - initializing_client()->SetOfferToReceiveAudioVideo(true, true);
|
| - initializing_client()->Negotiate();
|
| - // TODO(deadbeef): kIceConnectionConnected currently means both ICE and DTLS
|
| - // are connected. This is an important distinction. Once we have separate ICE
|
| - // and DTLS state, this check needs to use the DTLS state.
|
| - EXPECT_TRUE_SIMULATED_WAIT(
|
| - (receiving_client()->ice_connection_state() ==
|
| - webrtc::PeerConnectionInterface::kIceConnectionConnected ||
|
| - receiving_client()->ice_connection_state() ==
|
| - webrtc::PeerConnectionInterface::kIceConnectionCompleted) &&
|
| - (initializing_client()->ice_connection_state() ==
|
| - webrtc::PeerConnectionInterface::kIceConnectionConnected ||
|
| - initializing_client()->ice_connection_state() ==
|
| - webrtc::PeerConnectionInterface::kIceConnectionCompleted),
|
| - total_connection_time_ms, fake_clock);
|
| - // Need to free the clients here since they're using things we created on
|
| - // the stack.
|
| - delete set_initializing_client(nullptr);
|
| - delete set_receiving_client(nullptr);
|
| -}
|
| -
|
| -class IceServerParsingTest : public testing::Test {
|
| - public:
|
| - // Convenience for parsing a single URL.
|
| - bool ParseUrl(const std::string& url) {
|
| - return ParseUrl(url, std::string(), std::string());
|
| - }
|
| -
|
| - bool ParseUrl(const std::string& url,
|
| - const std::string& username,
|
| - const std::string& password) {
|
| - PeerConnectionInterface::IceServers servers;
|
| - PeerConnectionInterface::IceServer server;
|
| - server.urls.push_back(url);
|
| - server.username = username;
|
| - server.password = password;
|
| - servers.push_back(server);
|
| - return webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_);
|
| - }
|
| -
|
| - protected:
|
| - cricket::ServerAddresses stun_servers_;
|
| - std::vector<cricket::RelayServerConfig> turn_servers_;
|
| -};
|
| -
|
| -// Make sure all STUN/TURN prefixes are parsed correctly.
|
| -TEST_F(IceServerParsingTest, ParseStunPrefixes) {
|
| - EXPECT_TRUE(ParseUrl("stun:hostname"));
|
| - EXPECT_EQ(1U, stun_servers_.size());
|
| - EXPECT_EQ(0U, turn_servers_.size());
|
| - stun_servers_.clear();
|
| -
|
| - EXPECT_TRUE(ParseUrl("stuns:hostname"));
|
| - EXPECT_EQ(1U, stun_servers_.size());
|
| - EXPECT_EQ(0U, turn_servers_.size());
|
| - stun_servers_.clear();
|
| -
|
| - EXPECT_TRUE(ParseUrl("turn:hostname"));
|
| - EXPECT_EQ(0U, stun_servers_.size());
|
| - EXPECT_EQ(1U, turn_servers_.size());
|
| - EXPECT_FALSE(turn_servers_[0].ports[0].secure);
|
| - turn_servers_.clear();
|
| -
|
| - EXPECT_TRUE(ParseUrl("turns:hostname"));
|
| - EXPECT_EQ(0U, stun_servers_.size());
|
| - EXPECT_EQ(1U, turn_servers_.size());
|
| - EXPECT_TRUE(turn_servers_[0].ports[0].secure);
|
| - turn_servers_.clear();
|
| -
|
| - // invalid prefixes
|
| - EXPECT_FALSE(ParseUrl("stunn:hostname"));
|
| - EXPECT_FALSE(ParseUrl(":hostname"));
|
| - EXPECT_FALSE(ParseUrl(":"));
|
| - EXPECT_FALSE(ParseUrl(""));
|
| -}
|
| -
|
| -TEST_F(IceServerParsingTest, VerifyDefaults) {
|
| - // TURNS defaults
|
| - EXPECT_TRUE(ParseUrl("turns:hostname"));
|
| - EXPECT_EQ(1U, turn_servers_.size());
|
| - EXPECT_EQ(5349, turn_servers_[0].ports[0].address.port());
|
| - EXPECT_EQ(cricket::PROTO_TCP, turn_servers_[0].ports[0].proto);
|
| - turn_servers_.clear();
|
| -
|
| - // TURN defaults
|
| - EXPECT_TRUE(ParseUrl("turn:hostname"));
|
| - EXPECT_EQ(1U, turn_servers_.size());
|
| - EXPECT_EQ(3478, turn_servers_[0].ports[0].address.port());
|
| - EXPECT_EQ(cricket::PROTO_UDP, turn_servers_[0].ports[0].proto);
|
| - turn_servers_.clear();
|
| -
|
| - // STUN defaults
|
| - EXPECT_TRUE(ParseUrl("stun:hostname"));
|
| - EXPECT_EQ(1U, stun_servers_.size());
|
| - EXPECT_EQ(3478, stun_servers_.begin()->port());
|
| - stun_servers_.clear();
|
| -}
|
| -
|
| -// Check that the 6 combinations of IPv4/IPv6/hostname and with/without port
|
| -// can be parsed correctly.
|
| -TEST_F(IceServerParsingTest, ParseHostnameAndPort) {
|
| - EXPECT_TRUE(ParseUrl("stun:1.2.3.4:1234"));
|
| - EXPECT_EQ(1U, stun_servers_.size());
|
| - EXPECT_EQ("1.2.3.4", stun_servers_.begin()->hostname());
|
| - EXPECT_EQ(1234, stun_servers_.begin()->port());
|
| - stun_servers_.clear();
|
| -
|
| - EXPECT_TRUE(ParseUrl("stun:[1:2:3:4:5:6:7:8]:4321"));
|
| - EXPECT_EQ(1U, stun_servers_.size());
|
| - EXPECT_EQ("1:2:3:4:5:6:7:8", stun_servers_.begin()->hostname());
|
| - EXPECT_EQ(4321, stun_servers_.begin()->port());
|
| - stun_servers_.clear();
|
| -
|
| - EXPECT_TRUE(ParseUrl("stun:hostname:9999"));
|
| - EXPECT_EQ(1U, stun_servers_.size());
|
| - EXPECT_EQ("hostname", stun_servers_.begin()->hostname());
|
| - EXPECT_EQ(9999, stun_servers_.begin()->port());
|
| - stun_servers_.clear();
|
| -
|
| - EXPECT_TRUE(ParseUrl("stun:1.2.3.4"));
|
| - EXPECT_EQ(1U, stun_servers_.size());
|
| - EXPECT_EQ("1.2.3.4", stun_servers_.begin()->hostname());
|
| - EXPECT_EQ(3478, stun_servers_.begin()->port());
|
| - stun_servers_.clear();
|
| -
|
| - EXPECT_TRUE(ParseUrl("stun:[1:2:3:4:5:6:7:8]"));
|
| - EXPECT_EQ(1U, stun_servers_.size());
|
| - EXPECT_EQ("1:2:3:4:5:6:7:8", stun_servers_.begin()->hostname());
|
| - EXPECT_EQ(3478, stun_servers_.begin()->port());
|
| - stun_servers_.clear();
|
| -
|
| - EXPECT_TRUE(ParseUrl("stun:hostname"));
|
| - EXPECT_EQ(1U, stun_servers_.size());
|
| - EXPECT_EQ("hostname", stun_servers_.begin()->hostname());
|
| - EXPECT_EQ(3478, stun_servers_.begin()->port());
|
| - stun_servers_.clear();
|
| -
|
| - // Try some invalid hostname:port strings.
|
| - EXPECT_FALSE(ParseUrl("stun:hostname:99a99"));
|
| - EXPECT_FALSE(ParseUrl("stun:hostname:-1"));
|
| - EXPECT_FALSE(ParseUrl("stun:hostname:port:more"));
|
| - EXPECT_FALSE(ParseUrl("stun:hostname:port more"));
|
| - EXPECT_FALSE(ParseUrl("stun:hostname:"));
|
| - EXPECT_FALSE(ParseUrl("stun:[1:2:3:4:5:6:7:8]junk:1000"));
|
| - EXPECT_FALSE(ParseUrl("stun::5555"));
|
| - EXPECT_FALSE(ParseUrl("stun:"));
|
| -}
|
| -
|
| -// Test parsing the "?transport=xxx" part of the URL.
|
| -TEST_F(IceServerParsingTest, ParseTransport) {
|
| - EXPECT_TRUE(ParseUrl("turn:hostname:1234?transport=tcp"));
|
| - EXPECT_EQ(1U, turn_servers_.size());
|
| - EXPECT_EQ(cricket::PROTO_TCP, turn_servers_[0].ports[0].proto);
|
| - turn_servers_.clear();
|
| -
|
| - EXPECT_TRUE(ParseUrl("turn:hostname?transport=udp"));
|
| - EXPECT_EQ(1U, turn_servers_.size());
|
| - EXPECT_EQ(cricket::PROTO_UDP, turn_servers_[0].ports[0].proto);
|
| - turn_servers_.clear();
|
| -
|
| - EXPECT_FALSE(ParseUrl("turn:hostname?transport=invalid"));
|
| -}
|
| -
|
| -// Test parsing ICE username contained in URL.
|
| -TEST_F(IceServerParsingTest, ParseUsername) {
|
| - EXPECT_TRUE(ParseUrl("turn:user@hostname"));
|
| - EXPECT_EQ(1U, turn_servers_.size());
|
| - EXPECT_EQ("user", turn_servers_[0].credentials.username);
|
| - turn_servers_.clear();
|
| -
|
| - EXPECT_FALSE(ParseUrl("turn:@hostname"));
|
| - EXPECT_FALSE(ParseUrl("turn:username@"));
|
| - EXPECT_FALSE(ParseUrl("turn:@"));
|
| - EXPECT_FALSE(ParseUrl("turn:user@name@hostname"));
|
| -}
|
| -
|
| -// Test that username and password from IceServer is copied into the resulting
|
| -// RelayServerConfig.
|
| -TEST_F(IceServerParsingTest, CopyUsernameAndPasswordFromIceServer) {
|
| - EXPECT_TRUE(ParseUrl("turn:hostname", "username", "password"));
|
| - EXPECT_EQ(1U, turn_servers_.size());
|
| - EXPECT_EQ("username", turn_servers_[0].credentials.username);
|
| - EXPECT_EQ("password", turn_servers_[0].credentials.password);
|
| -}
|
| -
|
| -// Ensure that if a server has multiple URLs, each one is parsed.
|
| -TEST_F(IceServerParsingTest, ParseMultipleUrls) {
|
| - PeerConnectionInterface::IceServers servers;
|
| - PeerConnectionInterface::IceServer server;
|
| - server.urls.push_back("stun:hostname");
|
| - server.urls.push_back("turn:hostname");
|
| - servers.push_back(server);
|
| - EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
|
| - EXPECT_EQ(1U, stun_servers_.size());
|
| - EXPECT_EQ(1U, turn_servers_.size());
|
| -}
|
| -
|
| -// Ensure that TURN servers are given unique priorities,
|
| -// so that their resulting candidates have unique priorities.
|
| -TEST_F(IceServerParsingTest, TurnServerPrioritiesUnique) {
|
| - PeerConnectionInterface::IceServers servers;
|
| - PeerConnectionInterface::IceServer server;
|
| - server.urls.push_back("turn:hostname");
|
| - server.urls.push_back("turn:hostname2");
|
| - servers.push_back(server);
|
| - EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
|
| - EXPECT_EQ(2U, turn_servers_.size());
|
| - EXPECT_NE(turn_servers_[0].priority, turn_servers_[1].priority);
|
| -}
|
| -
|
| -#endif // if !defined(THREAD_SANITIZER)
|
| -
|
| -} // namespace
|
|
|
Chromium Code Reviews
Issue 2514883002:
Create //webrtc/api:libjingle_peerconnection_api + refactorings. (Closed)
