Update libjingle to 55618622.
Update libyuv to r826.
TEST=try bots
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2889004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5038 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/app/webrtc/localaudiosource.cc b/talk/app/webrtc/localaudiosource.cc
index 3663aac..2cd472a 100644
--- a/talk/app/webrtc/localaudiosource.cc
+++ b/talk/app/webrtc/localaudiosource.cc
@@ -53,7 +53,9 @@
"googHighpassFilter";
const char MediaConstraintsInterface::kTypingNoiseDetection[] =
"googTypingNoiseDetection";
-const char MediaConstraintsInterface::kInternalAecDump[] = "internalAecDump";
+const char MediaConstraintsInterface::kAudioMirroring[] = "googAudioMirroring";
+// TODO(perkj): Remove kInternalAecDump once its not used by Chrome.
+const char MediaConstraintsInterface::kInternalAecDump[] = "deprecatedAecDump";
namespace {
@@ -90,10 +92,10 @@
options->noise_suppression.Set(value);
else if (iter->key == MediaConstraintsInterface::kHighpassFilter)
options->highpass_filter.Set(value);
- else if (iter->key == MediaConstraintsInterface::kInternalAecDump)
- options->aec_dump.Set(value);
else if (iter->key == MediaConstraintsInterface::kTypingNoiseDetection)
options->typing_detection.Set(value);
+ else if (iter->key == MediaConstraintsInterface::kAudioMirroring)
+ options->stereo_swapping.Set(value);
else
success = false;
}
@@ -103,14 +105,16 @@
} // namespace
talk_base::scoped_refptr<LocalAudioSource> LocalAudioSource::Create(
+ const PeerConnectionFactoryInterface::Options& options,
const MediaConstraintsInterface* constraints) {
talk_base::scoped_refptr<LocalAudioSource> source(
new talk_base::RefCountedObject<LocalAudioSource>());
- source->Initialize(constraints);
+ source->Initialize(options, constraints);
return source;
}
void LocalAudioSource::Initialize(
+ const PeerConnectionFactoryInterface::Options& options,
const MediaConstraintsInterface* constraints) {
if (!constraints)
return;
@@ -119,12 +123,14 @@
// constraints.
FromConstraints(constraints->GetOptional(), &options_);
- cricket::AudioOptions options;
- if (!FromConstraints(constraints->GetMandatory(), &options)) {
+ cricket::AudioOptions audio_options;
+ if (!FromConstraints(constraints->GetMandatory(), &audio_options)) {
source_state_ = kEnded;
return;
}
- options_.SetAll(options);
+ options_.SetAll(audio_options);
+ if (options.enable_aec_dump)
+ options_.aec_dump.Set(true);
source_state_ = kLive;
}
diff --git a/talk/app/webrtc/localaudiosource.h b/talk/app/webrtc/localaudiosource.h
index e0fda03..fb769ed 100644
--- a/talk/app/webrtc/localaudiosource.h
+++ b/talk/app/webrtc/localaudiosource.h
@@ -30,6 +30,7 @@
#include "talk/app/webrtc/mediastreaminterface.h"
#include "talk/app/webrtc/notifier.h"
+#include "talk/app/webrtc/peerconnectioninterface.h"
#include "talk/base/scoped_ptr.h"
#include "talk/media/base/mediachannel.h"
@@ -44,6 +45,7 @@
public:
// Creates an instance of LocalAudioSource.
static talk_base::scoped_refptr<LocalAudioSource> Create(
+ const PeerConnectionFactoryInterface::Options& options,
const MediaConstraintsInterface* constraints);
virtual SourceState state() const { return source_state_; }
@@ -58,7 +60,8 @@
}
private:
- void Initialize(const MediaConstraintsInterface* constraints);
+ void Initialize(const PeerConnectionFactoryInterface::Options& options,
+ const MediaConstraintsInterface* constraints);
cricket::AudioOptions options_;
SourceState source_state_;
diff --git a/talk/app/webrtc/localaudiosource_unittest.cc b/talk/app/webrtc/localaudiosource_unittest.cc
index ae07787..f8880e0 100644
--- a/talk/app/webrtc/localaudiosource_unittest.cc
+++ b/talk/app/webrtc/localaudiosource_unittest.cc
@@ -39,6 +39,7 @@
using webrtc::LocalAudioSource;
using webrtc::MediaConstraintsInterface;
using webrtc::MediaSourceInterface;
+using webrtc::PeerConnectionFactoryInterface;
TEST(LocalAudioSourceTest, SetValidOptions) {
webrtc::FakeConstraints constraints;
@@ -52,7 +53,8 @@
constraints.AddOptional(MediaConstraintsInterface::kHighpassFilter, true);
talk_base::scoped_refptr<LocalAudioSource> source =
- LocalAudioSource::Create(&constraints);
+ LocalAudioSource::Create(PeerConnectionFactoryInterface::Options(),
+ &constraints);
bool value;
EXPECT_TRUE(source->options().echo_cancellation.Get(&value));
@@ -72,7 +74,8 @@
TEST(LocalAudioSourceTest, OptionNotSet) {
webrtc::FakeConstraints constraints;
talk_base::scoped_refptr<LocalAudioSource> source =
- LocalAudioSource::Create(&constraints);
+ LocalAudioSource::Create(PeerConnectionFactoryInterface::Options(),
+ &constraints);
bool value;
EXPECT_FALSE(source->options().highpass_filter.Get(&value));
}
@@ -83,7 +86,8 @@
constraints.AddOptional(MediaConstraintsInterface::kEchoCancellation, true);
talk_base::scoped_refptr<LocalAudioSource> source =
- LocalAudioSource::Create(&constraints);
+ LocalAudioSource::Create(PeerConnectionFactoryInterface::Options(),
+ &constraints);
bool value;
EXPECT_TRUE(source->options().echo_cancellation.Get(&value));
@@ -96,7 +100,8 @@
constraints.AddOptional("invalidKey", false);
talk_base::scoped_refptr<LocalAudioSource> source =
- LocalAudioSource::Create(&constraints);
+ LocalAudioSource::Create(PeerConnectionFactoryInterface::Options(),
+ &constraints);
EXPECT_EQ(MediaSourceInterface::kLive, source->state());
bool value;
@@ -110,7 +115,8 @@
constraints.AddMandatory("invalidKey", false);
talk_base::scoped_refptr<LocalAudioSource> source =
- LocalAudioSource::Create(&constraints);
+ LocalAudioSource::Create(PeerConnectionFactoryInterface::Options(),
+ &constraints);
EXPECT_EQ(MediaSourceInterface::kEnded, source->state());
bool value;
diff --git a/talk/app/webrtc/mediaconstraintsinterface.h b/talk/app/webrtc/mediaconstraintsinterface.h
index 48022dd..97d5cf6 100644
--- a/talk/app/webrtc/mediaconstraintsinterface.h
+++ b/talk/app/webrtc/mediaconstraintsinterface.h
@@ -81,6 +81,7 @@
static const char kNoiseSuppression[]; // googNoiseSuppression
static const char kHighpassFilter[]; // googHighpassFilter
static const char kTypingNoiseDetection[]; // googTypingNoiseDetection
+ static const char kAudioMirroring[]; // googAudioMirroring
// Google-specific constraint keys for a local video source
static const char kNoiseReduction[]; // googNoiseReduction
@@ -106,6 +107,8 @@
static const char kEnableDtlsSrtp[]; // Enable DTLS-SRTP
// Temporary pseudo-constraints used to enable DataChannels
static const char kEnableRtpDataChannels[]; // Enable RTP DataChannels
+ // TODO(perkj): Remove kEnableSctpDataChannels once Chrome use
+ // PeerConnectionFactory::SetOptions.
static const char kEnableSctpDataChannels[]; // Enable SCTP DataChannels
// The prefix of internal-only constraints whose JS set values should be
@@ -116,9 +119,8 @@
// line flags. So they are prefixed with "internal" so JS values will be
// removed.
// Used by a local audio source.
+ // TODO(perkj): Remove once Chrome use PeerConnectionFactory::SetOptions.
static const char kInternalAecDump[]; // internalAecDump
- // Used for disabling security and use plain RTP.
- static const char kInternalDisableEncryption[]; // internalDisableEncryption
protected:
// Dtor protected as objects shouldn't be deleted via this interface
diff --git a/talk/app/webrtc/mediastreamsignaling.cc b/talk/app/webrtc/mediastreamsignaling.cc
index 771a4e8..ef9f3e0 100644
--- a/talk/app/webrtc/mediastreamsignaling.cc
+++ b/talk/app/webrtc/mediastreamsignaling.cc
@@ -267,6 +267,10 @@
}
scoped_refptr<DataChannel> channel(
data_channel_factory_->CreateDataChannel(label, &config));
+ if (!channel.get()) {
+ LOG(LS_ERROR) << "Failed to create DataChannel from the OPEN message.";
+ return false;
+ }
data_channels_[label] = channel;
stream_observer_->OnAddDataChannel(channel);
// It's immediately ready to use.
diff --git a/talk/app/webrtc/peerconnection.cc b/talk/app/webrtc/peerconnection.cc
index 7fafad7..de88f88 100644
--- a/talk/app/webrtc/peerconnection.cc
+++ b/talk/app/webrtc/peerconnection.cc
@@ -317,7 +317,8 @@
stats_.set_session(session_.get());
// Initialize the WebRtcSession. It creates transport channels etc.
- if (!session_->Initialize(constraints, dtls_identity_service))
+ if (!session_->Initialize(factory_->options(), constraints,
+ dtls_identity_service))
return false;
// Register PeerConnection as receiver of local ice candidates.
diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc
index 9de6ec9..76d9cd7 100644
--- a/talk/app/webrtc/peerconnection_unittest.cc
+++ b/talk/app/webrtc/peerconnection_unittest.cc
@@ -157,8 +157,8 @@
// Disable highpass filter so that we can get all the test audio frames.
constraints.AddMandatory(
MediaConstraintsInterface::kHighpassFilter, false);
- talk_base::scoped_refptr<webrtc::LocalAudioSource> source =
- webrtc::LocalAudioSource::Create(&constraints);
+ talk_base::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.
talk_base::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
diff --git a/talk/app/webrtc/peerconnectionfactory.cc b/talk/app/webrtc/peerconnectionfactory.cc
index 7d30fab..e8b8f63 100644
--- a/talk/app/webrtc/peerconnectionfactory.cc
+++ b/talk/app/webrtc/peerconnectionfactory.cc
@@ -261,7 +261,7 @@
PeerConnectionFactory::CreateAudioSource_s(
const MediaConstraintsInterface* constraints) {
talk_base::scoped_refptr<LocalAudioSource> source(
- LocalAudioSource::Create(constraints));
+ LocalAudioSource::Create(options_, constraints));
return source;
}
diff --git a/talk/app/webrtc/peerconnectionfactory.h b/talk/app/webrtc/peerconnectionfactory.h
index 7faf609..dff885d 100644
--- a/talk/app/webrtc/peerconnectionfactory.h
+++ b/talk/app/webrtc/peerconnectionfactory.h
@@ -40,6 +40,10 @@
class PeerConnectionFactory : public PeerConnectionFactoryInterface,
public talk_base::MessageHandler {
public:
+ virtual void SetOptions(const Options& options) {
+ options_ = options;
+ }
+
virtual talk_base::scoped_refptr<PeerConnectionInterface>
CreatePeerConnection(
const PeerConnectionInterface::IceServers& configuration,
@@ -77,6 +81,7 @@
virtual cricket::ChannelManager* channel_manager();
virtual talk_base::Thread* signaling_thread();
virtual talk_base::Thread* worker_thread();
+ const Options& options() const { return options_; }
protected:
PeerConnectionFactory();
@@ -109,6 +114,7 @@
bool owns_ptrs_;
talk_base::Thread* signaling_thread_;
talk_base::Thread* worker_thread_;
+ Options options_;
talk_base::scoped_refptr<PortAllocatorFactoryInterface> allocator_factory_;
// External Audio device used for audio playback.
talk_base::scoped_refptr<AudioDeviceModule> default_adm_;
diff --git a/talk/app/webrtc/peerconnectioninterface.h b/talk/app/webrtc/peerconnectioninterface.h
index bd86f3e..a127dad 100644
--- a/talk/app/webrtc/peerconnectioninterface.h
+++ b/talk/app/webrtc/peerconnectioninterface.h
@@ -390,6 +390,19 @@
// argument.
class PeerConnectionFactoryInterface : public talk_base::RefCountInterface {
public:
+ class Options {
+ public:
+ Options() :
+ enable_aec_dump(false),
+ disable_encryption(false),
+ disable_sctp_data_channels(false) {
+ }
+ bool enable_aec_dump;
+ bool disable_encryption;
+ bool disable_sctp_data_channels;
+ };
+
+ virtual void SetOptions(const Options& options) = 0;
virtual talk_base::scoped_refptr<PeerConnectionInterface>
CreatePeerConnection(
const PeerConnectionInterface::IceServers& configuration,
diff --git a/talk/app/webrtc/statscollector.cc b/talk/app/webrtc/statscollector.cc
index 42ae662..db7cac4 100644
--- a/talk/app/webrtc/statscollector.cc
+++ b/talk/app/webrtc/statscollector.cc
@@ -82,6 +82,15 @@
"googFrameRateDecoded";
const char StatsReport::kStatsValueNameFrameRateOutput[] =
"googFrameRateOutput";
+const char StatsReport::kStatsValueNameDecodeMs[] = "googDecodeMs";
+const char StatsReport::kStatsValueNameMaxDecodeMs[] = "googMaxDecodeMs";
+const char StatsReport::kStatsValueNameCurrentDelayMs[] = "googCurrentDelayMs";
+const char StatsReport::kStatsValueNameTargetDelayMs[] = "googTargetDelayMs";
+const char StatsReport::kStatsValueNameJitterBufferMs[] = "googJitterBufferMs";
+const char StatsReport::kStatsValueNameMinPlayoutDelayMs[] =
+ "googMinPlayoutDelayMs";
+const char StatsReport::kStatsValueNameRenderDelayMs[] = "googRenderDelayMs";
+
const char StatsReport::kStatsValueNameFrameRateInput[] = "googFrameRateInput";
const char StatsReport::kStatsValueNameFrameRateSent[] = "googFrameRateSent";
const char StatsReport::kStatsValueNameFrameWidthReceived[] =
@@ -119,6 +128,7 @@
const char StatsReport::kStatsReportTypeSession[] = "googLibjingleSession";
const char StatsReport::kStatsReportTypeBwe[] = "VideoBwe";
+const char StatsReport::kStatsReportTypeRemoteSsrc[] = "remoteSsrc";
const char StatsReport::kStatsReportTypeSsrc[] = "ssrc";
const char StatsReport::kStatsReportTypeTrack[] = "googTrack";
const char StatsReport::kStatsReportTypeIceCandidate[] = "iceCandidate";
@@ -241,6 +251,21 @@
info.framerate_decoded);
report->AddValue(StatsReport::kStatsValueNameFrameRateOutput,
info.framerate_output);
+
+ report->AddValue(StatsReport::kStatsValueNameDecodeMs,
+ info.decode_ms);
+ report->AddValue(StatsReport::kStatsValueNameMaxDecodeMs,
+ info.max_decode_ms);
+ report->AddValue(StatsReport::kStatsValueNameCurrentDelayMs,
+ info.current_delay_ms);
+ report->AddValue(StatsReport::kStatsValueNameTargetDelayMs,
+ info.target_delay_ms);
+ report->AddValue(StatsReport::kStatsValueNameJitterBufferMs,
+ info.jitter_buffer_ms);
+ report->AddValue(StatsReport::kStatsValueNameMinPlayoutDelayMs,
+ info.min_playout_delay_ms);
+ report->AddValue(StatsReport::kStatsValueNameRenderDelayMs,
+ info.render_delay_ms);
}
void ExtractStats(const cricket::VideoSenderInfo& info, StatsReport* report) {
@@ -293,6 +318,18 @@
info.bucket_delay);
}
+void ExtractRemoteStats(const cricket::MediaSenderInfo& info,
+ StatsReport* report) {
+ report->timestamp = info.remote_stats[0].timestamp;
+ // TODO(hta): Extract some stats here.
+}
+
+void ExtractRemoteStats(const cricket::MediaReceiverInfo& info,
+ StatsReport* report) {
+ report->timestamp = info.remote_stats[0].timestamp;
+ // TODO(hta): Extract some stats here.
+}
+
uint32 ExtractSsrc(const cricket::VoiceReceiverInfo& info) {
return info.ssrc;
}
@@ -319,11 +356,20 @@
for (; it != data.end(); ++it) {
std::string id;
uint32 ssrc = ExtractSsrc(*it);
- StatsReport* report = collector->PrepareReport(ssrc, transport_id);
+ // Each object can result in 2 objects, a local and a remote object.
+ // TODO(hta): Handle the case of multiple SSRCs per object.
+ StatsReport* report = collector->PrepareLocalReport(ssrc, transport_id);
if (!report) {
continue;
}
ExtractStats(*it, report);
+ if (it->remote_stats.size() > 0) {
+ report = collector->PrepareRemoteReport(ssrc, transport_id);
+ if (!report) {
+ continue;
+ }
+ ExtractRemoteStats(*it, report);
+ }
}
};
@@ -406,8 +452,9 @@
}
}
-StatsReport* StatsCollector::PrepareReport(uint32 ssrc,
- const std::string& transport_id) {
+StatsReport* StatsCollector::PrepareLocalReport(
+ uint32 ssrc,
+ const std::string& transport_id) {
std::string ssrc_id = talk_base::ToString<uint32>(ssrc);
StatsMap::iterator it = reports_.find(StatsId(
StatsReport::kStatsReportTypeSsrc, ssrc_id));
@@ -427,10 +474,8 @@
&track_id);
}
- StatsReport* report = &reports_[
- StatsId(StatsReport::kStatsReportTypeSsrc, ssrc_id)];
- report->id = StatsId(StatsReport::kStatsReportTypeSsrc, ssrc_id);
- report->type = StatsReport::kStatsReportTypeSsrc;
+ StatsReport* report = GetOrCreateReport(StatsReport::kStatsReportTypeSsrc,
+ ssrc_id);
// Clear out stats from previous GatherStats calls if any.
if (report->timestamp != stats_gathering_started_) {
@@ -446,6 +491,44 @@
return report;
}
+StatsReport* StatsCollector::PrepareRemoteReport(
+ uint32 ssrc,
+ const std::string& transport_id) {
+ std::string ssrc_id = talk_base::ToString<uint32>(ssrc);
+ StatsMap::iterator it = reports_.find(StatsId(
+ StatsReport::kStatsReportTypeRemoteSsrc, ssrc_id));
+
+ std::string track_id;
+ if (it == reports_.end()) {
+ if (!session()->GetTrackIdBySsrc(ssrc, &track_id)) {
+ LOG(LS_WARNING) << "The SSRC " << ssrc
+ << " is not associated with a track";
+ return NULL;
+ }
+ } else {
+ // Keeps the old track id since we want to report the stats for inactive
+ // tracks.
+ ExtractValueFromReport(it->second,
+ StatsReport::kStatsValueNameTrackId,
+ &track_id);
+ }
+
+ StatsReport* report = GetOrCreateReport(
+ StatsReport::kStatsReportTypeRemoteSsrc, ssrc_id);
+
+ // Clear out stats from previous GatherStats calls if any.
+ // The timestamp will be added later. Zero it for debugging.
+ report->values.clear();
+ report->timestamp = 0;
+
+ report->AddValue(StatsReport::kStatsValueNameSsrc, ssrc_id);
+ report->AddValue(StatsReport::kStatsValueNameTrackId, track_id);
+ // Add the mapping of SSRC to transport.
+ report->AddValue(StatsReport::kStatsValueNameTransportId,
+ transport_id);
+ return report;
+}
+
std::string StatsCollector::AddOneCertificateReport(
const talk_base::SSLCertificate* cert, const std::string& issuer_id) {
// TODO(bemasc): Move this computation to a helper class that caches these
@@ -592,6 +675,9 @@
info.local_candidate.address().ToString());
report.AddValue(StatsReport::kStatsValueNameRemoteAddress,
info.remote_candidate.address().ToString());
+ report.AddValue(StatsReport::kStatsValueNameRtt, info.rtt);
+ report.AddValue(StatsReport::kStatsValueNameTransportType,
+ info.local_candidate.protocol());
reports_[report.id] = report;
}
}
@@ -668,4 +754,19 @@
return true;
}
+StatsReport* StatsCollector::GetOrCreateReport(const std::string& type,
+ const std::string& id) {
+ std::string statsid = StatsId(type, id);
+ StatsReport* report = NULL;
+ std::map<std::string, StatsReport>::iterator it = reports_.find(statsid);
+ if (it == reports_.end()) {
+ report = &reports_[statsid]; // Create new element.
+ report->id = statsid;
+ report->type = type;
+ } else {
+ report = &reports_[statsid];
+ }
+ return report;
+}
+
} // namespace webrtc
diff --git a/talk/app/webrtc/statscollector.h b/talk/app/webrtc/statscollector.h
index c34b5a0..01da059 100644
--- a/talk/app/webrtc/statscollector.h
+++ b/talk/app/webrtc/statscollector.h
@@ -65,9 +65,11 @@
// |reports|.
bool GetStats(MediaStreamTrackInterface* track, StatsReports* reports);
- WebRtcSession* session() { return session_; }
- // Prepare an SSRC report for the given ssrc. Used internally.
- StatsReport* PrepareReport(uint32 ssrc, const std::string& transport);
+ // Prepare an SSRC report for the given ssrc. Used internally
+ // in the ExtractStatsFromList template.
+ StatsReport* PrepareLocalReport(uint32 ssrc, const std::string& transport);
+ // Prepare an SSRC report for the given remote ssrc. Used internally.
+ StatsReport* PrepareRemoteReport(uint32 ssrc, const std::string& transport);
// Extracts the ID of a Transport belonging to an SSRC. Used internally.
bool GetTransportIdFromProxy(const std::string& proxy,
std::string* transport_id);
@@ -88,9 +90,12 @@
void ExtractVideoInfo();
double GetTimeNow();
void BuildSsrcToTransportId();
+ WebRtcSession* session() { return session_; }
+ webrtc::StatsReport* GetOrCreateReport(const std::string& type,
+ const std::string& id);
// A map from the report id to the report.
- std::map<std::string, webrtc::StatsReport> reports_;
+ std::map<std::string, StatsReport> reports_;
// Raw pointer to the session the statistics are gathered from.
WebRtcSession* session_;
double stats_gathering_started_;
diff --git a/talk/app/webrtc/statscollector_unittest.cc b/talk/app/webrtc/statscollector_unittest.cc
index d9cde5d..66a5ee0 100644
--- a/talk/app/webrtc/statscollector_unittest.cc
+++ b/talk/app/webrtc/statscollector_unittest.cc
@@ -58,6 +58,10 @@
const char kNotFound[] = "NOT FOUND";
const char kNoReports[] = "NO REPORTS";
+// Constant names for track identification.
+const char kTrackId[] = "somename";
+const uint32 kSsrcOfTrack = 1234;
+
class MockWebRtcSession : public webrtc::WebRtcSession {
public:
explicit MockWebRtcSession(cricket::ChannelManager* channel_manager)
@@ -207,11 +211,37 @@
new cricket::ChannelManager(media_engine_,
new cricket::FakeDeviceManager(),
talk_base::Thread::Current())),
- session_(channel_manager_.get()) {
+ session_(channel_manager_.get()),
+ track_id_(kTrackId) {
// By default, we ignore session GetStats calls.
EXPECT_CALL(session_, GetStats(_)).WillRepeatedly(Return(false));
}
+ // This creates a standard setup with a transport called "trspname"
+ // having one transport channel
+ // and the specified virtual connection name.
+ void InitSessionStats(const std::string vc_name) {
+ const std::string kTransportName("trspname");
+ cricket::TransportStats transport_stats;
+ cricket::TransportChannelStats channel_stats;
+ channel_stats.component = 1;
+ transport_stats.content_name = kTransportName;
+ transport_stats.channel_stats.push_back(channel_stats);
+
+ session_stats_.transport_stats[kTransportName] = transport_stats;
+ session_stats_.proxy_to_transport[vc_name] = kTransportName;
+ }
+
+ // Adds a track with a given SSRC into the stats.
+ void AddVideoTrackStats() {
+ stream_ = webrtc::MediaStream::Create("streamlabel");
+ track_= webrtc::VideoTrack::Create(kTrackId, NULL);
+ stream_->AddTrack(track_);
+ EXPECT_CALL(session_, GetTrackIdBySsrc(kSsrcOfTrack, _))
+ .WillRepeatedly(DoAll(SetArgPointee<1>(track_id_),
+ Return(true)));
+ }
+
void TestCertificateReports(const talk_base::FakeSSLCertificate& local_cert,
const std::vector<std::string>& local_ders,
const talk_base::FakeSSLCertificate& remote_cert,
@@ -283,9 +313,14 @@
EXPECT_NE(kNotFound, remote_certificate_id);
CheckCertChainReports(reports, remote_ders, remote_certificate_id);
}
+
cricket::FakeMediaEngine* media_engine_;
talk_base::scoped_ptr<cricket::ChannelManager> channel_manager_;
MockWebRtcSession session_;
+ cricket::SessionStats session_stats_;
+ talk_base::scoped_refptr<webrtc::MediaStream> stream_;
+ talk_base::scoped_refptr<webrtc::VideoTrack> track_;
+ std::string track_id_;
};
// This test verifies that 64-bit counters are passed successfully.
@@ -297,17 +332,13 @@
webrtc::StatsReports reports; // returned values.
cricket::VideoSenderInfo video_sender_info;
cricket::VideoMediaInfo stats_read;
- const uint32 kSsrcOfTrack = 1234;
- const std::string kNameOfTrack("somename");
// The number of bytes must be larger than 0xFFFFFFFF for this test.
const int64 kBytesSent = 12345678901234LL;
const std::string kBytesSentString("12345678901234");
stats.set_session(&session_);
- talk_base::scoped_refptr<webrtc::MediaStream> stream(
- webrtc::MediaStream::Create("streamlabel"));
- stream->AddTrack(webrtc::VideoTrack::Create(kNameOfTrack, NULL));
- stats.AddStream(stream);
+ AddVideoTrackStats();
+ stats.AddStream(stream_);
// Construct a stats value to read.
video_sender_info.ssrcs.push_back(1234);
@@ -319,9 +350,6 @@
EXPECT_CALL(*media_channel, GetStats(_))
.WillOnce(DoAll(SetArgPointee<0>(stats_read),
Return(true)));
- EXPECT_CALL(session_, GetTrackIdBySsrc(kSsrcOfTrack, _))
- .WillOnce(DoAll(SetArgPointee<1>(kNameOfTrack),
- Return(true)));
stats.UpdateStats();
stats.GetStats(NULL, &reports);
std::string result = ExtractSsrcStatsValue(reports, "bytesSent");
@@ -339,16 +367,12 @@
cricket::VideoMediaInfo stats_read;
// Set up an SSRC just to test that we get both kinds of stats back: SSRC and
// BWE.
- const uint32 kSsrcOfTrack = 1234;
- const std::string kNameOfTrack("somename");
const int64 kBytesSent = 12345678901234LL;
const std::string kBytesSentString("12345678901234");
stats.set_session(&session_);
- talk_base::scoped_refptr<webrtc::MediaStream> stream(
- webrtc::MediaStream::Create("streamlabel"));
- stream->AddTrack(webrtc::VideoTrack::Create(kNameOfTrack, NULL));
- stats.AddStream(stream);
+ AddVideoTrackStats();
+ stats.AddStream(stream_);
// Construct a stats value to read.
video_sender_info.ssrcs.push_back(1234);
@@ -365,9 +389,7 @@
EXPECT_CALL(*media_channel, GetStats(_))
.WillOnce(DoAll(SetArgPointee<0>(stats_read),
Return(true)));
- EXPECT_CALL(session_, GetTrackIdBySsrc(kSsrcOfTrack, _))
- .WillOnce(DoAll(SetArgPointee<1>(kNameOfTrack),
- Return(true)));
+
stats.UpdateStats();
stats.GetStats(NULL, &reports);
std::string result = ExtractSsrcStatsValue(reports, "bytesSent");
@@ -417,13 +439,8 @@
MockVideoMediaChannel* media_channel = new MockVideoMediaChannel;
cricket::VideoChannel video_channel(talk_base::Thread::Current(),
media_engine_, media_channel, &session_, "", false, NULL);
- const std::string kTrackId("somename");
- talk_base::scoped_refptr<webrtc::MediaStream> stream(
- webrtc::MediaStream::Create("streamlabel"));
- talk_base::scoped_refptr<webrtc::VideoTrack> track =
- webrtc::VideoTrack::Create(kTrackId, NULL);
- stream->AddTrack(track);
- stats.AddStream(stream);
+ AddVideoTrackStats();
+ stats.AddStream(stream_);
stats.set_session(&session_);
@@ -449,13 +466,8 @@
MockVideoMediaChannel* media_channel = new MockVideoMediaChannel;
cricket::VideoChannel video_channel(talk_base::Thread::Current(),
media_engine_, media_channel, &session_, "", false, NULL);
- const std::string kTrackId("somename");
- talk_base::scoped_refptr<webrtc::MediaStream> stream(
- webrtc::MediaStream::Create("streamlabel"));
- talk_base::scoped_refptr<webrtc::VideoTrack> track =
- webrtc::VideoTrack::Create(kTrackId, NULL);
- stream->AddTrack(track);
- stats.AddStream(stream);
+ AddVideoTrackStats();
+ stats.AddStream(stream_);
stats.set_session(&session_);
@@ -464,7 +476,6 @@
// Constructs an ssrc stats update.
cricket::VideoSenderInfo video_sender_info;
cricket::VideoMediaInfo stats_read;
- const uint32 kSsrcOfTrack = 1234;
const int64 kBytesSent = 12345678901234LL;
// Construct a stats value to read.
@@ -477,23 +488,20 @@
EXPECT_CALL(*media_channel, GetStats(_))
.WillOnce(DoAll(SetArgPointee<0>(stats_read),
Return(true)));
- EXPECT_CALL(session_, GetTrackIdBySsrc(kSsrcOfTrack, _))
- .WillOnce(DoAll(SetArgPointee<1>(kTrackId),
- Return(true)));
stats.UpdateStats();
stats.GetStats(NULL, &reports);
- // |reports| should contain one session report, one track report, and one ssrc
- // report.
- EXPECT_EQ((size_t)3, reports.size());
+ // |reports| should contain at least one session report, one track report,
+ // and one ssrc report.
+ EXPECT_LE((size_t)3, reports.size());
const webrtc::StatsReport* track_report = FindNthReportByType(
reports, webrtc::StatsReport::kStatsReportTypeTrack, 1);
EXPECT_FALSE(track_report == NULL);
- stats.GetStats(track, &reports);
- // |reports| should contain one session report, one track report, and one ssrc
- // report.
- EXPECT_EQ((size_t)3, reports.size());
+ stats.GetStats(track_, &reports);
+ // |reports| should contain at least one session report, one track report,
+ // and one ssrc report.
+ EXPECT_LE((size_t)3, reports.size());
track_report = FindNthReportByType(
reports, webrtc::StatsReport::kStatsReportTypeTrack, 1);
EXPECT_FALSE(track_report == NULL);
@@ -516,13 +524,8 @@
const std::string kVcName("vcname");
cricket::VideoChannel video_channel(talk_base::Thread::Current(),
media_engine_, media_channel, &session_, kVcName, false, NULL);
- const std::string kTrackId("somename");
- talk_base::scoped_refptr<webrtc::MediaStream> stream(
- webrtc::MediaStream::Create("streamlabel"));
- talk_base::scoped_refptr<webrtc::VideoTrack> track =
- webrtc::VideoTrack::Create(kTrackId, NULL);
- stream->AddTrack(track);
- stats.AddStream(stream);
+ AddVideoTrackStats();
+ stats.AddStream(stream_);
stats.set_session(&session_);
@@ -531,7 +534,6 @@
// Constructs an ssrc stats update.
cricket::VideoSenderInfo video_sender_info;
cricket::VideoMediaInfo stats_read;
- const uint32 kSsrcOfTrack = 1234;
const int64 kBytesSent = 12345678901234LL;
// Construct a stats value to read.
@@ -544,23 +546,10 @@
EXPECT_CALL(*media_channel, GetStats(_))
.WillRepeatedly(DoAll(SetArgPointee<0>(stats_read),
Return(true)));
- EXPECT_CALL(session_, GetTrackIdBySsrc(kSsrcOfTrack, _))
- .WillOnce(DoAll(SetArgPointee<1>(kTrackId),
- Return(true)));
- // Instruct the session to return stats containing the transport channel.
- const std::string kTransportName("trspname");
- cricket::SessionStats session_stats;
- cricket::TransportStats transport_stats;
- cricket::TransportChannelStats channel_stats;
- channel_stats.component = 1;
- transport_stats.content_name = kTransportName;
- transport_stats.channel_stats.push_back(channel_stats);
-
- session_stats.transport_stats[kTransportName] = transport_stats;
- session_stats.proxy_to_transport[kVcName] = kTransportName;
+ InitSessionStats(kVcName);
EXPECT_CALL(session_, GetStats(_))
- .WillRepeatedly(DoAll(SetArgPointee<0>(session_stats),
+ .WillRepeatedly(DoAll(SetArgPointee<0>(session_stats_),
Return(true)));
stats.UpdateStats();
@@ -575,6 +564,78 @@
ASSERT_FALSE(transport_report == NULL);
}
+// This test verifies that a remote stats object will not be created for
+// an outgoing SSRC where remote stats are not returned.
+TEST_F(StatsCollectorTest, RemoteSsrcInfoIsAbsent) {
+ webrtc::StatsCollector stats; // Implementation under test.
+ MockVideoMediaChannel* media_channel = new MockVideoMediaChannel;
+ // The content_name known by the video channel.
+ const std::string kVcName("vcname");
+ cricket::VideoChannel video_channel(talk_base::Thread::Current(),
+ media_engine_, media_channel, &session_, kVcName, false, NULL);
+ AddVideoTrackStats();
+ stats.AddStream(stream_);
+
+ stats.set_session(&session_);
+
+ EXPECT_CALL(session_, video_channel())
+ .WillRepeatedly(ReturnNull());
+
+ stats.UpdateStats();
+ webrtc::StatsReports reports;
+ stats.GetStats(NULL, &reports);
+ const webrtc::StatsReport* remote_report = FindNthReportByType(reports,
+ webrtc::StatsReport::kStatsReportTypeRemoteSsrc, 1);
+ EXPECT_TRUE(remote_report == NULL);
+}
+
+// This test verifies that a remote stats object will be created for
+// an outgoing SSRC where stats are returned.
+TEST_F(StatsCollectorTest, RemoteSsrcInfoIsPresent) {
+ webrtc::StatsCollector stats; // Implementation under test.
+ MockVideoMediaChannel* media_channel = new MockVideoMediaChannel;
+ // The content_name known by the video channel.
+ const std::string kVcName("vcname");
+ cricket::VideoChannel video_channel(talk_base::Thread::Current(),
+ media_engine_, media_channel, &session_, kVcName, false, NULL);
+ AddVideoTrackStats();
+ stats.AddStream(stream_);
+
+ stats.set_session(&session_);
+
+ webrtc::StatsReports reports;
+
+ // Instruct the session to return stats containing the transport channel.
+ InitSessionStats(kVcName);
+ EXPECT_CALL(session_, GetStats(_))
+ .WillRepeatedly(DoAll(SetArgPointee<0>(session_stats_),
+ Return(true)));
+
+ // Constructs an ssrc stats update.
+ cricket::VideoMediaInfo stats_read;
+
+ cricket::SsrcReceiverInfo remote_ssrc_stats;
+ remote_ssrc_stats.timestamp = 12345.678;
+ remote_ssrc_stats.ssrc = kSsrcOfTrack;
+ cricket::VideoSenderInfo video_sender_info;
+ video_sender_info.ssrcs.push_back(kSsrcOfTrack);
+ video_sender_info.remote_stats.push_back(remote_ssrc_stats);
+ stats_read.senders.push_back(video_sender_info);
+
+ EXPECT_CALL(session_, video_channel())
+ .WillRepeatedly(Return(&video_channel));
+ EXPECT_CALL(*media_channel, GetStats(_))
+ .WillRepeatedly(DoAll(SetArgPointee<0>(stats_read),
+ Return(true)));
+
+ stats.UpdateStats();
+ stats.GetStats(NULL, &reports);
+ const webrtc::StatsReport* remote_report = FindNthReportByType(reports,
+ webrtc::StatsReport::kStatsReportTypeRemoteSsrc, 1);
+ EXPECT_FALSE(remote_report == NULL);
+ EXPECT_NE(0, remote_report->timestamp);
+}
+
// This test verifies that all chained certificates are correctly
// reported
TEST_F(StatsCollectorTest, ChainedCertificateReportsCreated) {
@@ -638,6 +699,7 @@
EXPECT_CALL(session_, GetStats(_))
.WillOnce(DoAll(SetArgPointee<0>(session_stats),
Return(true)));
+
EXPECT_CALL(session_, video_channel())
.WillRepeatedly(ReturnNull());
@@ -712,5 +774,4 @@
ASSERT_EQ(kNotFound, remote_certificate_id);
}
-
} // namespace
diff --git a/talk/app/webrtc/statstypes.h b/talk/app/webrtc/statstypes.h
index fb8c15b..e76aa86 100644
--- a/talk/app/webrtc/statstypes.h
+++ b/talk/app/webrtc/statstypes.h
@@ -88,6 +88,10 @@
// The |id| field is the SSRC in decimal form of the rtp stream.
static const char kStatsReportTypeSsrc[];
+ // StatsReport of |type| = "remoteSsrc" is statistics for a specific
+ // rtp stream, generated by the remote end of the connection.
+ static const char kStatsReportTypeRemoteSsrc[];
+
// StatsReport of |type| = "googTrack" is statistics for a specific media
// track. The |id| field is the track id.
static const char kStatsReportTypeTrack[];
@@ -137,6 +141,13 @@
static const char kStatsValueNameFrameRateReceived[];
static const char kStatsValueNameFrameRateDecoded[];
static const char kStatsValueNameFrameRateOutput[];
+ static const char kStatsValueNameDecodeMs[];
+ static const char kStatsValueNameMaxDecodeMs[];
+ static const char kStatsValueNameCurrentDelayMs[];
+ static const char kStatsValueNameTargetDelayMs[];
+ static const char kStatsValueNameJitterBufferMs[];
+ static const char kStatsValueNameMinPlayoutDelayMs[];
+ static const char kStatsValueNameRenderDelayMs[];
static const char kStatsValueNameFrameRateInput[];
static const char kStatsValueNameFrameRateSent[];
static const char kStatsValueNameFrameWidthReceived[];
diff --git a/talk/app/webrtc/test/fakeconstraints.h b/talk/app/webrtc/test/fakeconstraints.h
index 927432d..b23007e 100644
--- a/talk/app/webrtc/test/fakeconstraints.h
+++ b/talk/app/webrtc/test/fakeconstraints.h
@@ -119,7 +119,6 @@
}
void SetAllowDtlsSctpDataChannels() {
- SetMandatory(MediaConstraintsInterface::kEnableSctpDataChannels, true);
SetMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, true);
}
diff --git a/talk/app/webrtc/webrtcsdp_unittest.cc b/talk/app/webrtc/webrtcsdp_unittest.cc
index 6f2e388..777a707 100644
--- a/talk/app/webrtc/webrtcsdp_unittest.cc
+++ b/talk/app/webrtc/webrtcsdp_unittest.cc
@@ -282,7 +282,7 @@
"a=ice-ufrag:ufrag_data\r\n"
"a=ice-pwd:pwd_data\r\n"
"a=mid:data_content_name\r\n"
- "a=sctpmap:5000 webrtc-datachannel 65536\r\n";
+ "a=sctpmap:5000 webrtc-datachannel 1024\r\n";
static const char kSdpSctpDataChannelWithCandidatesString[] =
"m=application 2345 DTLS/SCTP 5000\r\n"
@@ -297,7 +297,7 @@
"a=ice-ufrag:ufrag_data\r\n"
"a=ice-pwd:pwd_data\r\n"
"a=mid:data_content_name\r\n"
- "a=sctpmap:5000 webrtc-datachannel 65536\r\n";
+ "a=sctpmap:5000 webrtc-datachannel 1024\r\n";
// One candidate reference string as per W3c spec.
diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc
index 74d1488..69a47c4 100644
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@@ -68,10 +68,7 @@
// line flag. So it is prefixed with kInternalConstraintPrefix so JS values
// will be removed.
const char MediaConstraintsInterface::kEnableSctpDataChannels[] =
- "internalSctpDataChannels";
-
-const char MediaConstraintsInterface::kInternalDisableEncryption[] =
- "internalDisableEncryption";
+ "deprecatedSctpDataChannels";
// Error messages
const char kSetLocalSdpFailed[] = "SetLocalDescription failed: ";
@@ -458,6 +455,7 @@
}
bool WebRtcSession::Initialize(
+ const PeerConnectionFactoryInterface::Options& options,
const MediaConstraintsInterface* constraints,
DTLSIdentityServiceInterface* dtls_identity_service) {
// TODO(perkj): Take |constraints| into consideration. Return false if not all
@@ -476,19 +474,16 @@
}
// Enable creation of RTP data channels if the kEnableRtpDataChannels is set.
- // It takes precendence over the kEnableSctpDataChannels constraint.
+ // It takes precendence over the disable_sctp_data_channels
+ // PeerConnectionFactoryInterface::Options.
if (FindConstraint(
constraints, MediaConstraintsInterface::kEnableRtpDataChannels,
&value, NULL) && value) {
LOG(LS_INFO) << "Allowing RTP data engine.";
data_channel_type_ = cricket::DCT_RTP;
} else {
- bool sctp_enabled = FindConstraint(
- constraints,
- MediaConstraintsInterface::kEnableSctpDataChannels,
- &value, NULL) && value;
// DTLS has to be enabled to use SCTP.
- if (sctp_enabled && dtls_enabled_) {
+ if (!options.disable_sctp_data_channels && dtls_enabled_) {
LOG(LS_INFO) << "Allowing SCTP data engine.";
data_channel_type_ = cricket::DCT_SCTP;
}
@@ -520,11 +515,7 @@
webrtc_session_desc_factory_->SignalIdentityReady.connect(
this, &WebRtcSession::OnIdentityReady);
- // Disable encryption if kDisableEncryption is set.
- if (FindConstraint(
- constraints,
- MediaConstraintsInterface::kInternalDisableEncryption,
- &value, NULL) && value) {
+ if (options.disable_encryption) {
webrtc_session_desc_factory_->set_secure(cricket::SEC_DISABLED);
}
diff --git a/talk/app/webrtc/webrtcsession.h b/talk/app/webrtc/webrtcsession.h
index 12e2291..dde33ca 100644
--- a/talk/app/webrtc/webrtcsession.h
+++ b/talk/app/webrtc/webrtcsession.h
@@ -110,7 +110,8 @@
MediaStreamSignaling* mediastream_signaling);
virtual ~WebRtcSession();
- bool Initialize(const MediaConstraintsInterface* constraints,
+ bool Initialize(const PeerConnectionFactoryInterface::Options& options,
+ const MediaConstraintsInterface* constraints,
DTLSIdentityServiceInterface* dtls_identity_service);
// Deletes the voice, video and data channel and changes the session state
// to STATE_RECEIVEDTERMINATE.
diff --git a/talk/app/webrtc/webrtcsession_unittest.cc b/talk/app/webrtc/webrtcsession_unittest.cc
index 493294f..ecbba18 100644
--- a/talk/app/webrtc/webrtcsession_unittest.cc
+++ b/talk/app/webrtc/webrtcsession_unittest.cc
@@ -81,6 +81,7 @@
using webrtc::IceCandidateCollection;
using webrtc::JsepIceCandidate;
using webrtc::JsepSessionDescription;
+using webrtc::PeerConnectionFactoryInterface;
using webrtc::PeerConnectionInterface;
using webrtc::SessionDescriptionInterface;
using webrtc::StreamCollection;
@@ -317,7 +318,8 @@
EXPECT_EQ(PeerConnectionInterface::kIceGatheringNew,
observer_.ice_gathering_state_);
- EXPECT_TRUE(session_->Initialize(constraints_.get(), identity_service));
+ EXPECT_TRUE(session_->Initialize(options_, constraints_.get(),
+ identity_service));
}
void InitWithDtmfCodec() {
@@ -919,6 +921,7 @@
cricket::TestStunServer stun_server_;
talk_base::FakeNetworkManager network_manager_;
cricket::BasicPortAllocator allocator_;
+ PeerConnectionFactoryInterface::Options options_;
talk_base::scoped_ptr<FakeConstraints> constraints_;
FakeMediaStreamSignaling mediastream_signaling_;
talk_base::scoped_ptr<WebRtcSessionForTest> session_;
@@ -1932,9 +1935,7 @@
}
TEST_F(WebRtcSessionTest, VerifyNoCryptoParamsInSDP) {
- constraints_.reset(new FakeConstraints());
- constraints_->AddOptional(
- webrtc::MediaConstraintsInterface::kInternalDisableEncryption, true);
+ options_.disable_encryption = true;
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
scoped_ptr<SessionDescriptionInterface> offer(
@@ -2424,9 +2425,7 @@
// This test verifies the crypto parameter when security is disabled.
TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescriptionWithDisabled) {
- constraints_.reset(new FakeConstraints());
- constraints_->AddOptional(
- webrtc::MediaConstraintsInterface::kInternalDisableEncryption, true);
+ options_.disable_encryption = true;
Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
talk_base::scoped_ptr<SessionDescriptionInterface> offer(
@@ -2567,8 +2566,8 @@
constraints_.reset(new FakeConstraints());
constraints_->AddOptional(
webrtc::MediaConstraintsInterface::kEnableRtpDataChannels, true);
- constraints_->AddOptional(
- webrtc::MediaConstraintsInterface::kEnableSctpDataChannels, true);
+ options_.disable_sctp_data_channels = false;
+
InitWithDtls(false);
SetLocalDescriptionWithDataChannel();
@@ -2578,9 +2577,6 @@
TEST_F(WebRtcSessionTest, TestCreateOfferWithSctpEnabledWithoutStreams) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
- constraints_.reset(new FakeConstraints());
- constraints_->AddOptional(
- webrtc::MediaConstraintsInterface::kEnableSctpDataChannels, true);
InitWithDtls(false);
talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
@@ -2591,9 +2587,6 @@
TEST_F(WebRtcSessionTest, TestCreateAnswerWithSctpInOfferAndNoStreams) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
SetFactoryDtlsSrtp();
- constraints_.reset(new FakeConstraints());
- constraints_->AddOptional(
- webrtc::MediaConstraintsInterface::kEnableSctpDataChannels, true);
InitWithDtls(false);
// Create remote offer with SCTP.
@@ -2613,8 +2606,6 @@
TEST_F(WebRtcSessionTest, TestSctpDataChannelWithoutDtls) {
constraints_.reset(new FakeConstraints());
constraints_->AddOptional(
- webrtc::MediaConstraintsInterface::kEnableSctpDataChannels, true);
- constraints_->AddOptional(
webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, false);
InitWithDtls(false);
@@ -2625,24 +2616,26 @@
TEST_F(WebRtcSessionTest, TestSctpDataChannelWithDtls) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
- constraints_.reset(new FakeConstraints());
- constraints_->AddOptional(
- webrtc::MediaConstraintsInterface::kEnableSctpDataChannels, true);
InitWithDtls(false);
SetLocalDescriptionWithDataChannel();
EXPECT_EQ(cricket::DCT_SCTP, data_engine_->last_channel_type());
}
+TEST_F(WebRtcSessionTest, TestDisableSctpDataChannels) {
+ MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
+ options_.disable_sctp_data_channels = true;
+ InitWithDtls(false);
+
+ SetLocalDescriptionWithDataChannel();
+ EXPECT_EQ(cricket::DCT_NONE, data_engine_->last_channel_type());
+}
+
TEST_F(WebRtcSessionTest, TestSctpDataChannelSendPortParsing) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
const int new_send_port = 9998;
const int new_recv_port = 7775;
- constraints_.reset(new FakeConstraints());
- constraints_->AddOptional(
- webrtc::MediaConstraintsInterface::kEnableSctpDataChannels, true);
-
InitWithDtls(false);
SetFactoryDtlsSrtp();
