| Index: talk/app/webrtc/mediastreamsignaling_unittest.cc
|
| diff --git a/talk/app/webrtc/mediastreamsignaling_unittest.cc b/talk/app/webrtc/mediastreamsignaling_unittest.cc
|
| deleted file mode 100644
|
| index 4f54df49b49ba079c4ffca190dbb8c32e1fd3bf0..0000000000000000000000000000000000000000
|
| --- a/talk/app/webrtc/mediastreamsignaling_unittest.cc
|
| +++ /dev/null
|
| @@ -1,1341 +0,0 @@
|
| -/*
|
| - * 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 <string>
|
| -#include <vector>
|
| -
|
| -#include "talk/app/webrtc/audiotrack.h"
|
| -#include "talk/app/webrtc/mediastream.h"
|
| -#include "talk/app/webrtc/mediastreamsignaling.h"
|
| -#include "talk/app/webrtc/sctputils.h"
|
| -#include "talk/app/webrtc/streamcollection.h"
|
| -#include "talk/app/webrtc/test/fakeconstraints.h"
|
| -#include "talk/app/webrtc/test/fakedatachannelprovider.h"
|
| -#include "talk/app/webrtc/videotrack.h"
|
| -#include "talk/media/base/fakemediaengine.h"
|
| -#include "webrtc/p2p/base/constants.h"
|
| -#include "webrtc/p2p/base/sessiondescription.h"
|
| -#include "talk/session/media/channelmanager.h"
|
| -#include "webrtc/base/gunit.h"
|
| -#include "webrtc/base/scoped_ptr.h"
|
| -#include "webrtc/base/stringutils.h"
|
| -#include "webrtc/base/thread.h"
|
| -
|
| -static const char kStreams[][8] = {"stream1", "stream2"};
|
| -static const char kAudioTracks[][32] = {"audiotrack0", "audiotrack1"};
|
| -static const char kVideoTracks[][32] = {"videotrack0", "videotrack1"};
|
| -
|
| -using webrtc::AudioTrack;
|
| -using webrtc::AudioTrackInterface;
|
| -using webrtc::AudioTrackVector;
|
| -using webrtc::VideoTrack;
|
| -using webrtc::VideoTrackInterface;
|
| -using webrtc::VideoTrackVector;
|
| -using webrtc::DataChannelInterface;
|
| -using webrtc::FakeConstraints;
|
| -using webrtc::IceCandidateInterface;
|
| -using webrtc::MediaConstraintsInterface;
|
| -using webrtc::MediaStreamInterface;
|
| -using webrtc::MediaStreamTrackInterface;
|
| -using webrtc::PeerConnectionInterface;
|
| -using webrtc::SdpParseError;
|
| -using webrtc::SessionDescriptionInterface;
|
| -using webrtc::StreamCollection;
|
| -using webrtc::StreamCollectionInterface;
|
| -
|
| -typedef PeerConnectionInterface::RTCOfferAnswerOptions RTCOfferAnswerOptions;
|
| -
|
| -// Reference SDP with a MediaStream with label "stream1" and audio track with
|
| -// id "audio_1" and a video track with id "video_1;
|
| -static const char kSdpStringWithStream1[] =
|
| - "v=0\r\n"
|
| - "o=- 0 0 IN IP4 127.0.0.1\r\n"
|
| - "s=-\r\n"
|
| - "t=0 0\r\n"
|
| - "m=audio 1 RTP/AVPF 103\r\n"
|
| - "a=mid:audio\r\n"
|
| - "a=rtpmap:103 ISAC/16000\r\n"
|
| - "a=ssrc:1 cname:stream1\r\n"
|
| - "a=ssrc:1 mslabel:stream1\r\n"
|
| - "a=ssrc:1 label:audiotrack0\r\n"
|
| - "m=video 1 RTP/AVPF 120\r\n"
|
| - "a=mid:video\r\n"
|
| - "a=rtpmap:120 VP8/90000\r\n"
|
| - "a=ssrc:2 cname:stream1\r\n"
|
| - "a=ssrc:2 mslabel:stream1\r\n"
|
| - "a=ssrc:2 label:videotrack0\r\n";
|
| -
|
| -// Reference SDP with two MediaStreams with label "stream1" and "stream2. Each
|
| -// MediaStreams have one audio track and one video track.
|
| -// This uses MSID.
|
| -static const char kSdpStringWith2Stream[] =
|
| - "v=0\r\n"
|
| - "o=- 0 0 IN IP4 127.0.0.1\r\n"
|
| - "s=-\r\n"
|
| - "t=0 0\r\n"
|
| - "a=msid-semantic: WMS stream1 stream2\r\n"
|
| - "m=audio 1 RTP/AVPF 103\r\n"
|
| - "a=mid:audio\r\n"
|
| - "a=rtpmap:103 ISAC/16000\r\n"
|
| - "a=ssrc:1 cname:stream1\r\n"
|
| - "a=ssrc:1 msid:stream1 audiotrack0\r\n"
|
| - "a=ssrc:3 cname:stream2\r\n"
|
| - "a=ssrc:3 msid:stream2 audiotrack1\r\n"
|
| - "m=video 1 RTP/AVPF 120\r\n"
|
| - "a=mid:video\r\n"
|
| - "a=rtpmap:120 VP8/0\r\n"
|
| - "a=ssrc:2 cname:stream1\r\n"
|
| - "a=ssrc:2 msid:stream1 videotrack0\r\n"
|
| - "a=ssrc:4 cname:stream2\r\n"
|
| - "a=ssrc:4 msid:stream2 videotrack1\r\n";
|
| -
|
| -// Reference SDP without MediaStreams. Msid is not supported.
|
| -static const char kSdpStringWithoutStreams[] =
|
| - "v=0\r\n"
|
| - "o=- 0 0 IN IP4 127.0.0.1\r\n"
|
| - "s=-\r\n"
|
| - "t=0 0\r\n"
|
| - "m=audio 1 RTP/AVPF 103\r\n"
|
| - "a=mid:audio\r\n"
|
| - "a=rtpmap:103 ISAC/16000\r\n"
|
| - "m=video 1 RTP/AVPF 120\r\n"
|
| - "a=mid:video\r\n"
|
| - "a=rtpmap:120 VP8/90000\r\n";
|
| -
|
| -// Reference SDP without MediaStreams. Msid is supported.
|
| -static const char kSdpStringWithMsidWithoutStreams[] =
|
| - "v=0\r\n"
|
| - "o=- 0 0 IN IP4 127.0.0.1\r\n"
|
| - "s=-\r\n"
|
| - "t=0 0\r\n"
|
| - "a=msid-semantic: WMS\r\n"
|
| - "m=audio 1 RTP/AVPF 103\r\n"
|
| - "a=mid:audio\r\n"
|
| - "a=rtpmap:103 ISAC/16000\r\n"
|
| - "m=video 1 RTP/AVPF 120\r\n"
|
| - "a=mid:video\r\n"
|
| - "a=rtpmap:120 VP8/90000\r\n";
|
| -
|
| -// Reference SDP without MediaStreams and audio only.
|
| -static const char kSdpStringWithoutStreamsAudioOnly[] =
|
| - "v=0\r\n"
|
| - "o=- 0 0 IN IP4 127.0.0.1\r\n"
|
| - "s=-\r\n"
|
| - "t=0 0\r\n"
|
| - "m=audio 1 RTP/AVPF 103\r\n"
|
| - "a=mid:audio\r\n"
|
| - "a=rtpmap:103 ISAC/16000\r\n";
|
| -
|
| -// Reference SENDONLY SDP without MediaStreams. Msid is not supported.
|
| -static const char kSdpStringSendOnlyWithWithoutStreams[] =
|
| - "v=0\r\n"
|
| - "o=- 0 0 IN IP4 127.0.0.1\r\n"
|
| - "s=-\r\n"
|
| - "t=0 0\r\n"
|
| - "m=audio 1 RTP/AVPF 103\r\n"
|
| - "a=mid:audio\r\n"
|
| - "a=sendonly"
|
| - "a=rtpmap:103 ISAC/16000\r\n"
|
| - "m=video 1 RTP/AVPF 120\r\n"
|
| - "a=mid:video\r\n"
|
| - "a=sendonly"
|
| - "a=rtpmap:120 VP8/90000\r\n";
|
| -
|
| -static const char kSdpStringInit[] =
|
| - "v=0\r\n"
|
| - "o=- 0 0 IN IP4 127.0.0.1\r\n"
|
| - "s=-\r\n"
|
| - "t=0 0\r\n"
|
| - "a=msid-semantic: WMS\r\n";
|
| -
|
| -static const char kSdpStringAudio[] =
|
| - "m=audio 1 RTP/AVPF 103\r\n"
|
| - "a=mid:audio\r\n"
|
| - "a=rtpmap:103 ISAC/16000\r\n";
|
| -
|
| -static const char kSdpStringVideo[] =
|
| - "m=video 1 RTP/AVPF 120\r\n"
|
| - "a=mid:video\r\n"
|
| - "a=rtpmap:120 VP8/90000\r\n";
|
| -
|
| -static const char kSdpStringMs1Audio0[] =
|
| - "a=ssrc:1 cname:stream1\r\n"
|
| - "a=ssrc:1 msid:stream1 audiotrack0\r\n";
|
| -
|
| -static const char kSdpStringMs1Video0[] =
|
| - "a=ssrc:2 cname:stream1\r\n"
|
| - "a=ssrc:2 msid:stream1 videotrack0\r\n";
|
| -
|
| -static const char kSdpStringMs1Audio1[] =
|
| - "a=ssrc:3 cname:stream1\r\n"
|
| - "a=ssrc:3 msid:stream1 audiotrack1\r\n";
|
| -
|
| -static const char kSdpStringMs1Video1[] =
|
| - "a=ssrc:4 cname:stream1\r\n"
|
| - "a=ssrc:4 msid:stream1 videotrack1\r\n";
|
| -
|
| -// Verifies that |options| contain all tracks in |collection| and that
|
| -// the |options| has set the the has_audio and has_video flags correct.
|
| -static void VerifyMediaOptions(StreamCollectionInterface* collection,
|
| - const cricket::MediaSessionOptions& options) {
|
| - if (!collection) {
|
| - return;
|
| - }
|
| -
|
| - size_t stream_index = 0;
|
| - for (size_t i = 0; i < collection->count(); ++i) {
|
| - MediaStreamInterface* stream = collection->at(i);
|
| - AudioTrackVector audio_tracks = stream->GetAudioTracks();
|
| - ASSERT_GE(options.streams.size(), stream_index + audio_tracks.size());
|
| - for (size_t j = 0; j < audio_tracks.size(); ++j) {
|
| - webrtc::AudioTrackInterface* audio = audio_tracks[j];
|
| - EXPECT_EQ(options.streams[stream_index].sync_label, stream->label());
|
| - EXPECT_EQ(options.streams[stream_index++].id, audio->id());
|
| - EXPECT_TRUE(options.has_audio());
|
| - }
|
| - VideoTrackVector video_tracks = stream->GetVideoTracks();
|
| - ASSERT_GE(options.streams.size(), stream_index + video_tracks.size());
|
| - for (size_t j = 0; j < video_tracks.size(); ++j) {
|
| - webrtc::VideoTrackInterface* video = video_tracks[j];
|
| - EXPECT_EQ(options.streams[stream_index].sync_label, stream->label());
|
| - EXPECT_EQ(options.streams[stream_index++].id, video->id());
|
| - EXPECT_TRUE(options.has_video());
|
| - }
|
| - }
|
| -}
|
| -
|
| -static bool CompareStreamCollections(StreamCollectionInterface* s1,
|
| - StreamCollectionInterface* s2) {
|
| - if (s1 == NULL || s2 == NULL || s1->count() != s2->count())
|
| - return false;
|
| -
|
| - for (size_t i = 0; i != s1->count(); ++i) {
|
| - if (s1->at(i)->label() != s2->at(i)->label())
|
| - return false;
|
| - webrtc::AudioTrackVector audio_tracks1 = s1->at(i)->GetAudioTracks();
|
| - webrtc::AudioTrackVector audio_tracks2 = s2->at(i)->GetAudioTracks();
|
| - webrtc::VideoTrackVector video_tracks1 = s1->at(i)->GetVideoTracks();
|
| - webrtc::VideoTrackVector video_tracks2 = s2->at(i)->GetVideoTracks();
|
| -
|
| - if (audio_tracks1.size() != audio_tracks2.size())
|
| - return false;
|
| - for (size_t j = 0; j != audio_tracks1.size(); ++j) {
|
| - if (audio_tracks1[j]->id() != audio_tracks2[j]->id())
|
| - return false;
|
| - }
|
| - if (video_tracks1.size() != video_tracks2.size())
|
| - return false;
|
| - for (size_t j = 0; j != video_tracks1.size(); ++j) {
|
| - if (video_tracks1[j]->id() != video_tracks2[j]->id())
|
| - return false;
|
| - }
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -class FakeDataChannelFactory : public webrtc::DataChannelFactory {
|
| - public:
|
| - FakeDataChannelFactory(FakeDataChannelProvider* provider,
|
| - cricket::DataChannelType dct,
|
| - webrtc::MediaStreamSignaling* media_stream_signaling)
|
| - : provider_(provider),
|
| - type_(dct),
|
| - media_stream_signaling_(media_stream_signaling) {}
|
| -
|
| - virtual rtc::scoped_refptr<webrtc::DataChannel> CreateDataChannel(
|
| - const std::string& label,
|
| - const webrtc::InternalDataChannelInit* config) {
|
| - last_init_ = *config;
|
| - rtc::scoped_refptr<webrtc::DataChannel> data_channel =
|
| - webrtc::DataChannel::Create(provider_, type_, label, *config);
|
| - media_stream_signaling_->AddDataChannel(data_channel);
|
| - return data_channel;
|
| - }
|
| -
|
| - const webrtc::InternalDataChannelInit& last_init() const {
|
| - return last_init_;
|
| - }
|
| -
|
| - private:
|
| - FakeDataChannelProvider* provider_;
|
| - cricket::DataChannelType type_;
|
| - webrtc::MediaStreamSignaling* media_stream_signaling_;
|
| - webrtc::InternalDataChannelInit last_init_;
|
| -};
|
| -
|
| -class MockSignalingObserver : public webrtc::MediaStreamSignalingObserver {
|
| - public:
|
| - MockSignalingObserver()
|
| - : remote_media_streams_(StreamCollection::Create()) {
|
| - }
|
| -
|
| - virtual ~MockSignalingObserver() {
|
| - }
|
| -
|
| - // New remote stream have been discovered.
|
| - virtual void OnAddRemoteStream(MediaStreamInterface* remote_stream) {
|
| - remote_media_streams_->AddStream(remote_stream);
|
| - }
|
| -
|
| - // Remote stream is no longer available.
|
| - virtual void OnRemoveRemoteStream(MediaStreamInterface* remote_stream) {
|
| - remote_media_streams_->RemoveStream(remote_stream);
|
| - }
|
| -
|
| - virtual void OnAddDataChannel(DataChannelInterface* data_channel) {
|
| - }
|
| -
|
| - virtual void OnAddLocalAudioTrack(MediaStreamInterface* stream,
|
| - AudioTrackInterface* audio_track,
|
| - uint32 ssrc) {
|
| - AddTrack(&local_audio_tracks_, stream, audio_track, ssrc);
|
| - }
|
| -
|
| - virtual void OnAddLocalVideoTrack(MediaStreamInterface* stream,
|
| - VideoTrackInterface* video_track,
|
| - uint32 ssrc) {
|
| - AddTrack(&local_video_tracks_, stream, video_track, ssrc);
|
| - }
|
| -
|
| - virtual void OnRemoveLocalAudioTrack(MediaStreamInterface* stream,
|
| - AudioTrackInterface* audio_track,
|
| - uint32 ssrc) {
|
| - RemoveTrack(&local_audio_tracks_, stream, audio_track);
|
| - }
|
| -
|
| - virtual void OnRemoveLocalVideoTrack(MediaStreamInterface* stream,
|
| - VideoTrackInterface* video_track) {
|
| - RemoveTrack(&local_video_tracks_, stream, video_track);
|
| - }
|
| -
|
| - virtual void OnAddRemoteAudioTrack(MediaStreamInterface* stream,
|
| - AudioTrackInterface* audio_track,
|
| - uint32 ssrc) {
|
| - AddTrack(&remote_audio_tracks_, stream, audio_track, ssrc);
|
| - }
|
| -
|
| - virtual void OnAddRemoteVideoTrack(MediaStreamInterface* stream,
|
| - VideoTrackInterface* video_track,
|
| - uint32 ssrc) {
|
| - AddTrack(&remote_video_tracks_, stream, video_track, ssrc);
|
| - }
|
| -
|
| - virtual void OnRemoveRemoteAudioTrack(MediaStreamInterface* stream,
|
| - AudioTrackInterface* audio_track) {
|
| - RemoveTrack(&remote_audio_tracks_, stream, audio_track);
|
| - }
|
| -
|
| - virtual void OnRemoveRemoteVideoTrack(MediaStreamInterface* stream,
|
| - VideoTrackInterface* video_track) {
|
| - RemoveTrack(&remote_video_tracks_, stream, video_track);
|
| - }
|
| -
|
| - virtual void OnRemoveLocalStream(MediaStreamInterface* stream) {
|
| - }
|
| -
|
| - MediaStreamInterface* RemoteStream(const std::string& label) {
|
| - return remote_media_streams_->find(label);
|
| - }
|
| -
|
| - StreamCollectionInterface* remote_streams() const {
|
| - return remote_media_streams_;
|
| - }
|
| -
|
| - size_t NumberOfRemoteAudioTracks() { return remote_audio_tracks_.size(); }
|
| -
|
| - void VerifyRemoteAudioTrack(const std::string& stream_label,
|
| - const std::string& track_id,
|
| - uint32 ssrc) {
|
| - VerifyTrack(remote_audio_tracks_, stream_label, track_id, ssrc);
|
| - }
|
| -
|
| - size_t NumberOfRemoteVideoTracks() { return remote_video_tracks_.size(); }
|
| -
|
| - void VerifyRemoteVideoTrack(const std::string& stream_label,
|
| - const std::string& track_id,
|
| - uint32 ssrc) {
|
| - VerifyTrack(remote_video_tracks_, stream_label, track_id, ssrc);
|
| - }
|
| -
|
| - size_t NumberOfLocalAudioTracks() { return local_audio_tracks_.size(); }
|
| - void VerifyLocalAudioTrack(const std::string& stream_label,
|
| - const std::string& track_id,
|
| - uint32 ssrc) {
|
| - VerifyTrack(local_audio_tracks_, stream_label, track_id, ssrc);
|
| - }
|
| -
|
| - size_t NumberOfLocalVideoTracks() { return local_video_tracks_.size(); }
|
| -
|
| - void VerifyLocalVideoTrack(const std::string& stream_label,
|
| - const std::string& track_id,
|
| - uint32 ssrc) {
|
| - VerifyTrack(local_video_tracks_, stream_label, track_id, ssrc);
|
| - }
|
| -
|
| - private:
|
| - struct TrackInfo {
|
| - TrackInfo() {}
|
| - TrackInfo(const std::string& stream_label,
|
| - const std::string track_id,
|
| - uint32 ssrc)
|
| - : stream_label(stream_label), track_id(track_id), ssrc(ssrc) {}
|
| - std::string stream_label;
|
| - std::string track_id;
|
| - uint32 ssrc;
|
| - };
|
| - typedef std::vector<TrackInfo> TrackInfos;
|
| -
|
| - void AddTrack(TrackInfos* track_infos,
|
| - MediaStreamInterface* stream,
|
| - MediaStreamTrackInterface* track,
|
| - uint32 ssrc) {
|
| - (*track_infos).push_back(TrackInfo(stream->label(), track->id(), ssrc));
|
| - }
|
| -
|
| - void RemoveTrack(TrackInfos* track_infos, MediaStreamInterface* stream,
|
| - MediaStreamTrackInterface* track) {
|
| - for (TrackInfos::iterator it = track_infos->begin();
|
| - it != track_infos->end(); ++it) {
|
| - if (it->stream_label == stream->label() && it->track_id == track->id()) {
|
| - track_infos->erase(it);
|
| - return;
|
| - }
|
| - }
|
| - ADD_FAILURE();
|
| - }
|
| -
|
| - const TrackInfo* FindTrackInfo(const TrackInfos& infos,
|
| - const std::string& stream_label,
|
| - const std::string track_id) const {
|
| - for (TrackInfos::const_iterator it = infos.begin();
|
| - it != infos.end(); ++it) {
|
| - if (it->stream_label == stream_label && it->track_id == track_id)
|
| - return &*it;
|
| - }
|
| - return NULL;
|
| - }
|
| -
|
| - void VerifyTrack(const TrackInfos& track_infos,
|
| - const std::string& stream_label,
|
| - const std::string& track_id,
|
| - uint32 ssrc) {
|
| - const TrackInfo* track_info = FindTrackInfo(track_infos,
|
| - stream_label,
|
| - track_id);
|
| - ASSERT_TRUE(track_info != NULL);
|
| - EXPECT_EQ(ssrc, track_info->ssrc);
|
| - }
|
| -
|
| - TrackInfos remote_audio_tracks_;
|
| - TrackInfos remote_video_tracks_;
|
| - TrackInfos local_audio_tracks_;
|
| - TrackInfos local_video_tracks_;
|
| -
|
| - rtc::scoped_refptr<StreamCollection> remote_media_streams_;
|
| -};
|
| -
|
| -class MediaStreamSignalingForTest : public webrtc::MediaStreamSignaling {
|
| - public:
|
| - MediaStreamSignalingForTest(MockSignalingObserver* observer,
|
| - cricket::ChannelManager* channel_manager)
|
| - : webrtc::MediaStreamSignaling(rtc::Thread::Current(), observer,
|
| - channel_manager) {
|
| - };
|
| -
|
| - using webrtc::MediaStreamSignaling::GetOptionsForOffer;
|
| - using webrtc::MediaStreamSignaling::GetOptionsForAnswer;
|
| - using webrtc::MediaStreamSignaling::OnRemoteDescriptionChanged;
|
| - using webrtc::MediaStreamSignaling::remote_streams;
|
| -};
|
| -
|
| -class MediaStreamSignalingTest: public testing::Test {
|
| - protected:
|
| - virtual void SetUp() {
|
| - observer_.reset(new MockSignalingObserver());
|
| - channel_manager_.reset(
|
| - new cricket::ChannelManager(new cricket::FakeMediaEngine(),
|
| - rtc::Thread::Current()));
|
| - signaling_.reset(new MediaStreamSignalingForTest(observer_.get(),
|
| - channel_manager_.get()));
|
| - data_channel_provider_.reset(new FakeDataChannelProvider());
|
| - }
|
| -
|
| - // Create a collection of streams.
|
| - // CreateStreamCollection(1) creates a collection that
|
| - // correspond to kSdpString1.
|
| - // CreateStreamCollection(2) correspond to kSdpString2.
|
| - rtc::scoped_refptr<StreamCollection>
|
| - CreateStreamCollection(int number_of_streams) {
|
| - rtc::scoped_refptr<StreamCollection> local_collection(
|
| - StreamCollection::Create());
|
| -
|
| - for (int i = 0; i < number_of_streams; ++i) {
|
| - rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
|
| - webrtc::MediaStream::Create(kStreams[i]));
|
| -
|
| - // Add a local audio track.
|
| - rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
|
| - webrtc::AudioTrack::Create(kAudioTracks[i], NULL));
|
| - stream->AddTrack(audio_track);
|
| -
|
| - // Add a local video track.
|
| - rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
|
| - webrtc::VideoTrack::Create(kVideoTracks[i], NULL));
|
| - stream->AddTrack(video_track);
|
| -
|
| - local_collection->AddStream(stream);
|
| - }
|
| - return local_collection;
|
| - }
|
| -
|
| - // This functions Creates a MediaStream with label kStreams[0] and
|
| - // |number_of_audio_tracks| and |number_of_video_tracks| tracks and the
|
| - // corresponding SessionDescriptionInterface. The SessionDescriptionInterface
|
| - // is returned in |desc| and the MediaStream is stored in
|
| - // |reference_collection_|
|
| - void CreateSessionDescriptionAndReference(
|
| - size_t number_of_audio_tracks,
|
| - size_t number_of_video_tracks,
|
| - SessionDescriptionInterface** desc) {
|
| - ASSERT_TRUE(desc != NULL);
|
| - ASSERT_LE(number_of_audio_tracks, 2u);
|
| - ASSERT_LE(number_of_video_tracks, 2u);
|
| -
|
| - reference_collection_ = StreamCollection::Create();
|
| - std::string sdp_ms1 = std::string(kSdpStringInit);
|
| -
|
| - std::string mediastream_label = kStreams[0];
|
| -
|
| - rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
|
| - webrtc::MediaStream::Create(mediastream_label));
|
| - reference_collection_->AddStream(stream);
|
| -
|
| - if (number_of_audio_tracks > 0) {
|
| - sdp_ms1 += std::string(kSdpStringAudio);
|
| - sdp_ms1 += std::string(kSdpStringMs1Audio0);
|
| - AddAudioTrack(kAudioTracks[0], stream);
|
| - }
|
| - if (number_of_audio_tracks > 1) {
|
| - sdp_ms1 += kSdpStringMs1Audio1;
|
| - AddAudioTrack(kAudioTracks[1], stream);
|
| - }
|
| -
|
| - if (number_of_video_tracks > 0) {
|
| - sdp_ms1 += std::string(kSdpStringVideo);
|
| - sdp_ms1 += std::string(kSdpStringMs1Video0);
|
| - AddVideoTrack(kVideoTracks[0], stream);
|
| - }
|
| - if (number_of_video_tracks > 1) {
|
| - sdp_ms1 += kSdpStringMs1Video1;
|
| - AddVideoTrack(kVideoTracks[1], stream);
|
| - }
|
| -
|
| - *desc = webrtc::CreateSessionDescription(
|
| - SessionDescriptionInterface::kOffer, sdp_ms1, NULL);
|
| - }
|
| -
|
| - void AddAudioTrack(const std::string& track_id,
|
| - MediaStreamInterface* stream) {
|
| - rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
|
| - webrtc::AudioTrack::Create(track_id, NULL));
|
| - ASSERT_TRUE(stream->AddTrack(audio_track));
|
| - }
|
| -
|
| - void AddVideoTrack(const std::string& track_id,
|
| - MediaStreamInterface* stream) {
|
| - rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
|
| - webrtc::VideoTrack::Create(track_id, NULL));
|
| - ASSERT_TRUE(stream->AddTrack(video_track));
|
| - }
|
| -
|
| - rtc::scoped_refptr<webrtc::DataChannel> AddDataChannel(
|
| - cricket::DataChannelType type, const std::string& label, int id) {
|
| - webrtc::InternalDataChannelInit config;
|
| - config.id = id;
|
| - rtc::scoped_refptr<webrtc::DataChannel> data_channel(
|
| - webrtc::DataChannel::Create(
|
| - data_channel_provider_.get(), type, label, config));
|
| - EXPECT_TRUE(data_channel.get() != NULL);
|
| - EXPECT_TRUE(signaling_->AddDataChannel(data_channel.get()));
|
| - return data_channel;
|
| - }
|
| -
|
| - // ChannelManager is used by VideoSource, so it should be released after all
|
| - // the video tracks. Put it as the first private variable should ensure that.
|
| - rtc::scoped_ptr<cricket::ChannelManager> channel_manager_;
|
| - rtc::scoped_refptr<StreamCollection> reference_collection_;
|
| - rtc::scoped_ptr<MockSignalingObserver> observer_;
|
| - rtc::scoped_ptr<MediaStreamSignalingForTest> signaling_;
|
| - rtc::scoped_ptr<FakeDataChannelProvider> data_channel_provider_;
|
| -};
|
| -
|
| -TEST_F(MediaStreamSignalingTest, GetOptionsForOfferWithInvalidAudioOption) {
|
| - RTCOfferAnswerOptions rtc_options;
|
| - rtc_options.offer_to_receive_audio = RTCOfferAnswerOptions::kUndefined - 1;
|
| -
|
| - cricket::MediaSessionOptions options;
|
| - EXPECT_FALSE(signaling_->GetOptionsForOffer(rtc_options, &options));
|
| -
|
| - rtc_options.offer_to_receive_audio =
|
| - RTCOfferAnswerOptions::kMaxOfferToReceiveMedia + 1;
|
| - EXPECT_FALSE(signaling_->GetOptionsForOffer(rtc_options, &options));
|
| -}
|
| -
|
| -
|
| -TEST_F(MediaStreamSignalingTest, GetOptionsForOfferWithInvalidVideoOption) {
|
| - RTCOfferAnswerOptions rtc_options;
|
| - rtc_options.offer_to_receive_video =
|
| - RTCOfferAnswerOptions::kUndefined - 1;
|
| -
|
| - cricket::MediaSessionOptions options;
|
| - EXPECT_FALSE(signaling_->GetOptionsForOffer(rtc_options, &options));
|
| -
|
| - rtc_options.offer_to_receive_video =
|
| - RTCOfferAnswerOptions::kMaxOfferToReceiveMedia + 1;
|
| - EXPECT_FALSE(signaling_->GetOptionsForOffer(rtc_options, &options));
|
| -}
|
| -
|
| -// Test that a MediaSessionOptions is created for an offer if
|
| -// OfferToReceiveAudio and OfferToReceiveVideo options are set but no
|
| -// MediaStreams are sent.
|
| -TEST_F(MediaStreamSignalingTest, GetMediaSessionOptionsForOfferWithAudioVideo) {
|
| - RTCOfferAnswerOptions rtc_options;
|
| - rtc_options.offer_to_receive_audio = 1;
|
| - rtc_options.offer_to_receive_video = 1;
|
| -
|
| - cricket::MediaSessionOptions options;
|
| - EXPECT_TRUE(signaling_->GetOptionsForOffer(rtc_options, &options));
|
| - EXPECT_TRUE(options.has_audio());
|
| - EXPECT_TRUE(options.has_video());
|
| - EXPECT_TRUE(options.bundle_enabled);
|
| -}
|
| -
|
| -// Test that a correct MediaSessionOptions is created for an offer if
|
| -// OfferToReceiveAudio is set but no MediaStreams are sent.
|
| -TEST_F(MediaStreamSignalingTest, GetMediaSessionOptionsForOfferWithAudio) {
|
| - RTCOfferAnswerOptions rtc_options;
|
| - rtc_options.offer_to_receive_audio = 1;
|
| -
|
| - cricket::MediaSessionOptions options;
|
| - EXPECT_TRUE(signaling_->GetOptionsForOffer(rtc_options, &options));
|
| - EXPECT_TRUE(options.has_audio());
|
| - EXPECT_FALSE(options.has_video());
|
| - EXPECT_TRUE(options.bundle_enabled);
|
| -}
|
| -
|
| -// Test that a correct MediaSessionOptions is created for an offer if
|
| -// the default OfferOptons is used or MediaStreams are sent.
|
| -TEST_F(MediaStreamSignalingTest, GetDefaultMediaSessionOptionsForOffer) {
|
| - RTCOfferAnswerOptions rtc_options;
|
| -
|
| - cricket::MediaSessionOptions options;
|
| - EXPECT_TRUE(signaling_->GetOptionsForOffer(rtc_options, &options));
|
| - EXPECT_FALSE(options.has_audio());
|
| - EXPECT_FALSE(options.has_video());
|
| - EXPECT_FALSE(options.bundle_enabled);
|
| - EXPECT_TRUE(options.vad_enabled);
|
| - EXPECT_FALSE(options.transport_options.ice_restart);
|
| -}
|
| -
|
| -// Test that a correct MediaSessionOptions is created for an offer if
|
| -// OfferToReceiveVideo is set but no MediaStreams are sent.
|
| -TEST_F(MediaStreamSignalingTest, GetMediaSessionOptionsForOfferWithVideo) {
|
| - RTCOfferAnswerOptions rtc_options;
|
| - rtc_options.offer_to_receive_audio = 0;
|
| - rtc_options.offer_to_receive_video = 1;
|
| -
|
| - cricket::MediaSessionOptions options;
|
| - EXPECT_TRUE(signaling_->GetOptionsForOffer(rtc_options, &options));
|
| - EXPECT_FALSE(options.has_audio());
|
| - EXPECT_TRUE(options.has_video());
|
| - EXPECT_TRUE(options.bundle_enabled);
|
| -}
|
| -
|
| -// Test that a correct MediaSessionOptions is created for an offer if
|
| -// UseRtpMux is set to false.
|
| -TEST_F(MediaStreamSignalingTest,
|
| - GetMediaSessionOptionsForOfferWithBundleDisabled) {
|
| - RTCOfferAnswerOptions rtc_options;
|
| - rtc_options.offer_to_receive_audio = 1;
|
| - rtc_options.offer_to_receive_video = 1;
|
| - rtc_options.use_rtp_mux = false;
|
| -
|
| - cricket::MediaSessionOptions options;
|
| - EXPECT_TRUE(signaling_->GetOptionsForOffer(rtc_options, &options));
|
| - EXPECT_TRUE(options.has_audio());
|
| - EXPECT_TRUE(options.has_video());
|
| - EXPECT_FALSE(options.bundle_enabled);
|
| -}
|
| -
|
| -// Test that a correct MediaSessionOptions is created to restart ice if
|
| -// IceRestart is set. It also tests that subsequent MediaSessionOptions don't
|
| -// have |transport_options.ice_restart| set.
|
| -TEST_F(MediaStreamSignalingTest,
|
| - GetMediaSessionOptionsForOfferWithIceRestart) {
|
| - RTCOfferAnswerOptions rtc_options;
|
| - rtc_options.ice_restart = true;
|
| -
|
| - cricket::MediaSessionOptions options;
|
| - EXPECT_TRUE(signaling_->GetOptionsForOffer(rtc_options, &options));
|
| - EXPECT_TRUE(options.transport_options.ice_restart);
|
| -
|
| - rtc_options = RTCOfferAnswerOptions();
|
| - EXPECT_TRUE(signaling_->GetOptionsForOffer(rtc_options, &options));
|
| - EXPECT_FALSE(options.transport_options.ice_restart);
|
| -}
|
| -
|
| -// Test that a correct MediaSessionOptions are created for an offer if
|
| -// a MediaStream is sent and later updated with a new track.
|
| -// MediaConstraints are not used.
|
| -TEST_F(MediaStreamSignalingTest, AddTrackToLocalMediaStream) {
|
| - RTCOfferAnswerOptions rtc_options;
|
| - rtc::scoped_refptr<StreamCollection> local_streams(
|
| - CreateStreamCollection(1));
|
| - MediaStreamInterface* local_stream = local_streams->at(0);
|
| - EXPECT_TRUE(signaling_->AddLocalStream(local_stream));
|
| - cricket::MediaSessionOptions options;
|
| - EXPECT_TRUE(signaling_->GetOptionsForOffer(rtc_options, &options));
|
| - VerifyMediaOptions(local_streams, options);
|
| -
|
| - cricket::MediaSessionOptions updated_options;
|
| - local_stream->AddTrack(AudioTrack::Create(kAudioTracks[1], NULL));
|
| - EXPECT_TRUE(signaling_->GetOptionsForOffer(rtc_options, &options));
|
| - VerifyMediaOptions(local_streams, options);
|
| -}
|
| -
|
| -// Test that the MediaConstraints in an answer don't affect if audio and video
|
| -// is offered in an offer but that if kOfferToReceiveAudio or
|
| -// kOfferToReceiveVideo constraints are true in an offer, the media type will be
|
| -// included in subsequent answers.
|
| -TEST_F(MediaStreamSignalingTest, MediaConstraintsInAnswer) {
|
| - FakeConstraints answer_c;
|
| - answer_c.SetMandatoryReceiveAudio(true);
|
| - answer_c.SetMandatoryReceiveVideo(true);
|
| -
|
| - cricket::MediaSessionOptions answer_options;
|
| - EXPECT_TRUE(signaling_->GetOptionsForAnswer(&answer_c, &answer_options));
|
| - EXPECT_TRUE(answer_options.has_audio());
|
| - EXPECT_TRUE(answer_options.has_video());
|
| -
|
| - RTCOfferAnswerOptions rtc_offer_optoins;
|
| -
|
| - cricket::MediaSessionOptions offer_options;
|
| - EXPECT_TRUE(
|
| - signaling_->GetOptionsForOffer(rtc_offer_optoins, &offer_options));
|
| - EXPECT_FALSE(offer_options.has_audio());
|
| - EXPECT_FALSE(offer_options.has_video());
|
| -
|
| - RTCOfferAnswerOptions updated_rtc_offer_optoins;
|
| - updated_rtc_offer_optoins.offer_to_receive_audio = 1;
|
| - updated_rtc_offer_optoins.offer_to_receive_video = 1;
|
| -
|
| - cricket::MediaSessionOptions updated_offer_options;
|
| - EXPECT_TRUE(signaling_->GetOptionsForOffer(updated_rtc_offer_optoins,
|
| - &updated_offer_options));
|
| - EXPECT_TRUE(updated_offer_options.has_audio());
|
| - EXPECT_TRUE(updated_offer_options.has_video());
|
| -
|
| - // Since an offer has been created with both audio and video, subsequent
|
| - // offers and answers should contain both audio and video.
|
| - // Answers will only contain the media types that exist in the offer
|
| - // regardless of the value of |updated_answer_options.has_audio| and
|
| - // |updated_answer_options.has_video|.
|
| - FakeConstraints updated_answer_c;
|
| - answer_c.SetMandatoryReceiveAudio(false);
|
| - answer_c.SetMandatoryReceiveVideo(false);
|
| -
|
| - cricket::MediaSessionOptions updated_answer_options;
|
| - EXPECT_TRUE(signaling_->GetOptionsForAnswer(&updated_answer_c,
|
| - &updated_answer_options));
|
| - EXPECT_TRUE(updated_answer_options.has_audio());
|
| - EXPECT_TRUE(updated_answer_options.has_video());
|
| -
|
| - RTCOfferAnswerOptions default_rtc_options;
|
| - EXPECT_TRUE(signaling_->GetOptionsForOffer(default_rtc_options,
|
| - &updated_offer_options));
|
| - // By default, |has_audio| or |has_video| are false if there is no media
|
| - // track.
|
| - EXPECT_FALSE(updated_offer_options.has_audio());
|
| - EXPECT_FALSE(updated_offer_options.has_video());
|
| -}
|
| -
|
| -// This test verifies that the remote MediaStreams corresponding to a received
|
| -// SDP string is created. In this test the two separate MediaStreams are
|
| -// signaled.
|
| -TEST_F(MediaStreamSignalingTest, UpdateRemoteStreams) {
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc(
|
| - webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
|
| - kSdpStringWithStream1, NULL));
|
| - EXPECT_TRUE(desc != NULL);
|
| - signaling_->OnRemoteDescriptionChanged(desc.get());
|
| -
|
| - rtc::scoped_refptr<StreamCollection> reference(
|
| - CreateStreamCollection(1));
|
| - EXPECT_TRUE(CompareStreamCollections(signaling_->remote_streams(),
|
| - reference.get()));
|
| - EXPECT_TRUE(CompareStreamCollections(observer_->remote_streams(),
|
| - reference.get()));
|
| - EXPECT_EQ(1u, observer_->NumberOfRemoteAudioTracks());
|
| - observer_->VerifyRemoteAudioTrack(kStreams[0], kAudioTracks[0], 1);
|
| - EXPECT_EQ(1u, observer_->NumberOfRemoteVideoTracks());
|
| - observer_->VerifyRemoteVideoTrack(kStreams[0], kVideoTracks[0], 2);
|
| - ASSERT_EQ(1u, observer_->remote_streams()->count());
|
| - MediaStreamInterface* remote_stream = observer_->remote_streams()->at(0);
|
| - EXPECT_TRUE(remote_stream->GetVideoTracks()[0]->GetSource() != NULL);
|
| -
|
| - // Create a session description based on another SDP with another
|
| - // MediaStream.
|
| - rtc::scoped_ptr<SessionDescriptionInterface> update_desc(
|
| - webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
|
| - kSdpStringWith2Stream, NULL));
|
| - EXPECT_TRUE(update_desc != NULL);
|
| - signaling_->OnRemoteDescriptionChanged(update_desc.get());
|
| -
|
| - rtc::scoped_refptr<StreamCollection> reference2(
|
| - CreateStreamCollection(2));
|
| - EXPECT_TRUE(CompareStreamCollections(signaling_->remote_streams(),
|
| - reference2.get()));
|
| - EXPECT_TRUE(CompareStreamCollections(observer_->remote_streams(),
|
| - reference2.get()));
|
| -
|
| - EXPECT_EQ(2u, observer_->NumberOfRemoteAudioTracks());
|
| - observer_->VerifyRemoteAudioTrack(kStreams[0], kAudioTracks[0], 1);
|
| - observer_->VerifyRemoteAudioTrack(kStreams[1], kAudioTracks[1], 3);
|
| - EXPECT_EQ(2u, observer_->NumberOfRemoteVideoTracks());
|
| - observer_->VerifyRemoteVideoTrack(kStreams[0], kVideoTracks[0], 2);
|
| - observer_->VerifyRemoteVideoTrack(kStreams[1], kVideoTracks[1], 4);
|
| -}
|
| -
|
| -// This test verifies that the remote MediaStreams corresponding to a received
|
| -// SDP string is created. In this test the same remote MediaStream is signaled
|
| -// but MediaStream tracks are added and removed.
|
| -TEST_F(MediaStreamSignalingTest, AddRemoveTrackFromExistingRemoteMediaStream) {
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc_ms1;
|
| - CreateSessionDescriptionAndReference(1, 1, desc_ms1.use());
|
| - signaling_->OnRemoteDescriptionChanged(desc_ms1.get());
|
| - EXPECT_TRUE(CompareStreamCollections(signaling_->remote_streams(),
|
| - reference_collection_));
|
| -
|
| - // Add extra audio and video tracks to the same MediaStream.
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc_ms1_two_tracks;
|
| - CreateSessionDescriptionAndReference(2, 2, desc_ms1_two_tracks.use());
|
| - signaling_->OnRemoteDescriptionChanged(desc_ms1_two_tracks.get());
|
| - EXPECT_TRUE(CompareStreamCollections(signaling_->remote_streams(),
|
| - reference_collection_));
|
| - EXPECT_TRUE(CompareStreamCollections(observer_->remote_streams(),
|
| - reference_collection_));
|
| -
|
| - // Remove the extra audio and video tracks again.
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc_ms2;
|
| - CreateSessionDescriptionAndReference(1, 1, desc_ms2.use());
|
| - signaling_->OnRemoteDescriptionChanged(desc_ms2.get());
|
| - EXPECT_TRUE(CompareStreamCollections(signaling_->remote_streams(),
|
| - reference_collection_));
|
| - EXPECT_TRUE(CompareStreamCollections(observer_->remote_streams(),
|
| - reference_collection_));
|
| -}
|
| -
|
| -// This test that remote tracks are ended if a
|
| -// local session description is set that rejects the media content type.
|
| -TEST_F(MediaStreamSignalingTest, RejectMediaContent) {
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc(
|
| - webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
|
| - kSdpStringWithStream1, NULL));
|
| - EXPECT_TRUE(desc != NULL);
|
| - signaling_->OnRemoteDescriptionChanged(desc.get());
|
| -
|
| - ASSERT_EQ(1u, observer_->remote_streams()->count());
|
| - MediaStreamInterface* remote_stream = observer_->remote_streams()->at(0);
|
| - ASSERT_EQ(1u, remote_stream->GetVideoTracks().size());
|
| - ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
|
| -
|
| - rtc::scoped_refptr<webrtc::VideoTrackInterface> remote_video =
|
| - remote_stream->GetVideoTracks()[0];
|
| - EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_video->state());
|
| - rtc::scoped_refptr<webrtc::AudioTrackInterface> remote_audio =
|
| - remote_stream->GetAudioTracks()[0];
|
| - EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_audio->state());
|
| -
|
| - cricket::ContentInfo* video_info =
|
| - desc->description()->GetContentByName("video");
|
| - ASSERT_TRUE(video_info != NULL);
|
| - video_info->rejected = true;
|
| - signaling_->OnLocalDescriptionChanged(desc.get());
|
| - EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, remote_video->state());
|
| - EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_audio->state());
|
| -
|
| - cricket::ContentInfo* audio_info =
|
| - desc->description()->GetContentByName("audio");
|
| - ASSERT_TRUE(audio_info != NULL);
|
| - audio_info->rejected = true;
|
| - signaling_->OnLocalDescriptionChanged(desc.get());
|
| - EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, remote_audio->state());
|
| -}
|
| -
|
| -// This test that it won't crash if the remote track as been removed outside
|
| -// of MediaStreamSignaling and then MediaStreamSignaling tries to reject
|
| -// this track.
|
| -TEST_F(MediaStreamSignalingTest, RemoveTrackThenRejectMediaContent) {
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc(
|
| - webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
|
| - kSdpStringWithStream1, NULL));
|
| - EXPECT_TRUE(desc != NULL);
|
| - signaling_->OnRemoteDescriptionChanged(desc.get());
|
| -
|
| - MediaStreamInterface* remote_stream = observer_->remote_streams()->at(0);
|
| - remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0]);
|
| - remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
|
| -
|
| - cricket::ContentInfo* video_info =
|
| - desc->description()->GetContentByName("video");
|
| - video_info->rejected = true;
|
| - signaling_->OnLocalDescriptionChanged(desc.get());
|
| -
|
| - cricket::ContentInfo* audio_info =
|
| - desc->description()->GetContentByName("audio");
|
| - audio_info->rejected = true;
|
| - signaling_->OnLocalDescriptionChanged(desc.get());
|
| -
|
| - // No crash is a pass.
|
| -}
|
| -
|
| -// This tests that a default MediaStream is created if a remote session
|
| -// description doesn't contain any streams and no MSID support.
|
| -// It also tests that the default stream is updated if a video m-line is added
|
| -// in a subsequent session description.
|
| -TEST_F(MediaStreamSignalingTest, SdpWithoutMsidCreatesDefaultStream) {
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc_audio_only(
|
| - webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
|
| - kSdpStringWithoutStreamsAudioOnly,
|
| - NULL));
|
| - ASSERT_TRUE(desc_audio_only != NULL);
|
| - signaling_->OnRemoteDescriptionChanged(desc_audio_only.get());
|
| -
|
| - EXPECT_EQ(1u, signaling_->remote_streams()->count());
|
| - ASSERT_EQ(1u, observer_->remote_streams()->count());
|
| - MediaStreamInterface* remote_stream = observer_->remote_streams()->at(0);
|
| -
|
| - EXPECT_EQ(1u, remote_stream->GetAudioTracks().size());
|
| - EXPECT_EQ(0u, remote_stream->GetVideoTracks().size());
|
| - EXPECT_EQ("default", remote_stream->label());
|
| -
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc(
|
| - webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
|
| - kSdpStringWithoutStreams, NULL));
|
| - ASSERT_TRUE(desc != NULL);
|
| - signaling_->OnRemoteDescriptionChanged(desc.get());
|
| - EXPECT_EQ(1u, signaling_->remote_streams()->count());
|
| - ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
|
| - EXPECT_EQ("defaulta0", remote_stream->GetAudioTracks()[0]->id());
|
| - ASSERT_EQ(1u, remote_stream->GetVideoTracks().size());
|
| - EXPECT_EQ("defaultv0", remote_stream->GetVideoTracks()[0]->id());
|
| - observer_->VerifyRemoteAudioTrack("default", "defaulta0", 0);
|
| - observer_->VerifyRemoteVideoTrack("default", "defaultv0", 0);
|
| -}
|
| -
|
| -// This tests that a default MediaStream is created if a remote session
|
| -// description doesn't contain any streams and media direction is send only.
|
| -TEST_F(MediaStreamSignalingTest, RecvOnlySdpWithoutMsidCreatesDefaultStream) {
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc(
|
| - webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
|
| - kSdpStringSendOnlyWithWithoutStreams,
|
| - NULL));
|
| - ASSERT_TRUE(desc != NULL);
|
| - signaling_->OnRemoteDescriptionChanged(desc.get());
|
| -
|
| - EXPECT_EQ(1u, signaling_->remote_streams()->count());
|
| - ASSERT_EQ(1u, observer_->remote_streams()->count());
|
| - MediaStreamInterface* remote_stream = observer_->remote_streams()->at(0);
|
| -
|
| - EXPECT_EQ(1u, remote_stream->GetAudioTracks().size());
|
| - EXPECT_EQ(1u, remote_stream->GetVideoTracks().size());
|
| - EXPECT_EQ("default", remote_stream->label());
|
| -}
|
| -
|
| -// This tests that it won't crash when MediaStreamSignaling tries to remove
|
| -// a remote track that as already been removed from the mediastream.
|
| -TEST_F(MediaStreamSignalingTest, RemoveAlreadyGoneRemoteStream) {
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc_audio_only(
|
| - webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
|
| - kSdpStringWithoutStreams,
|
| - NULL));
|
| - ASSERT_TRUE(desc_audio_only != NULL);
|
| - signaling_->OnRemoteDescriptionChanged(desc_audio_only.get());
|
| - MediaStreamInterface* remote_stream = observer_->remote_streams()->at(0);
|
| - remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
|
| - remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0]);
|
| -
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc(
|
| - webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
|
| - kSdpStringWithoutStreams, NULL));
|
| - ASSERT_TRUE(desc != NULL);
|
| - signaling_->OnRemoteDescriptionChanged(desc.get());
|
| -
|
| - // No crash is a pass.
|
| -}
|
| -
|
| -// This tests that a default MediaStream is created if the remote session
|
| -// description doesn't contain any streams and don't contain an indication if
|
| -// MSID is supported.
|
| -TEST_F(MediaStreamSignalingTest,
|
| - SdpWithoutMsidAndStreamsCreatesDefaultStream) {
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc(
|
| - webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
|
| - kSdpStringWithoutStreams,
|
| - NULL));
|
| - ASSERT_TRUE(desc != NULL);
|
| - signaling_->OnRemoteDescriptionChanged(desc.get());
|
| -
|
| - ASSERT_EQ(1u, observer_->remote_streams()->count());
|
| - MediaStreamInterface* remote_stream = observer_->remote_streams()->at(0);
|
| - EXPECT_EQ(1u, remote_stream->GetAudioTracks().size());
|
| - EXPECT_EQ(1u, remote_stream->GetVideoTracks().size());
|
| -}
|
| -
|
| -// This tests that a default MediaStream is not created if the remote session
|
| -// description doesn't contain any streams but does support MSID.
|
| -TEST_F(MediaStreamSignalingTest, SdpWitMsidDontCreatesDefaultStream) {
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc_msid_without_streams(
|
| - webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
|
| - kSdpStringWithMsidWithoutStreams,
|
| - NULL));
|
| - signaling_->OnRemoteDescriptionChanged(desc_msid_without_streams.get());
|
| - EXPECT_EQ(0u, observer_->remote_streams()->count());
|
| -}
|
| -
|
| -// This test that a default MediaStream is not created if a remote session
|
| -// description is updated to not have any MediaStreams.
|
| -TEST_F(MediaStreamSignalingTest, VerifyDefaultStreamIsNotCreated) {
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc(
|
| - webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
|
| - kSdpStringWithStream1,
|
| - NULL));
|
| - ASSERT_TRUE(desc != NULL);
|
| - signaling_->OnRemoteDescriptionChanged(desc.get());
|
| - rtc::scoped_refptr<StreamCollection> reference(
|
| - CreateStreamCollection(1));
|
| - EXPECT_TRUE(CompareStreamCollections(observer_->remote_streams(),
|
| - reference.get()));
|
| -
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc_without_streams(
|
| - webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
|
| - kSdpStringWithoutStreams,
|
| - NULL));
|
| - signaling_->OnRemoteDescriptionChanged(desc_without_streams.get());
|
| - EXPECT_EQ(0u, observer_->remote_streams()->count());
|
| -}
|
| -
|
| -// This test that the correct MediaStreamSignalingObserver methods are called
|
| -// when MediaStreamSignaling::OnLocalDescriptionChanged is called with an
|
| -// updated local session description.
|
| -TEST_F(MediaStreamSignalingTest, LocalDescriptionChanged) {
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc_1;
|
| - CreateSessionDescriptionAndReference(2, 2, desc_1.use());
|
| -
|
| - signaling_->AddLocalStream(reference_collection_->at(0));
|
| - signaling_->OnLocalDescriptionChanged(desc_1.get());
|
| - EXPECT_EQ(2u, observer_->NumberOfLocalAudioTracks());
|
| - EXPECT_EQ(2u, observer_->NumberOfLocalVideoTracks());
|
| - observer_->VerifyLocalAudioTrack(kStreams[0], kAudioTracks[0], 1);
|
| - observer_->VerifyLocalVideoTrack(kStreams[0], kVideoTracks[0], 2);
|
| - observer_->VerifyLocalAudioTrack(kStreams[0], kAudioTracks[1], 3);
|
| - observer_->VerifyLocalVideoTrack(kStreams[0], kVideoTracks[1], 4);
|
| -
|
| - // Remove an audio and video track.
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc_2;
|
| - CreateSessionDescriptionAndReference(1, 1, desc_2.use());
|
| - signaling_->OnLocalDescriptionChanged(desc_2.get());
|
| - EXPECT_EQ(1u, observer_->NumberOfLocalAudioTracks());
|
| - EXPECT_EQ(1u, observer_->NumberOfLocalVideoTracks());
|
| - observer_->VerifyLocalAudioTrack(kStreams[0], kAudioTracks[0], 1);
|
| - observer_->VerifyLocalVideoTrack(kStreams[0], kVideoTracks[0], 2);
|
| -}
|
| -
|
| -// This test that the correct MediaStreamSignalingObserver methods are called
|
| -// when MediaStreamSignaling::AddLocalStream is called after
|
| -// MediaStreamSignaling::OnLocalDescriptionChanged is called.
|
| -TEST_F(MediaStreamSignalingTest, AddLocalStreamAfterLocalDescriptionChanged) {
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc_1;
|
| - CreateSessionDescriptionAndReference(2, 2, desc_1.use());
|
| -
|
| - signaling_->OnLocalDescriptionChanged(desc_1.get());
|
| - EXPECT_EQ(0u, observer_->NumberOfLocalAudioTracks());
|
| - EXPECT_EQ(0u, observer_->NumberOfLocalVideoTracks());
|
| -
|
| - signaling_->AddLocalStream(reference_collection_->at(0));
|
| - EXPECT_EQ(2u, observer_->NumberOfLocalAudioTracks());
|
| - EXPECT_EQ(2u, observer_->NumberOfLocalVideoTracks());
|
| - observer_->VerifyLocalAudioTrack(kStreams[0], kAudioTracks[0], 1);
|
| - observer_->VerifyLocalVideoTrack(kStreams[0], kVideoTracks[0], 2);
|
| - observer_->VerifyLocalAudioTrack(kStreams[0], kAudioTracks[1], 3);
|
| - observer_->VerifyLocalVideoTrack(kStreams[0], kVideoTracks[1], 4);
|
| -}
|
| -
|
| -// This test that the correct MediaStreamSignalingObserver methods are called
|
| -// if the ssrc on a local track is changed when
|
| -// MediaStreamSignaling::OnLocalDescriptionChanged is called.
|
| -TEST_F(MediaStreamSignalingTest, ChangeSsrcOnTrackInLocalSessionDescription) {
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc;
|
| - CreateSessionDescriptionAndReference(1, 1, desc.use());
|
| -
|
| - signaling_->AddLocalStream(reference_collection_->at(0));
|
| - signaling_->OnLocalDescriptionChanged(desc.get());
|
| - EXPECT_EQ(1u, observer_->NumberOfLocalAudioTracks());
|
| - EXPECT_EQ(1u, observer_->NumberOfLocalVideoTracks());
|
| - observer_->VerifyLocalAudioTrack(kStreams[0], kAudioTracks[0], 1);
|
| - observer_->VerifyLocalVideoTrack(kStreams[0], kVideoTracks[0], 2);
|
| -
|
| - // Change the ssrc of the audio and video track.
|
| - std::string sdp;
|
| - desc->ToString(&sdp);
|
| - std::string ssrc_org = "a=ssrc:1";
|
| - std::string ssrc_to = "a=ssrc:97";
|
| - rtc::replace_substrs(ssrc_org.c_str(), ssrc_org.length(),
|
| - ssrc_to.c_str(), ssrc_to.length(),
|
| - &sdp);
|
| - ssrc_org = "a=ssrc:2";
|
| - ssrc_to = "a=ssrc:98";
|
| - rtc::replace_substrs(ssrc_org.c_str(), ssrc_org.length(),
|
| - ssrc_to.c_str(), ssrc_to.length(),
|
| - &sdp);
|
| - rtc::scoped_ptr<SessionDescriptionInterface> updated_desc(
|
| - webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
|
| - sdp, NULL));
|
| -
|
| - signaling_->OnLocalDescriptionChanged(updated_desc.get());
|
| - EXPECT_EQ(1u, observer_->NumberOfLocalAudioTracks());
|
| - EXPECT_EQ(1u, observer_->NumberOfLocalVideoTracks());
|
| - observer_->VerifyLocalAudioTrack(kStreams[0], kAudioTracks[0], 97);
|
| - observer_->VerifyLocalVideoTrack(kStreams[0], kVideoTracks[0], 98);
|
| -}
|
| -
|
| -// This test that the correct MediaStreamSignalingObserver methods are called
|
| -// if a new session description is set with the same tracks but they are now
|
| -// sent on a another MediaStream.
|
| -TEST_F(MediaStreamSignalingTest, SignalSameTracksInSeparateMediaStream) {
|
| - rtc::scoped_ptr<SessionDescriptionInterface> desc;
|
| - CreateSessionDescriptionAndReference(1, 1, desc.use());
|
| -
|
| - signaling_->AddLocalStream(reference_collection_->at(0));
|
| - signaling_->OnLocalDescriptionChanged(desc.get());
|
| - EXPECT_EQ(1u, observer_->NumberOfLocalAudioTracks());
|
| - EXPECT_EQ(1u, observer_->NumberOfLocalVideoTracks());
|
| -
|
| - std::string stream_label_0 = kStreams[0];
|
| - observer_->VerifyLocalAudioTrack(stream_label_0, kAudioTracks[0], 1);
|
| - observer_->VerifyLocalVideoTrack(stream_label_0, kVideoTracks[0], 2);
|
| -
|
| - // Add a new MediaStream but with the same tracks as in the first stream.
|
| - std::string stream_label_1 = kStreams[1];
|
| - rtc::scoped_refptr<webrtc::MediaStreamInterface> stream_1(
|
| - webrtc::MediaStream::Create(kStreams[1]));
|
| - stream_1->AddTrack(reference_collection_->at(0)->GetVideoTracks()[0]);
|
| - stream_1->AddTrack(reference_collection_->at(0)->GetAudioTracks()[0]);
|
| - signaling_->AddLocalStream(stream_1);
|
| -
|
| - // Replace msid in the original SDP.
|
| - std::string sdp;
|
| - desc->ToString(&sdp);
|
| - rtc::replace_substrs(
|
| - kStreams[0], strlen(kStreams[0]), kStreams[1], strlen(kStreams[1]), &sdp);
|
| -
|
| - rtc::scoped_ptr<SessionDescriptionInterface> updated_desc(
|
| - webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
|
| - sdp, NULL));
|
| -
|
| - signaling_->OnLocalDescriptionChanged(updated_desc.get());
|
| - observer_->VerifyLocalAudioTrack(kStreams[1], kAudioTracks[0], 1);
|
| - observer_->VerifyLocalVideoTrack(kStreams[1], kVideoTracks[0], 2);
|
| - EXPECT_EQ(1u, observer_->NumberOfLocalAudioTracks());
|
| - EXPECT_EQ(1u, observer_->NumberOfLocalVideoTracks());
|
| -}
|
| -
|
| -// Verifies that an even SCTP id is allocated for SSL_CLIENT and an odd id for
|
| -// SSL_SERVER.
|
| -TEST_F(MediaStreamSignalingTest, SctpIdAllocationBasedOnRole) {
|
| - int id;
|
| - ASSERT_TRUE(signaling_->AllocateSctpSid(rtc::SSL_SERVER, &id));
|
| - EXPECT_EQ(1, id);
|
| - ASSERT_TRUE(signaling_->AllocateSctpSid(rtc::SSL_CLIENT, &id));
|
| - EXPECT_EQ(0, id);
|
| - ASSERT_TRUE(signaling_->AllocateSctpSid(rtc::SSL_SERVER, &id));
|
| - EXPECT_EQ(3, id);
|
| - ASSERT_TRUE(signaling_->AllocateSctpSid(rtc::SSL_CLIENT, &id));
|
| - EXPECT_EQ(2, id);
|
| -}
|
| -
|
| -// Verifies that SCTP ids of existing DataChannels are not reused.
|
| -TEST_F(MediaStreamSignalingTest, SctpIdAllocationNoReuse) {
|
| - int old_id = 1;
|
| - AddDataChannel(cricket::DCT_SCTP, "a", old_id);
|
| -
|
| - int new_id;
|
| - ASSERT_TRUE(signaling_->AllocateSctpSid(rtc::SSL_SERVER, &new_id));
|
| - EXPECT_NE(old_id, new_id);
|
| -
|
| - // Creates a DataChannel with id 0.
|
| - old_id = 0;
|
| - AddDataChannel(cricket::DCT_SCTP, "a", old_id);
|
| - ASSERT_TRUE(signaling_->AllocateSctpSid(rtc::SSL_CLIENT, &new_id));
|
| - EXPECT_NE(old_id, new_id);
|
| -}
|
| -
|
| -// Verifies that SCTP ids of removed DataChannels can be reused.
|
| -TEST_F(MediaStreamSignalingTest, SctpIdReusedForRemovedDataChannel) {
|
| - int odd_id = 1;
|
| - int even_id = 0;
|
| - AddDataChannel(cricket::DCT_SCTP, "a", odd_id);
|
| - AddDataChannel(cricket::DCT_SCTP, "a", even_id);
|
| -
|
| - int allocated_id = -1;
|
| - ASSERT_TRUE(signaling_->AllocateSctpSid(rtc::SSL_SERVER,
|
| - &allocated_id));
|
| - EXPECT_EQ(odd_id + 2, allocated_id);
|
| - AddDataChannel(cricket::DCT_SCTP, "a", allocated_id);
|
| -
|
| - ASSERT_TRUE(signaling_->AllocateSctpSid(rtc::SSL_CLIENT,
|
| - &allocated_id));
|
| - EXPECT_EQ(even_id + 2, allocated_id);
|
| - AddDataChannel(cricket::DCT_SCTP, "a", allocated_id);
|
| -
|
| - signaling_->RemoveSctpDataChannel(odd_id);
|
| - signaling_->RemoveSctpDataChannel(even_id);
|
| -
|
| - // Verifies that removed DataChannel ids are reused.
|
| - ASSERT_TRUE(signaling_->AllocateSctpSid(rtc::SSL_SERVER,
|
| - &allocated_id));
|
| - EXPECT_EQ(odd_id, allocated_id);
|
| -
|
| - ASSERT_TRUE(signaling_->AllocateSctpSid(rtc::SSL_CLIENT,
|
| - &allocated_id));
|
| - EXPECT_EQ(even_id, allocated_id);
|
| -
|
| - // Verifies that used higher DataChannel ids are not reused.
|
| - ASSERT_TRUE(signaling_->AllocateSctpSid(rtc::SSL_SERVER,
|
| - &allocated_id));
|
| - EXPECT_NE(odd_id + 2, allocated_id);
|
| -
|
| - ASSERT_TRUE(signaling_->AllocateSctpSid(rtc::SSL_CLIENT,
|
| - &allocated_id));
|
| - EXPECT_NE(even_id + 2, allocated_id);
|
| -
|
| -}
|
| -
|
| -// Verifies that duplicated label is not allowed for RTP data channel.
|
| -TEST_F(MediaStreamSignalingTest, RtpDuplicatedLabelNotAllowed) {
|
| - AddDataChannel(cricket::DCT_RTP, "a", -1);
|
| -
|
| - webrtc::InternalDataChannelInit config;
|
| - rtc::scoped_refptr<webrtc::DataChannel> data_channel =
|
| - webrtc::DataChannel::Create(
|
| - data_channel_provider_.get(), cricket::DCT_RTP, "a", config);
|
| - ASSERT_TRUE(data_channel.get() != NULL);
|
| - EXPECT_FALSE(signaling_->AddDataChannel(data_channel.get()));
|
| -}
|
| -
|
| -// Verifies that duplicated label is allowed for SCTP data channel.
|
| -TEST_F(MediaStreamSignalingTest, SctpDuplicatedLabelAllowed) {
|
| - AddDataChannel(cricket::DCT_SCTP, "a", -1);
|
| - AddDataChannel(cricket::DCT_SCTP, "a", -1);
|
| -}
|
| -
|
| -// Verifies the correct configuration is used to create DataChannel from an OPEN
|
| -// message.
|
| -TEST_F(MediaStreamSignalingTest, CreateDataChannelFromOpenMessage) {
|
| - FakeDataChannelFactory fake_factory(data_channel_provider_.get(),
|
| - cricket::DCT_SCTP,
|
| - signaling_.get());
|
| - signaling_->SetDataChannelFactory(&fake_factory);
|
| - webrtc::DataChannelInit config;
|
| - config.id = 1;
|
| - rtc::Buffer payload;
|
| - webrtc::WriteDataChannelOpenMessage("a", config, &payload);
|
| - cricket::ReceiveDataParams params;
|
| - params.ssrc = config.id;
|
| - EXPECT_TRUE(signaling_->AddDataChannelFromOpenMessage(params, payload));
|
| - EXPECT_EQ(config.id, fake_factory.last_init().id);
|
| - EXPECT_FALSE(fake_factory.last_init().negotiated);
|
| - EXPECT_EQ(webrtc::InternalDataChannelInit::kAcker,
|
| - fake_factory.last_init().open_handshake_role);
|
| -}
|
| -
|
| -// Verifies that duplicated label from OPEN message is allowed.
|
| -TEST_F(MediaStreamSignalingTest, DuplicatedLabelFromOpenMessageAllowed) {
|
| - AddDataChannel(cricket::DCT_SCTP, "a", -1);
|
| -
|
| - FakeDataChannelFactory fake_factory(data_channel_provider_.get(),
|
| - cricket::DCT_SCTP,
|
| - signaling_.get());
|
| - signaling_->SetDataChannelFactory(&fake_factory);
|
| - webrtc::DataChannelInit config;
|
| - config.id = 0;
|
| - rtc::Buffer payload;
|
| - webrtc::WriteDataChannelOpenMessage("a", config, &payload);
|
| - cricket::ReceiveDataParams params;
|
| - params.ssrc = config.id;
|
| - EXPECT_TRUE(signaling_->AddDataChannelFromOpenMessage(params, payload));
|
| -}
|
| -
|
| -// Verifies that a DataChannel closed remotely is closed locally.
|
| -TEST_F(MediaStreamSignalingTest,
|
| - SctpDataChannelClosedLocallyWhenClosedRemotely) {
|
| - webrtc::InternalDataChannelInit config;
|
| - config.id = 0;
|
| -
|
| - rtc::scoped_refptr<webrtc::DataChannel> data_channel =
|
| - webrtc::DataChannel::Create(
|
| - data_channel_provider_.get(), cricket::DCT_SCTP, "a", config);
|
| - ASSERT_TRUE(data_channel.get() != NULL);
|
| - EXPECT_EQ(webrtc::DataChannelInterface::kConnecting,
|
| - data_channel->state());
|
| -
|
| - EXPECT_TRUE(signaling_->AddDataChannel(data_channel.get()));
|
| -
|
| - signaling_->OnRemoteSctpDataChannelClosed(config.id);
|
| - EXPECT_EQ(webrtc::DataChannelInterface::kClosed, data_channel->state());
|
| -}
|
| -
|
| -// Verifies that DataChannel added from OPEN message is added to
|
| -// MediaStreamSignaling only once (webrtc issue 3778).
|
| -TEST_F(MediaStreamSignalingTest, DataChannelFromOpenMessageAddedOnce) {
|
| - FakeDataChannelFactory fake_factory(data_channel_provider_.get(),
|
| - cricket::DCT_SCTP,
|
| - signaling_.get());
|
| - signaling_->SetDataChannelFactory(&fake_factory);
|
| - webrtc::DataChannelInit config;
|
| - config.id = 1;
|
| - rtc::Buffer payload;
|
| - webrtc::WriteDataChannelOpenMessage("a", config, &payload);
|
| - cricket::ReceiveDataParams params;
|
| - params.ssrc = config.id;
|
| - EXPECT_TRUE(signaling_->AddDataChannelFromOpenMessage(params, payload));
|
| - EXPECT_TRUE(signaling_->HasDataChannels());
|
| -
|
| - // Removes the DataChannel and verifies that no DataChannel is left.
|
| - signaling_->RemoveSctpDataChannel(config.id);
|
| - EXPECT_FALSE(signaling_->HasDataChannels());
|
| -}
|
|
|