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Update talk to 51242664.

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TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2090005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4594 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrike@webrtc.org 2013-08-22 14:45:42 +00:00
parent c095f510b6
commit 7666db79fa
5 changed files with 79 additions and 23 deletions
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28
talk/app/webrtc/mediastreamsignaling.cc
View File

@ -599,15 +599,19 @@ void MediaStreamSignaling::OnRemoteTrackRemoved(
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
talk_base::scoped_refptr<AudioTrackInterface> audio_track =
stream->FindAudioTrack(track_id);
audio_track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
stream->RemoveTrack(audio_track);
stream_observer_->OnRemoveRemoteAudioTrack(stream, audio_track);
if (audio_track) {
audio_track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
stream->RemoveTrack(audio_track);
stream_observer_->OnRemoveRemoteAudioTrack(stream, audio_track);
}
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
talk_base::scoped_refptr<VideoTrackInterface> video_track =
stream->FindVideoTrack(track_id);
video_track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
stream->RemoveTrack(video_track);
stream_observer_->OnRemoveRemoteVideoTrack(stream, video_track);
if (video_track) {
video_track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
stream->RemoveTrack(video_track);
stream_observer_->OnRemoveRemoteVideoTrack(stream, video_track);
}
} else {
ASSERT(false && "Invalid media type");
}
@ -621,11 +625,19 @@ void MediaStreamSignaling::RejectRemoteTracks(cricket::MediaType media_type) {
MediaStreamInterface* stream = remote_streams_->find(info.stream_label);
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
AudioTrackInterface* track = stream->FindAudioTrack(info.track_id);
track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
// There's no guarantee the track is still available, e.g. the track may
// have been removed from the stream by javascript.
if (track) {
track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
}
}
if (media_type == cricket::MEDIA_TYPE_VIDEO) {
VideoTrackInterface* track = stream->FindVideoTrack(info.track_id);
track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
// There's no guarantee the track is still available, e.g. the track may
// have been removed from the stream by javascript.
if (track) {
track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
}
}
}
}

49
talk/app/webrtc/mediastreamsignaling_unittest.cc
View File

@ -778,6 +778,33 @@ TEST_F(MediaStreamSignalingTest, RejectMediaContent) {
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) {
talk_base::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
@ -812,6 +839,28 @@ TEST_F(MediaStreamSignalingTest, SdpWithoutMsidCreatesDefaultStream) {
observer_->VerifyRemoteVideoTrack("default", "defaultv0", 0);
}
// 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) {
talk_base::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]);
talk_base::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.

20
talk/app/webrtc/webrtcsession_unittest.cc
View File

@ -313,15 +313,7 @@ class WebRtcSessionTest : public testing::Test {
Init(NULL);
}
void InitWithDtls() {
constraints_.reset(new FakeConstraints());
constraints_->AddOptional(
webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, true);
Init(NULL);
}
void InitWithAsyncDtls(bool identity_request_should_fail) {
void InitWithDtls(bool identity_request_should_fail = false) {
constraints_.reset(new FakeConstraints());
constraints_->AddOptional(
webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, true);
@ -819,7 +811,7 @@ class WebRtcSessionTest : public testing::Test {
void VerifyMultipleAsyncCreateDescription(
bool success, CreateSessionDescriptionRequest::Type type) {
InitWithAsyncDtls(!success);
InitWithDtls(!success);
if (type == CreateSessionDescriptionRequest::kAnswer) {
cricket::MediaSessionOptions options;
@ -2526,7 +2518,7 @@ TEST_F(WebRtcSessionTest, TestSctpDataChannelWithDtls) {
// identity generation is finished.
TEST_F(WebRtcSessionTest, TestCreateOfferBeforeIdentityRequestReturnSuccess) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
InitWithAsyncDtls(false);
InitWithDtls(false);
EXPECT_TRUE(session_->waiting_for_identity());
talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
@ -2537,7 +2529,7 @@ TEST_F(WebRtcSessionTest, TestCreateOfferBeforeIdentityRequestReturnSuccess) {
// identity generation is finished.
TEST_F(WebRtcSessionTest, TestCreateAnswerBeforeIdentityRequestReturnSuccess) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
InitWithAsyncDtls(false);
InitWithDtls(false);
cricket::MediaSessionOptions options;
scoped_ptr<JsepSessionDescription> offer(
@ -2553,7 +2545,7 @@ TEST_F(WebRtcSessionTest, TestCreateAnswerBeforeIdentityRequestReturnSuccess) {
// identity generation is finished.
TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnSuccess) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
InitWithAsyncDtls(false);
InitWithDtls(false);
EXPECT_TRUE_WAIT(!session_->waiting_for_identity(), 1000);
talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
@ -2564,7 +2556,7 @@ TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnSuccess) {
// identity generation fails.
TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnFailure) {
MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
InitWithAsyncDtls(true);
InitWithDtls(true);
EXPECT_TRUE_WAIT(!session_->waiting_for_identity(), 1000);
talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));

3
talk/media/webrtc/fakewebrtcvideoengine.h
View File

@ -714,6 +714,9 @@ class FakeWebRtcVideoEngine
WEBRTC_STUB(DeregisterDecoderObserver, (const int));
WEBRTC_STUB(SendKeyFrame, (const int));
WEBRTC_STUB(WaitForFirstKeyFrame, (const int, const bool));
#ifdef USE_WEBRTC_DEV_BRANCH
WEBRTC_STUB(SetDecodeErrorMode, (const int, const ViEDecodeErrorMode));
#endif
WEBRTC_STUB(StartDebugRecording, (int, const char*));
WEBRTC_STUB(StopDebugRecording, (int));

2
talk/media/webrtc/webrtcvoiceengine_unittest.cc
View File

@ -125,7 +125,7 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
EXPECT_TRUE(SetupEngine());
// Remove stream added in Setup, which is corresponding to default channel.
int default_channel_num = voe_.GetLastChannel();
uint32 default_send_ssrc;
uint32 default_send_ssrc = 0u;
EXPECT_EQ(0, voe_.GetLocalSSRC(default_channel_num, default_send_ssrc));
EXPECT_EQ(kSsrc1, default_send_ssrc);
EXPECT_TRUE(channel_->RemoveSendStream(default_send_ssrc));