Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(135)

Issues Search
    My Issues | Starred     Open | Closed | All

Unified Diff: talk/app/webrtc/mediastreamsignaling_unittest.cc

Issue 1393563002: Moving MediaStreamSignaling logic into PeerConnection. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixing copyright header Created 5 years, 2 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « talk/app/webrtc/mediastreamsignaling.cc ('k') | talk/app/webrtc/peerconnection.h » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
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 23337058d157bb9754542c0944ccdf8d3a70d632..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_t ssrc) {
- AddTrack(&local_audio_tracks_, stream, audio_track, ssrc);
- }
-
- virtual void OnAddLocalVideoTrack(MediaStreamInterface* stream,
- VideoTrackInterface* video_track,
- uint32_t ssrc) {
- AddTrack(&local_video_tracks_, stream, video_track, ssrc);
- }
-
- virtual void OnRemoveLocalAudioTrack(MediaStreamInterface* stream,
- AudioTrackInterface* audio_track,
- uint32_t 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_t ssrc) {
- AddTrack(&remote_audio_tracks_, stream, audio_track, ssrc);
- }
-
- virtual void OnAddRemoteVideoTrack(MediaStreamInterface* stream,
- VideoTrackInterface* video_track,
- uint32_t 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_t 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_t 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_t 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_t 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_t ssrc)
- : stream_label(stream_label), track_id(track_id), ssrc(ssrc) {}
- std::string stream_label;
- std::string track_id;
- uint32_t ssrc;
- };
- typedef std::vector<TrackInfo> TrackInfos;
-
- void AddTrack(TrackInfos* track_infos,
- MediaStreamInterface* stream,
- MediaStreamTrackInterface* track,
- uint32_t 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_t 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());
-}
« no previous file with comments | « talk/app/webrtc/mediastreamsignaling.cc ('k') | talk/app/webrtc/peerconnection.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698