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webrtc / src / 5bb91b08b9b2e89f0fc0363b5eaa0aa8492ea998 / . / talk / app / webrtc / peerconnection_unittest.cc
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/*
* libjingle
* Copyright 2012 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include <stdio.h>
#include <algorithm>
#include <list>
#include <map>
#include <vector>
#include "talk/app/webrtc/dtmfsender.h"
#include "talk/app/webrtc/fakeportallocatorfactory.h"
#include "talk/app/webrtc/localaudiosource.h"
#include "talk/app/webrtc/mediastreaminterface.h"
#include "talk/app/webrtc/peerconnectionfactory.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "talk/app/webrtc/test/fakeaudiocapturemodule.h"
#include "talk/app/webrtc/test/fakeconstraints.h"
#include "talk/app/webrtc/test/fakedtlsidentityservice.h"
#include "talk/app/webrtc/test/fakeperiodicvideocapturer.h"
#include "talk/app/webrtc/test/fakevideotrackrenderer.h"
#include "talk/app/webrtc/test/mockpeerconnectionobservers.h"
#include "talk/app/webrtc/videosourceinterface.h"
#include "talk/media/webrtc/fakewebrtcvideoengine.h"
#include "talk/session/media/mediasession.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/physicalsocketserver.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/ssladapter.h"
#include "webrtc/base/sslstreamadapter.h"
#include "webrtc/base/thread.h"
#include "webrtc/base/virtualsocketserver.h"
#include "webrtc/p2p/base/constants.h"
#include "webrtc/p2p/base/sessiondescription.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::MediaStreamTrackInterface;
using webrtc::MockCreateSessionDescriptionObserver;
using webrtc::MockDataChannelObserver;
using webrtc::MockSetSessionDescriptionObserver;
using webrtc::MockStatsObserver;
using webrtc::PeerConnectionInterface;
using webrtc::SessionDescriptionInterface;
using webrtc::StreamCollectionInterface;
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 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 char kDefaultSrtpCipher[] = "AES_CM_128_HMAC_SHA1_32";
#endif
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));
}
}
class SignalingMessageReceiver {
public:
protected:
SignalingMessageReceiver() {}
virtual ~SignalingMessageReceiver() {}
};
class JsepMessageReceiver : public 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:
JsepMessageReceiver() {}
virtual ~JsepMessageReceiver() {}
};
template <typename MessageReceiver>
class PeerConnectionTestClientBase
: public webrtc::PeerConnectionObserver,
public MessageReceiver {
public:
~PeerConnectionTestClientBase() {
while (!fake_video_renderers_.empty()) {
RenderMap::iterator it = fake_video_renderers_.begin();
delete it->second;
fake_video_renderers_.erase(it);
}
}
virtual void Negotiate() = 0;
virtual void Negotiate(bool audio, bool video) = 0;
virtual 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>(peer_connection_->local_streams()->count()));
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
peer_connection_factory_->CreateLocalMediaStream(stream_label);
if (audio && can_receive_audio()) {
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;
rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
peer_connection_factory_->CreateAudioTrack(label, source));
stream->AddTrack(audio_track);
}
if (video && can_receive_video()) {
stream->AddTrack(CreateLocalVideoTrack(stream_label));
}
EXPECT_TRUE(peer_connection_->AddStream(stream));
}
size_t NumberOfLocalMediaStreams() {
return peer_connection_->local_streams()->count();
}
bool SessionActive() {
return peer_connection_->signaling_state() ==
webrtc::PeerConnectionInterface::kStable;
}
void set_signaling_message_receiver(
MessageReceiver* signaling_message_receiver) {
signaling_message_receiver_ = signaling_message_receiver;
}
void EnableVideoDecoderFactory() {
video_decoder_factory_enabled_ = true;
fake_video_decoder_factory_->AddSupportedVideoCodecType(
webrtc::kVideoCodecVP8);
}
bool AudioFramesReceivedCheck(int number_of_frames) const {
return number_of_frames <= 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;
}
for (std::vector<FakeWebRtcVideoDecoder*>::const_iterator
it = decoders.begin(); it != decoders.end(); ++it) {
if (number_of_frames > (*it)->GetNumFramesReceived()) {
return false;
}
}
return true;
} else {
if (fake_video_renderers_.empty()) {
return number_of_frames <= 0;
}
for (RenderMap::const_iterator it = fake_video_renderers_.begin();
it != fake_video_renderers_.end(); ++it) {
if (number_of_frames > it->second->num_rendered_frames()) {
return false;
}
}
return true;
}
}
// Verify the CreateDtmfSender interface
void VerifyDtmf() {
rtc::scoped_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(NULL) == NULL);
rtc::scoped_refptr<webrtc::AudioTrackInterface> non_localtrack(
peer_connection_factory_->CreateAudioTrack("dummy_track",
NULL));
EXPECT_TRUE(peer_connection_->CreateDtmfSender(non_localtrack) == NULL);
// 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() != NULL);
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() != NULL);
ASSERT_TRUE(peer_connection_->local_description() != NULL);
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 SetExpectIceRestart(bool expect_restart) {
expect_ice_restart_ = expect_restart;
}
bool ExpectIceRestart() const { return expect_ice_restart_; }
void VerifyLocalIceUfragAndPassword() {
ASSERT_TRUE(peer_connection_->local_description() != NULL);
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);
}
}
}
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);
return observer->AudioOutputLevel();
}
int GetAudioInputLevelStats() {
rtc::scoped_refptr<MockStatsObserver>
observer(new rtc::RefCountedObject<MockStatsObserver>());
EXPECT_TRUE(peer_connection_->GetStats(
observer, NULL, PeerConnectionInterface::kStatsOutputLevelStandard));
EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
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);
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);
return observer->BytesSent();
}
int GetAvailableReceivedBandwidthStats() {
rtc::scoped_refptr<MockStatsObserver>
observer(new rtc::RefCountedObject<MockStatsObserver>());
EXPECT_TRUE(peer_connection_->GetStats(
observer, NULL, PeerConnectionInterface::kStatsOutputLevelStandard));
EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
int bw = observer->AvailableReceiveBandwidth();
return bw;
}
std::string GetDtlsCipherStats() {
rtc::scoped_refptr<MockStatsObserver>
observer(new rtc::RefCountedObject<MockStatsObserver>());
EXPECT_TRUE(peer_connection_->GetStats(
observer, NULL, PeerConnectionInterface::kStatsOutputLevelStandard));
EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
return observer->DtlsCipher();
}
std::string GetSrtpCipherStats() {
rtc::scoped_refptr<MockStatsObserver>
observer(new rtc::RefCountedObject<MockStatsObserver>());
EXPECT_TRUE(peer_connection_->GetStats(
observer, NULL, PeerConnectionInterface::kStatsOutputLevelStandard));
EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
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();
}
size_t number_of_remote_streams() {
if (!pc())
return 0;
return pc()->remote_streams()->count();
}
StreamCollectionInterface* remote_streams() {
if (!pc()) {
ADD_FAILURE();
return NULL;
}
return pc()->remote_streams();
}
StreamCollectionInterface* local_streams() {
if (!pc()) {
ADD_FAILURE();
return NULL;
}
return pc()->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();
}
// PeerConnectionObserver callbacks.
virtual void OnMessage(const std::string&) {}
virtual void OnSignalingMessage(const std::string& /*msg*/) {}
virtual void OnSignalingChange(
webrtc::PeerConnectionInterface::SignalingState new_state) {
EXPECT_EQ(peer_connection_->signaling_state(), new_state);
}
virtual void OnAddStream(webrtc::MediaStreamInterface* media_stream) {
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] = new webrtc::FakeVideoTrackRenderer(
media_stream->GetVideoTracks()[i]);
}
}
virtual void OnRemoveStream(webrtc::MediaStreamInterface* media_stream) {}
virtual void OnRenegotiationNeeded() {}
virtual void OnIceConnectionChange(
webrtc::PeerConnectionInterface::IceConnectionState new_state) {
EXPECT_EQ(peer_connection_->ice_connection_state(), new_state);
}
virtual void OnIceGatheringChange(
webrtc::PeerConnectionInterface::IceGatheringState new_state) {
EXPECT_EQ(peer_connection_->ice_gathering_state(), new_state);
}
virtual void OnIceCandidate(
const webrtc::IceCandidateInterface* /*candidate*/) {}
webrtc::PeerConnectionInterface* pc() {
return peer_connection_.get();
}
void StopVideoCapturers() {
for (std::vector<cricket::VideoCapturer*>::iterator it =
video_capturers_.begin(); it != video_capturers_.end(); ++it) {
(*it)->Stop();
}
}
protected:
explicit PeerConnectionTestClientBase(const std::string& id)
: id_(id),
expect_ice_restart_(false),
fake_video_decoder_factory_(NULL),
fake_video_encoder_factory_(NULL),
video_decoder_factory_enabled_(false),
signaling_message_receiver_(NULL) {
}
bool Init(const MediaConstraintsInterface* constraints) {
EXPECT_TRUE(!peer_connection_);
EXPECT_TRUE(!peer_connection_factory_);
allocator_factory_ = webrtc::FakePortAllocatorFactory::Create();
if (!allocator_factory_) {
return false;
}
fake_audio_capture_module_ = FakeAudioCaptureModule::Create(
rtc::Thread::Current());
if (fake_audio_capture_module_ == NULL) {
return false;
}
fake_video_decoder_factory_ = new FakeWebRtcVideoDecoderFactory();
fake_video_encoder_factory_ = new FakeWebRtcVideoEncoderFactory();
peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
rtc::Thread::Current(), rtc::Thread::Current(),
fake_audio_capture_module_, fake_video_encoder_factory_,
fake_video_decoder_factory_);
if (!peer_connection_factory_) {
return false;
}
peer_connection_ = CreatePeerConnection(allocator_factory_.get(),
constraints);
return peer_connection_.get() != NULL;
}
virtual rtc::scoped_refptr<webrtc::PeerConnectionInterface>
CreatePeerConnection(webrtc::PortAllocatorFactoryInterface* factory,
const MediaConstraintsInterface* constraints) = 0;
MessageReceiver* signaling_message_receiver() {
return signaling_message_receiver_;
}
webrtc::PeerConnectionFactoryInterface* peer_connection_factory() {
return peer_connection_factory_.get();
}
virtual bool can_receive_audio() = 0;
virtual bool can_receive_video() = 0;
const std::string& id() const { return id_; }
private:
class DummyDtmfObserver : public DtmfSenderObserverInterface {
public:
DummyDtmfObserver() : completed_(false) {}
// Implements DtmfSenderObserverInterface.
void OnToneChange(const std::string& tone) {
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_;
};
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();
video_capturers_.push_back(fake_capturer);
rtc::scoped_refptr<webrtc::VideoSourceInterface> source =
peer_connection_factory_->CreateVideoSource(
fake_capturer, &source_constraints);
std::string label = stream_label + kVideoTrackLabelBase;
return peer_connection_factory_->CreateVideoTrack(label, source);
}
std::string id_;
rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface>
allocator_factory_;
rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
peer_connection_factory_;
typedef std::pair<std::string, std::string> IceUfragPwdPair;
std::map<int, IceUfragPwdPair> ice_ufrag_pwd_;
bool expect_ice_restart_;
// Needed to keep track of number of frames send.
rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
// Needed to keep track of number of frames received.
typedef std::map<std::string, webrtc::FakeVideoTrackRenderer*> RenderMap;
RenderMap fake_video_renderers_;
// Needed to keep track of number of frames received when external decoder
// used.
FakeWebRtcVideoDecoderFactory* fake_video_decoder_factory_;
FakeWebRtcVideoEncoderFactory* fake_video_encoder_factory_;
bool video_decoder_factory_enabled_;
webrtc::FakeConstraints video_constraints_;
// For remote peer communication.
MessageReceiver* signaling_message_receiver_;
// Store references to the video capturers we've created, so that we can stop
// them, if required.
std::vector<cricket::VideoCapturer*> video_capturers_;
};
class JsepTestClient
: public PeerConnectionTestClientBase<JsepMessageReceiver> {
public:
static JsepTestClient* CreateClient(
const std::string& id,
const MediaConstraintsInterface* constraints) {
JsepTestClient* client(new JsepTestClient(id));
if (!client->Init(constraints)) {
delete client;
return NULL;
}
return client;
}
~JsepTestClient() {}
virtual void Negotiate() {
Negotiate(true, true);
}
virtual void Negotiate(bool audio, bool video) {
rtc::scoped_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(offer.use()));
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()));
signaling_message_receiver()->ReceiveSdpMessage(
webrtc::SessionDescriptionInterface::kOffer, sdp);
}
// JsepMessageReceiver callback.
virtual void ReceiveSdpMessage(const std::string& type,
std::string& msg) {
FilterIncomingSdpMessage(&msg);
if (type == webrtc::SessionDescriptionInterface::kOffer) {
HandleIncomingOffer(msg);
} else {
HandleIncomingAnswer(msg);
}
}
// JsepMessageReceiver callback.
virtual void ReceiveIceMessage(const std::string& sdp_mid,
int sdp_mline_index,
const std::string& msg) {
LOG(INFO) << id() << "ReceiveIceMessage";
rtc::scoped_ptr<webrtc::IceCandidateInterface> candidate(
webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, msg, NULL));
EXPECT_TRUE(pc()->AddIceCandidate(candidate.get()));
}
// Implements PeerConnectionObserver functions needed by Jsep.
virtual void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) {
LOG(INFO) << id() << "OnIceCandidate";
std::string ice_sdp;
EXPECT_TRUE(candidate->ToString(&ice_sdp));
if (signaling_message_receiver() == NULL) {
// Remote party may be deleted.
return;
}
signaling_message_receiver()->ReceiveIceMessage(candidate->sdp_mid(),
candidate->sdp_mline_index(), ice_sdp);
}
void IceRestart() {
session_description_constraints_.SetMandatoryIceRestart(true);
SetExpectIceRestart(true);
}
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;
session_description_constraints_.SetMandatoryReceiveAudio(audio);
}
void SetReceiveVideo(bool video) {
if (video && can_receive_video())
return;
session_description_constraints_.SetMandatoryReceiveVideo(video);
}
void RemoveMsidFromReceivedSdp(bool remove) {
remove_msid_ = remove;
}
void RemoveSdesCryptoFromReceivedSdp(bool remove) {
remove_sdes_ = remove;
}
void RemoveBundleFromReceivedSdp(bool remove) {
remove_bundle_ = remove;
}
virtual bool can_receive_audio() {
bool value;
if (webrtc::FindConstraint(&session_description_constraints_,
MediaConstraintsInterface::kOfferToReceiveAudio, &value, NULL)) {
return value;
}
return true;
}
virtual bool can_receive_video() {
bool value;
if (webrtc::FindConstraint(&session_description_constraints_,
MediaConstraintsInterface::kOfferToReceiveVideo, &value, NULL)) {
return value;
}
return true;
}
virtual void OnIceComplete() {
LOG(INFO) << id() << "OnIceComplete";
}
virtual void OnDataChannel(DataChannelInterface* data_channel) {
LOG(INFO) << id() << "OnDataChannel";
data_channel_ = data_channel;
data_observer_.reset(new MockDataChannelObserver(data_channel));
}
void CreateDataChannel() {
data_channel_ = pc()->CreateDataChannel(kDataChannelLabel,
NULL);
ASSERT_TRUE(data_channel_.get() != NULL);
data_observer_.reset(new MockDataChannelObserver(data_channel_));
}
DataChannelInterface* data_channel() { return data_channel_; }
const MockDataChannelObserver* data_observer() const {
return data_observer_.get();
}
protected:
explicit JsepTestClient(const std::string& id)
: PeerConnectionTestClientBase<JsepMessageReceiver>(id),
remove_msid_(false),
remove_bundle_(false),
remove_sdes_(false) {
}
virtual rtc::scoped_refptr<webrtc::PeerConnectionInterface>
CreatePeerConnection(webrtc::PortAllocatorFactoryInterface* factory,
const MediaConstraintsInterface* constraints) {
// CreatePeerConnection with IceServers.
webrtc::PeerConnectionInterface::IceServers ice_servers;
webrtc::PeerConnectionInterface::IceServer ice_server;
ice_server.uri = "stun:stun.l.google.com:19302";
ice_servers.push_back(ice_server);
FakeIdentityService* dtls_service =
rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
new FakeIdentityService() : NULL;
return peer_connection_factory()->CreatePeerConnection(
ice_servers, constraints, factory, dtls_service, this);
}
void HandleIncomingOffer(const std::string& msg) {
LOG(INFO) << id() << "HandleIncomingOffer ";
if (NumberOfLocalMediaStreams() == 0) {
// If we are not sending any streams ourselves it is time to add some.
AddMediaStream(true, true);
}
rtc::scoped_ptr<SessionDescriptionInterface> desc(
webrtc::CreateSessionDescription("offer", msg, NULL));
EXPECT_TRUE(DoSetRemoteDescription(desc.release()));
rtc::scoped_ptr<SessionDescriptionInterface> answer;
EXPECT_TRUE(DoCreateAnswer(answer.use()));
std::string sdp;
EXPECT_TRUE(answer->ToString(&sdp));
EXPECT_TRUE(DoSetLocalDescription(answer.release()));
if (signaling_message_receiver()) {
signaling_message_receiver()->ReceiveSdpMessage(
webrtc::SessionDescriptionInterface::kAnswer, sdp);
}
}
void HandleIncomingAnswer(const std::string& msg) {
LOG(INFO) << id() << "HandleIncomingAnswer";
rtc::scoped_ptr<SessionDescriptionInterface> desc(
webrtc::CreateSessionDescription("answer", msg, NULL));
EXPECT_TRUE(DoSetRemoteDescription(desc.release()));
}
bool DoCreateOfferAnswer(SessionDescriptionInterface** desc,
bool offer) {
rtc::scoped_refptr<MockCreateSessionDescriptionObserver>
observer(new rtc::RefCountedObject<
MockCreateSessionDescriptionObserver>());
if (offer) {
pc()->CreateOffer(observer, &session_description_constraints_);
} else {
pc()->CreateAnswer(observer, &session_description_constraints_);
}
EXPECT_EQ_WAIT(true, observer->called(), kMaxWaitMs);
*desc = observer->release_desc();
if (observer->result() && ExpectIceRestart()) {
EXPECT_EQ(0u, (*desc)->candidates(0)->count());
}
return observer->result();
}
bool DoCreateOffer(SessionDescriptionInterface** desc) {
return DoCreateOfferAnswer(desc, true);
}
bool DoCreateAnswer(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);
}
}
private:
webrtc::FakeConstraints session_description_constraints_;
bool remove_msid_; // True if MSID should be removed in received SDP.
bool remove_bundle_; // True if bundle should be removed in received SDP.
bool remove_sdes_; // True if a=crypto should be removed in received SDP.
rtc::scoped_refptr<DataChannelInterface> data_channel_;
rtc::scoped_ptr<MockDataChannelObserver> data_observer_;
};
template <typename SignalingClass>
class P2PTestConductor : public testing::Test {
public:
P2PTestConductor()
: pss_(new rtc::PhysicalSocketServer),
ss_(new rtc::VirtualSocketServer(pss_.get())),
ss_scope_(ss_.get()) {}
bool SessionActive() {
return initiating_client_->SessionActive() &&
receiving_client_->SessionActive();
}
// Return true if the number of frames provided have been received or it is
// known that that will never occur (e.g. no frames will be sent or
// captured).
bool FramesNotPending(int audio_frames_to_receive,
int video_frames_to_receive) {
return VideoFramesReceivedCheck(video_frames_to_receive) &&
AudioFramesReceivedCheck(audio_frames_to_receive);
}
bool AudioFramesReceivedCheck(int frames_received) {
return initiating_client_->AudioFramesReceivedCheck(frames_received) &&
receiving_client_->AudioFramesReceivedCheck(frames_received);
}
bool VideoFramesReceivedCheck(int frames_received) {
return initiating_client_->VideoFramesReceivedCheck(frames_received) &&
receiving_client_->VideoFramesReceivedCheck(frames_received);
}
void VerifyDtmf() {
initiating_client_->VerifyDtmf();
receiving_client_->VerifyDtmf();
}
void TestUpdateOfferWithRejectedContent() {
initiating_client_->Negotiate(true, false);
EXPECT_TRUE_WAIT(
FramesNotPending(kEndAudioFrameCount * 2, kEndVideoFrameCount),
kMaxWaitForFramesMs);
// There shouldn't be any more video frame after the new offer is
// negotiated.
EXPECT_FALSE(VideoFramesReceivedCheck(kEndVideoFrameCount + 1));
}
void VerifyRenderedSize(int width, int height) {
EXPECT_EQ(width, receiving_client()->rendered_width());
EXPECT_EQ(height, receiving_client()->rendered_height());
EXPECT_EQ(width, initializing_client()->rendered_width());
EXPECT_EQ(height, initializing_client()->rendered_height());
}
void VerifySessionDescriptions() {
initiating_client_->VerifyRejectedMediaInSessionDescription();
receiving_client_->VerifyRejectedMediaInSessionDescription();
initiating_client_->VerifyLocalIceUfragAndPassword();
receiving_client_->VerifyLocalIceUfragAndPassword();
}
~P2PTestConductor() {
if (initiating_client_) {
initiating_client_->set_signaling_message_receiver(NULL);
}
if (receiving_client_) {
receiving_client_->set_signaling_message_receiver(NULL);
}
}
bool CreateTestClients() {
return CreateTestClients(NULL, NULL);
}
bool CreateTestClients(MediaConstraintsInterface* init_constraints,
MediaConstraintsInterface* recv_constraints) {
initiating_client_.reset(SignalingClass::CreateClient("Caller: ",
init_constraints));
receiving_client_.reset(SignalingClass::CreateClient("Callee: ",
recv_constraints));
if (!initiating_client_ || !receiving_client_) {
return false;
}
initiating_client_->set_signaling_message_receiver(receiving_client_.get());
receiving_client_->set_signaling_message_receiver(initiating_client_.get());
return true;
}
void SetVideoConstraints(const webrtc::FakeConstraints& init_constraints,
const webrtc::FakeConstraints& recv_constraints) {
initiating_client_->SetVideoConstraints(init_constraints);
receiving_client_->SetVideoConstraints(recv_constraints);
}
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();
const int kMaxWaitForActivationMs = 5000;
// 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;
// TODO(ronghuawu): Add test to cover the case of sendonly and recvonly.
if (!initiating_client_->can_receive_audio() ||
!receiving_client_->can_receive_audio()) {
audio_frame_count = -1;
}
int video_frame_count = kEndVideoFrameCount;
if (!initiating_client_->can_receive_video() ||
!receiving_client_->can_receive_video()) {
video_frame_count = -1;
}
if (audio_frame_count != -1 || video_frame_count != -1) {
// 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);
}
if (initiating_client_->can_receive_audio() ||
initiating_client_->can_receive_video()) {
// The initiating client can receive media, so it must produce candidates
// that will serve as destinations for that media.
// TODO(bemasc): Understand why the state is not already Complete here, as
// seems to be the case for the receiving client. This may indicate a bug
// in the ICE gathering system.
EXPECT_NE(webrtc::PeerConnectionInterface::kIceGatheringNew,
initiating_client_->ice_gathering_state());
}
if (receiving_client_->can_receive_audio() ||
receiving_client_->can_receive_video()) {
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
receiving_client_->ice_gathering_state(),
kMaxWaitForFramesMs);
}
EXPECT_TRUE_WAIT(FramesNotPending(audio_frame_count, video_frame_count),
kMaxWaitForFramesMs);
}
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));
}
}
SignalingClass* initializing_client() { return initiating_client_.get(); }
SignalingClass* receiving_client() { return receiving_client_.get(); }
private:
rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
rtc::scoped_ptr<rtc::VirtualSocketServer> ss_;
rtc::SocketServerScope ss_scope_;
rtc::scoped_ptr<SignalingClass> initiating_client_;
rtc::scoped_ptr<SignalingClass> receiving_client_;
};
typedef P2PTestConductor<JsepTestClient> JsepPeerConnectionP2PTestClient;
// Disable for TSan v2, see
// https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
#if !defined(THREAD_SANITIZER)
// 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(JsepPeerConnectionP2PTestClient, 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(JsepPeerConnectionP2PTestClient, 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(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTest1280By720) {
ASSERT_TRUE(CreateTestClients());
FakeConstraints constraint;
constraint.SetMandatoryMinWidth(1280);
constraint.SetMandatoryMinHeight(720);
SetVideoConstraints(constraint, constraint);
LocalP2PTest();
VerifyRenderedSize(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(JsepPeerConnectionP2PTestClient, LocalP2PTestDtls) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints setup_constraints;
setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
true);
ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
LocalP2PTest();
VerifyRenderedSize(640, 480);
}
// This test sets up a audio call initially and then upgrades to audio/video,
// using DTLS.
TEST_F(JsepPeerConnectionP2PTestClient, 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 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(JsepPeerConnectionP2PTestClient, 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();
VerifyRenderedSize(640, 480);
}
// This test sets up a Jsep call between two parties, and the callee only
// accept to receive video.
TEST_F(JsepPeerConnectionP2PTestClient, 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(JsepPeerConnectionP2PTestClient, 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(JsepPeerConnectionP2PTestClient, 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.
// Disabled due to b/14955157.
TEST_F(JsepPeerConnectionP2PTestClient,
DISABLED_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.
// Disabled due to b/14955157.
TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_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();
}
// This test sets up a Jsep call between two parties and the initiating peer
// sends two steams.
// TODO(perkj): Disabled due to
// https://code.google.com/p/webrtc/issues/detail?id=1454
TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_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(JsepPeerConnectionP2PTestClient, 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(JsepPeerConnectionP2PTestClient, 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(JsepPeerConnectionP2PTestClient, 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(JsepPeerConnectionP2PTestClient, 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 we can get negotiated ciphers.
TEST_F(JsepPeerConnectionP2PTestClient, GetNegotiatedCiphersStats) {
ASSERT_TRUE(CreateTestClients());
LocalP2PTest();
EXPECT_EQ_WAIT(
rtc::SSLStreamAdapter::GetDefaultSslCipher(),
initializing_client()->GetDtlsCipherStats(),
kMaxWaitForStatsMs);
EXPECT_EQ_WAIT(
kDefaultSrtpCipher,
initializing_client()->GetSrtpCipherStats(),
kMaxWaitForStatsMs);
}
// This test sets up a call between two parties with audio, video and data.
TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDataChannel) {
FakeConstraints setup_constraints;
setup_constraints.SetAllowRtpDataChannels();
ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
initializing_client()->CreateDataChannel();
LocalP2PTest();
ASSERT_TRUE(initializing_client()->data_channel() != NULL);
ASSERT_TRUE(receiving_client()->data_channel() != NULL);
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 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(JsepPeerConnectionP2PTestClient, 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() != NULL);
ASSERT_TRUE(receiving_client()->data_channel() != NULL);
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(JsepPeerConnectionP2PTestClient, 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() != NULL);
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(JsepPeerConnectionP2PTestClient, 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() != NULL);
ASSERT_TRUE(receiving_client()->data_channel() != NULL);
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(JsepPeerConnectionP2PTestClient, 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(JsepPeerConnectionP2PTestClient, 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);
}
// 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(JsepPeerConnectionP2PTestClient,
DISABLED_LocalP2PTestWithVideoDecoderFactory) {
ASSERT_TRUE(CreateTestClients());
EnableVideoDecoderFactory();
LocalP2PTest();
}
#endif // if !defined(THREAD_SANITIZER)