Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
Reason for revert:
Causing a compiler warning, and causing WebRtcBrowserTest.CallAndModifyStream to fail.
Original issue's description:
> Adding the ability to create an RtpSender without a track.
>
> This CL also changes AddStream to immediately create a sender, rather
> than waiting until the track is seen in SDP. And the PeerConnection now
> builds the list of "send streams" from the list of senders, rather than
> the collection of local media streams.
>
> Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> Cr-Commit-Position: refs/heads/master@{#10414}
TBR=pthatcher@webrtc.org,pthatcher@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1426443007
Cr-Commit-Position: refs/heads/master@{#10417}
diff --git a/talk/app/webrtc/audiotrack.cc b/talk/app/webrtc/audiotrack.cc
index 28e9e27..b0c9129 100644
--- a/talk/app/webrtc/audiotrack.cc
+++ b/talk/app/webrtc/audiotrack.cc
@@ -31,7 +31,7 @@
namespace webrtc {
-const char MediaStreamTrackInterface::kAudioTrackKind[] = "audio";
+static const char kAudioTrackKind[] = "audio";
AudioTrack::AudioTrack(const std::string& label,
AudioSourceInterface* audio_source)
diff --git a/talk/app/webrtc/mediastreaminterface.h b/talk/app/webrtc/mediastreaminterface.h
index cac1735..5911e85 100644
--- a/talk/app/webrtc/mediastreaminterface.h
+++ b/talk/app/webrtc/mediastreaminterface.h
@@ -100,9 +100,6 @@
kFailed = 3, // Track negotiation failed.
};
- static const char kAudioTrackKind[];
- static const char kVideoTrackKind[];
-
virtual std::string kind() const = 0;
virtual std::string id() const = 0;
virtual bool enabled() const = 0;
diff --git a/talk/app/webrtc/mediastreamprovider.h b/talk/app/webrtc/mediastreamprovider.h
index a78b55a..1c62daf 100644
--- a/talk/app/webrtc/mediastreamprovider.h
+++ b/talk/app/webrtc/mediastreamprovider.h
@@ -50,8 +50,8 @@
// RtpSenders/Receivers to get to the BaseChannels. These interfaces should be
// refactored away eventually, as the classes converge.
-// This interface is called by AudioRtpSender/Receivers to change the settings
-// of an audio track connected to certain PeerConnection.
+// This interface is called by AudioTrackHandler classes in mediastreamhandler.h
+// to change the settings of an audio track connected to certain PeerConnection.
class AudioProviderInterface {
public:
// Enable/disable the audio playout of a remote audio track with |ssrc|.
@@ -71,8 +71,9 @@
virtual ~AudioProviderInterface() {}
};
-// This interface is called by VideoRtpSender/Receivers to change the settings
-// of a video track connected to a certain PeerConnection.
+// This interface is called by VideoTrackHandler classes in mediastreamhandler.h
+// to change the settings of a video track connected to a certain
+// PeerConnection.
class VideoProviderInterface {
public:
virtual bool SetCaptureDevice(uint32_t ssrc,
diff --git a/talk/app/webrtc/peerconnection.cc b/talk/app/webrtc/peerconnection.cc
index 138da5b..fecc19b 100644
--- a/talk/app/webrtc/peerconnection.cc
+++ b/talk/app/webrtc/peerconnection.cc
@@ -59,7 +59,6 @@
using webrtc::MediaConstraintsInterface;
using webrtc::MediaStreamInterface;
using webrtc::PeerConnectionInterface;
-using webrtc::RtpSenderInterface;
using webrtc::StreamCollection;
using webrtc::StunConfigurations;
using webrtc::TurnConfigurations;
@@ -366,15 +365,25 @@
}
// Add the stream and RTP data channel info to |session_options|.
-void AddSendStreams(
- cricket::MediaSessionOptions* session_options,
- const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& senders,
- const std::map<std::string, rtc::scoped_refptr<DataChannel>>&
- rtp_data_channels) {
+void SetStreams(cricket::MediaSessionOptions* session_options,
+ rtc::scoped_refptr<StreamCollection> streams,
+ const std::map<std::string, rtc::scoped_refptr<DataChannel>>&
+ rtp_data_channels) {
session_options->streams.clear();
- for (const auto& sender : senders) {
- session_options->AddSendStream(sender->media_type(), sender->id(),
- sender->stream_id());
+ if (streams != nullptr) {
+ for (size_t i = 0; i < streams->count(); ++i) {
+ MediaStreamInterface* stream = streams->at(i);
+ // For each audio track in the stream, add it to the MediaSessionOptions.
+ for (const auto& track : stream->GetAudioTracks()) {
+ session_options->AddSendStream(cricket::MEDIA_TYPE_AUDIO, track->id(),
+ stream->label());
+ }
+ // For each video track in the stream, add it to the MediaSessionOptions.
+ for (const auto& track : stream->GetVideoTracks()) {
+ session_options->AddSendStream(cricket::MEDIA_TYPE_VIDEO, track->id(),
+ stream->label());
+ }
+ }
}
// Check for data channels.
@@ -659,6 +668,8 @@
return remote_streams_;
}
+// TODO(deadbeef): Create RtpSenders immediately here, even if local
+// description hasn't yet been set.
bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
if (IsClosed()) {
return false;
@@ -669,46 +680,23 @@
local_streams_->AddStream(local_stream);
+ // Find tracks that have already been configured in SDP. This can occur if a
+ // local session description that contains the MSID of these tracks is set
+ // before AddLocalStream is called. It can also occur if the local session
+ // description is not changed and RemoveLocalStream is called and later
+ // AddLocalStream is called again with the same stream.
for (const auto& track : local_stream->GetAudioTracks()) {
- auto sender = FindSenderForTrack(track.get());
- if (sender == senders_.end()) {
- // Normal case; we've never seen this track before.
- AudioRtpSender* new_sender = new AudioRtpSender(
- track.get(), local_stream->label(), session_.get(), stats_.get());
- senders_.push_back(new_sender);
- // If the sender has already been configured in SDP, we call SetSsrc,
- // which will connect the sender to the underlying transport. This can
- // occur if a local session description that contains the ID of the sender
- // is set before AddStream is called. It can also occur if the local
- // session description is not changed and RemoveStream is called, and
- // later AddStream is called again with the same stream.
- const TrackInfo* track_info = FindTrackInfo(
- local_audio_tracks_, local_stream->label(), track->id());
- if (track_info) {
- new_sender->SetSsrc(track_info->ssrc);
- }
- } else {
- // We already have a sender for this track, so just change the stream_id
- // so that it's correct in the next call to CreateOffer.
- (*sender)->set_stream_id(local_stream->label());
+ const TrackInfo* track_info =
+ FindTrackInfo(local_audio_tracks_, local_stream->label(), track->id());
+ if (track_info) {
+ CreateAudioSender(local_stream, track.get(), track_info->ssrc);
}
}
for (const auto& track : local_stream->GetVideoTracks()) {
- auto sender = FindSenderForTrack(track.get());
- if (sender == senders_.end()) {
- // Normal case; we've never seen this track before.
- VideoRtpSender* new_sender = new VideoRtpSender(
- track.get(), local_stream->label(), session_.get());
- senders_.push_back(new_sender);
- const TrackInfo* track_info = FindTrackInfo(
- local_video_tracks_, local_stream->label(), track->id());
- if (track_info) {
- new_sender->SetSsrc(track_info->ssrc);
- }
- } else {
- // We already have a sender for this track, so just change the stream_id
- // so that it's correct in the next call to CreateOffer.
- (*sender)->set_stream_id(local_stream->label());
+ const TrackInfo* track_info =
+ FindTrackInfo(local_video_tracks_, local_stream->label(), track->id());
+ if (track_info) {
+ CreateVideoSender(local_stream, track.get(), track_info->ssrc);
}
}
@@ -718,27 +706,21 @@
}
// TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
-// indefinitely, when we have unified plan SDP.
+// indefinitely.
void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
for (const auto& track : local_stream->GetAudioTracks()) {
- auto sender = FindSenderForTrack(track.get());
- if (sender == senders_.end()) {
- LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
- << " doesn't exist.";
- continue;
+ const TrackInfo* track_info =
+ FindTrackInfo(local_audio_tracks_, local_stream->label(), track->id());
+ if (track_info) {
+ DestroyAudioSender(local_stream, track.get(), track_info->ssrc);
}
- (*sender)->Stop();
- senders_.erase(sender);
}
for (const auto& track : local_stream->GetVideoTracks()) {
- auto sender = FindSenderForTrack(track.get());
- if (sender == senders_.end()) {
- LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
- << " doesn't exist.";
- continue;
+ const TrackInfo* track_info =
+ FindTrackInfo(local_video_tracks_, local_stream->label(), track->id());
+ if (track_info) {
+ DestroyVideoSender(local_stream, track.get());
}
- (*sender)->Stop();
- senders_.erase(sender);
}
local_streams_->RemoveStream(local_stream);
@@ -769,21 +751,6 @@
return DtmfSenderProxy::Create(signaling_thread(), sender.get());
}
-rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
- const std::string& kind) {
- RtpSenderInterface* new_sender;
- if (kind == MediaStreamTrackInterface::kAudioTrackKind) {
- new_sender = new AudioRtpSender(session_.get(), stats_.get());
- } else if (kind == MediaStreamTrackInterface::kVideoTrackKind) {
- new_sender = new VideoRtpSender(session_.get());
- } else {
- LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
- return rtc::scoped_refptr<RtpSenderInterface>();
- }
- senders_.push_back(new_sender);
- return RtpSenderProxy::Create(signaling_thread(), new_sender);
-}
-
std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
const {
std::vector<rtc::scoped_refptr<RtpSenderInterface>> senders;
@@ -1300,6 +1267,49 @@
}
}
+void PeerConnection::CreateAudioSender(MediaStreamInterface* stream,
+ AudioTrackInterface* audio_track,
+ uint32_t ssrc) {
+ senders_.push_back(new AudioRtpSender(audio_track, ssrc, session_.get()));
+ stats_->AddLocalAudioTrack(audio_track, ssrc);
+}
+
+void PeerConnection::CreateVideoSender(MediaStreamInterface* stream,
+ VideoTrackInterface* video_track,
+ uint32_t ssrc) {
+ senders_.push_back(new VideoRtpSender(video_track, ssrc, session_.get()));
+}
+
+// TODO(deadbeef): Keep RtpSenders around even if track goes away in local
+// description.
+void PeerConnection::DestroyAudioSender(MediaStreamInterface* stream,
+ AudioTrackInterface* audio_track,
+ uint32_t ssrc) {
+ auto it = FindSenderForTrack(audio_track);
+ if (it == senders_.end()) {
+ LOG(LS_WARNING) << "RtpSender for track with id " << audio_track->id()
+ << " doesn't exist.";
+ return;
+ } else {
+ (*it)->Stop();
+ senders_.erase(it);
+ }
+ stats_->RemoveLocalAudioTrack(audio_track, ssrc);
+}
+
+void PeerConnection::DestroyVideoSender(MediaStreamInterface* stream,
+ VideoTrackInterface* video_track) {
+ auto it = FindSenderForTrack(video_track);
+ if (it == senders_.end()) {
+ LOG(LS_WARNING) << "RtpSender for track with id " << video_track->id()
+ << " doesn't exist.";
+ return;
+ } else {
+ (*it)->Stop();
+ senders_.erase(it);
+ }
+}
+
void PeerConnection::OnIceConnectionChange(
PeerConnectionInterface::IceConnectionState new_state) {
RTC_DCHECK(signaling_thread()->IsCurrent());
@@ -1375,7 +1385,7 @@
return false;
}
- AddSendStreams(session_options, senders_, rtp_data_channels_);
+ SetStreams(session_options, local_streams_, rtp_data_channels_);
// Offer to receive audio/video if the constraint is not set and there are
// send streams, or we're currently receiving.
if (rtc_options.offer_to_receive_audio == RTCOfferAnswerOptions::kUndefined) {
@@ -1408,7 +1418,7 @@
return false;
}
- AddSendStreams(session_options, senders_, rtp_data_channels_);
+ SetStreams(session_options, local_streams_, rtp_data_channels_);
session_options->bundle_enabled =
session_options->bundle_enabled &&
(session_options->has_audio() || session_options->has_video() ||
@@ -1430,7 +1440,8 @@
TrackInfos* current_tracks = GetRemoteTracks(media_type);
// Find removed tracks. I.e., tracks where the track id or ssrc don't match
- // the new StreamParam.
+ // the
+ // new StreamParam.
auto track_it = current_tracks->begin();
while (track_it != current_tracks->end()) {
const TrackInfo& info = *track_it;
@@ -1632,44 +1643,62 @@
const std::string& track_id,
uint32_t ssrc,
cricket::MediaType media_type) {
- RtpSenderInterface* sender = FindSenderById(track_id);
- if (!sender) {
- LOG(LS_WARNING) << "An unknown RtpSender with id " << track_id
- << " has been configured in the local description.";
+ MediaStreamInterface* stream = local_streams_->find(stream_label);
+ if (!stream) {
+ LOG(LS_WARNING) << "An unknown local MediaStream with label "
+ << stream_label << " has been configured.";
return;
}
- if (sender->media_type() != media_type) {
- LOG(LS_WARNING) << "An RtpSender has been configured in the local"
- << " description with an unexpected media type.";
- return;
+ if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ AudioTrackInterface* audio_track = stream->FindAudioTrack(track_id);
+ if (!audio_track) {
+ LOG(LS_WARNING) << "An unknown local AudioTrack with id , " << track_id
+ << " has been configured.";
+ return;
+ }
+ CreateAudioSender(stream, audio_track, ssrc);
+ } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+ VideoTrackInterface* video_track = stream->FindVideoTrack(track_id);
+ if (!video_track) {
+ LOG(LS_WARNING) << "An unknown local VideoTrack with id , " << track_id
+ << " has been configured.";
+ return;
+ }
+ CreateVideoSender(stream, video_track, ssrc);
+ } else {
+ RTC_DCHECK(false && "Invalid media type");
}
-
- sender->set_stream_id(stream_label);
- sender->SetSsrc(ssrc);
}
void PeerConnection::OnLocalTrackRemoved(const std::string& stream_label,
const std::string& track_id,
uint32_t ssrc,
cricket::MediaType media_type) {
- RtpSenderInterface* sender = FindSenderById(track_id);
- if (!sender) {
- // This is the normal case. I.e., RemoveStream has been called and the
+ MediaStreamInterface* stream = local_streams_->find(stream_label);
+ if (!stream) {
+ // This is the normal case. I.e., RemoveLocalStream has been called and the
// SessionDescriptions has been renegotiated.
return;
}
-
- // A sender has been removed from the SessionDescription but it's still
- // associated with the PeerConnection. This only occurs if the SDP doesn't
- // match with the calls to CreateSender, AddStream and RemoveStream.
- if (sender->media_type() != media_type) {
- LOG(LS_WARNING) << "An RtpSender has been configured in the local"
- << " description with an unexpected media type.";
- return;
+ // A track has been removed from the SessionDescription but the MediaStream
+ // is still associated with PeerConnection. This only occurs if the SDP
+ // doesn't match with the calls to AddLocalStream and RemoveLocalStream.
+ if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+ AudioTrackInterface* audio_track = stream->FindAudioTrack(track_id);
+ if (!audio_track) {
+ return;
+ }
+ DestroyAudioSender(stream, audio_track, ssrc);
+ } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+ VideoTrackInterface* video_track = stream->FindVideoTrack(track_id);
+ if (!video_track) {
+ return;
+ }
+ DestroyVideoSender(stream, video_track);
+ } else {
+ RTC_DCHECK(false && "Invalid media type.");
}
-
- sender->SetSsrc(0);
}
void PeerConnection::UpdateLocalRtpDataChannels(
@@ -1887,15 +1916,6 @@
DataChannelProxy::Create(signaling_thread(), channel));
}
-RtpSenderInterface* PeerConnection::FindSenderById(const std::string& id) {
- auto it =
- std::find_if(senders_.begin(), senders_.end(),
- [id](const rtc::scoped_refptr<RtpSenderInterface>& sender) {
- return sender->id() == id;
- });
- return it != senders_.end() ? it->get() : nullptr;
-}
-
std::vector<rtc::scoped_refptr<RtpSenderInterface>>::iterator
PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) {
return std::find_if(
diff --git a/talk/app/webrtc/peerconnection.h b/talk/app/webrtc/peerconnection.h
index c233388..2d388ae 100644
--- a/talk/app/webrtc/peerconnection.h
+++ b/talk/app/webrtc/peerconnection.h
@@ -91,9 +91,6 @@
rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
AudioTrackInterface* track) override;
- rtc::scoped_refptr<RtpSenderInterface> CreateSender(
- const std::string& kind) override;
-
std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
const override;
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
@@ -190,6 +187,12 @@
AudioTrackInterface* audio_track);
void DestroyVideoReceiver(MediaStreamInterface* stream,
VideoTrackInterface* video_track);
+ void CreateAudioSender(MediaStreamInterface* stream,
+ AudioTrackInterface* audio_track,
+ uint32_t ssrc);
+ void CreateVideoSender(MediaStreamInterface* stream,
+ VideoTrackInterface* video_track,
+ uint32_t ssrc);
void DestroyAudioSender(MediaStreamInterface* stream,
AudioTrackInterface* audio_track,
uint32_t ssrc);
@@ -325,8 +328,6 @@
void OnDataChannelOpenMessage(const std::string& label,
const InternalDataChannelInit& config);
- RtpSenderInterface* FindSenderById(const std::string& id);
-
std::vector<rtc::scoped_refptr<RtpSenderInterface>>::iterator
FindSenderForTrack(MediaStreamTrackInterface* track);
std::vector<rtc::scoped_refptr<RtpReceiverInterface>>::iterator
diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc
index abe5e1c..3cf66d6 100644
--- a/talk/app/webrtc/peerconnection_unittest.cc
+++ b/talk/app/webrtc/peerconnection_unittest.cc
@@ -96,7 +96,6 @@
#if !defined(THREAD_SANITIZER)
static const int kMaxWaitForStatsMs = 3000;
#endif
-static const int kMaxWaitForActivationMs = 5000;
static const int kMaxWaitForFramesMs = 10000;
static const int kEndAudioFrameCount = 3;
static const int kEndVideoFrameCount = 3;
@@ -251,7 +250,18 @@
peer_connection_factory_->CreateLocalMediaStream(stream_label);
if (audio && can_receive_audio()) {
- stream->AddTrack(CreateLocalAudioTrack(stream_label));
+ FakeConstraints constraints;
+ // Disable highpass filter so that we can get all the test audio frames.
+ constraints.AddMandatory(
+ MediaConstraintsInterface::kHighpassFilter, false);
+ rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
+ peer_connection_factory_->CreateAudioSource(&constraints);
+ // TODO(perkj): Test audio source when it is implemented. Currently audio
+ // always use the default input.
+ std::string label = stream_label + kAudioTrackLabelBase;
+ rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
+ peer_connection_factory_->CreateAudioTrack(label, source));
+ stream->AddTrack(audio_track);
}
if (video && can_receive_video()) {
stream->AddTrack(CreateLocalVideoTrack(stream_label));
@@ -347,35 +357,6 @@
data_observer_.reset(new MockDataChannelObserver(data_channel_));
}
- rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack(
- const std::string& stream_label) {
- FakeConstraints constraints;
- // Disable highpass filter so that we can get all the test audio frames.
- constraints.AddMandatory(MediaConstraintsInterface::kHighpassFilter, false);
- rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
- peer_connection_factory_->CreateAudioSource(&constraints);
- // TODO(perkj): Test audio source when it is implemented. Currently audio
- // always use the default input.
- std::string label = stream_label + kAudioTrackLabelBase;
- return peer_connection_factory_->CreateAudioTrack(label, source);
- }
-
- rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack(
- const std::string& stream_label) {
- // Set max frame rate to 10fps to reduce the risk of the tests to be flaky.
- FakeConstraints source_constraints = video_constraints_;
- source_constraints.SetMandatoryMaxFrameRate(10);
-
- cricket::FakeVideoCapturer* fake_capturer =
- new webrtc::FakePeriodicVideoCapturer();
- video_capturers_.push_back(fake_capturer);
- rtc::scoped_refptr<webrtc::VideoSourceInterface> source =
- peer_connection_factory_->CreateVideoSource(fake_capturer,
- &source_constraints);
- std::string label = stream_label + kVideoTrackLabelBase;
- return peer_connection_factory_->CreateVideoTrack(label, source);
- }
-
DataChannelInterface* data_channel() { return data_channel_; }
const MockDataChannelObserver* data_observer() const {
return data_observer_.get();
@@ -690,6 +671,22 @@
return peer_connection_.get() != nullptr;
}
+ rtc::scoped_refptr<webrtc::VideoTrackInterface>
+ CreateLocalVideoTrack(const std::string stream_label) {
+ // Set max frame rate to 10fps to reduce the risk of the tests to be flaky.
+ FakeConstraints source_constraints = video_constraints_;
+ source_constraints.SetMandatoryMaxFrameRate(10);
+
+ cricket::FakeVideoCapturer* fake_capturer =
+ new webrtc::FakePeriodicVideoCapturer();
+ video_capturers_.push_back(fake_capturer);
+ rtc::scoped_refptr<webrtc::VideoSourceInterface> source =
+ peer_connection_factory_->CreateVideoSource(
+ fake_capturer, &source_constraints);
+ std::string label = stream_label + kVideoTrackLabelBase;
+ return peer_connection_factory_->CreateVideoTrack(label, source);
+ }
+
rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(
webrtc::PortAllocatorFactoryInterface* factory,
const MediaConstraintsInterface* constraints) {
@@ -960,11 +957,13 @@
initiating_client_->AddMediaStream(true, true);
}
initiating_client_->Negotiate();
+ const int kMaxWaitForActivationMs = 5000;
// Assert true is used here since next tests are guaranteed to fail and
// would eat up 5 seconds.
ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
VerifySessionDescriptions();
+
int audio_frame_count = kEndAudioFrameCount;
// TODO(ronghuawu): Add test to cover the case of sendonly and recvonly.
if (!initiating_client_->can_receive_audio() ||
@@ -1574,34 +1573,6 @@
LocalP2PTest();
}
-// This tests that if we negotiate after calling CreateSender but before we
-// have a track, then set a track later, frames from the newly-set track are
-// received end-to-end.
-TEST_F(JsepPeerConnectionP2PTestClient, EarlyWarmupTest) {
- ASSERT_TRUE(CreateTestClients());
- auto audio_sender = initializing_client()->pc()->CreateSender("audio");
- auto video_sender = initializing_client()->pc()->CreateSender("video");
- initializing_client()->Negotiate();
- // Wait for ICE connection to complete, without any tracks.
- // Note that the receiving client WILL (in HandleIncomingOffer) create
- // tracks, so it's only the initiator here that's doing early warmup.
- ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
- VerifySessionDescriptions();
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
- initializing_client()->ice_connection_state(),
- kMaxWaitForFramesMs);
- EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
- receiving_client()->ice_connection_state(),
- kMaxWaitForFramesMs);
- // Now set the tracks, and expect frames to immediately start flowing.
- EXPECT_TRUE(
- audio_sender->SetTrack(initializing_client()->CreateLocalAudioTrack("")));
- EXPECT_TRUE(
- video_sender->SetTrack(initializing_client()->CreateLocalVideoTrack("")));
- EXPECT_TRUE_WAIT(FramesNotPending(kEndAudioFrameCount, kEndVideoFrameCount),
- kMaxWaitForFramesMs);
-}
-
class IceServerParsingTest : public testing::Test {
public:
// Convenience for parsing a single URL.
diff --git a/talk/app/webrtc/peerconnectioninterface.h b/talk/app/webrtc/peerconnectioninterface.h
index 317e663..77caa9d 100644
--- a/talk/app/webrtc/peerconnectioninterface.h
+++ b/talk/app/webrtc/peerconnectioninterface.h
@@ -337,12 +337,6 @@
AudioTrackInterface* track) = 0;
// TODO(deadbeef): Make these pure virtual once all subclasses implement them.
- // |kind| must be "audio" or "video".
- virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
- const std::string& kind) {
- return rtc::scoped_refptr<RtpSenderInterface>();
- }
-
virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
const {
return std::vector<rtc::scoped_refptr<RtpSenderInterface>>();
diff --git a/talk/app/webrtc/peerconnectioninterface_unittest.cc b/talk/app/webrtc/peerconnectioninterface_unittest.cc
index df5c38b..63163fd 100644
--- a/talk/app/webrtc/peerconnectioninterface_unittest.cc
+++ b/talk/app/webrtc/peerconnectioninterface_unittest.cc
@@ -2020,10 +2020,8 @@
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1]));
// Remove an audio and video track.
- pc_->RemoveStream(reference_collection_->at(0));
rtc::scoped_ptr<SessionDescriptionInterface> desc_2;
CreateSessionDescriptionAndReference(1, 1, desc_2.accept());
- pc_->AddStream(reference_collection_->at(0));
EXPECT_TRUE(DoSetLocalDescription(desc_2.release()));
senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());
diff --git a/talk/app/webrtc/peerconnectionproxy.h b/talk/app/webrtc/peerconnectionproxy.h
index 9b446c0..d207fbb 100644
--- a/talk/app/webrtc/peerconnectionproxy.h
+++ b/talk/app/webrtc/peerconnectionproxy.h
@@ -43,9 +43,6 @@
PROXY_METHOD1(void, RemoveStream, MediaStreamInterface*)
PROXY_METHOD1(rtc::scoped_refptr<DtmfSenderInterface>,
CreateDtmfSender, AudioTrackInterface*)
- PROXY_METHOD1(rtc::scoped_refptr<RtpSenderInterface>,
- CreateSender,
- const std::string&)
PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpSenderInterface>>,
GetSenders)
PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpReceiverInterface>>,
diff --git a/talk/app/webrtc/rtpsender.cc b/talk/app/webrtc/rtpsender.cc
index 5b3ad49..3a78f45 100644
--- a/talk/app/webrtc/rtpsender.cc
+++ b/talk/app/webrtc/rtpsender.cc
@@ -29,7 +29,6 @@
#include "talk/app/webrtc/localaudiosource.h"
#include "talk/app/webrtc/videosourceinterface.h"
-#include "webrtc/base/helpers.h"
namespace webrtc {
@@ -60,49 +59,34 @@
}
AudioRtpSender::AudioRtpSender(AudioTrackInterface* track,
- const std::string& stream_id,
- AudioProviderInterface* provider,
- StatsCollector* stats)
+ uint32_t ssrc,
+ AudioProviderInterface* provider)
: id_(track->id()),
- stream_id_(stream_id),
- provider_(provider),
- stats_(stats),
track_(track),
+ ssrc_(ssrc),
+ provider_(provider),
cached_track_enabled_(track->enabled()),
sink_adapter_(new LocalAudioSinkAdapter()) {
- RTC_DCHECK(provider != nullptr);
track_->RegisterObserver(this);
track_->AddSink(sink_adapter_.get());
+ Reconfigure();
}
-AudioRtpSender::AudioRtpSender(AudioProviderInterface* provider,
- StatsCollector* stats)
- : id_(rtc::CreateRandomUuid()),
- stream_id_(rtc::CreateRandomUuid()),
- provider_(provider),
- stats_(stats),
- sink_adapter_(new LocalAudioSinkAdapter()) {}
-
AudioRtpSender::~AudioRtpSender() {
+ track_->RemoveSink(sink_adapter_.get());
+ track_->UnregisterObserver(this);
Stop();
}
void AudioRtpSender::OnChanged() {
- RTC_DCHECK(!stopped_);
if (cached_track_enabled_ != track_->enabled()) {
cached_track_enabled_ = track_->enabled();
- if (can_send_track()) {
- SetAudioSend();
- }
+ Reconfigure();
}
}
bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
- if (stopped_) {
- LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
- return false;
- }
- if (track && track->kind() != MediaStreamTrackInterface::kAudioTrackKind) {
+ if (track->kind() != "audio") {
LOG(LS_ERROR) << "SetTrack called on audio RtpSender with " << track->kind()
<< " track.";
return false;
@@ -110,83 +94,36 @@
AudioTrackInterface* audio_track = static_cast<AudioTrackInterface*>(track);
// Detach from old track.
- if (track_) {
- track_->RemoveSink(sink_adapter_.get());
- track_->UnregisterObserver(this);
- }
-
- if (can_send_track() && stats_) {
- stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
- }
+ track_->RemoveSink(sink_adapter_.get());
+ track_->UnregisterObserver(this);
// Attach to new track.
- bool prev_can_send_track = can_send_track();
track_ = audio_track;
- if (track_) {
- cached_track_enabled_ = track_->enabled();
- track_->RegisterObserver(this);
- track_->AddSink(sink_adapter_.get());
- }
-
- // Update audio provider.
- if (can_send_track()) {
- SetAudioSend();
- if (stats_) {
- stats_->AddLocalAudioTrack(track_.get(), ssrc_);
- }
- } else if (prev_can_send_track) {
- cricket::AudioOptions options;
- provider_->SetAudioSend(ssrc_, false, options, nullptr);
- }
+ cached_track_enabled_ = track_->enabled();
+ track_->RegisterObserver(this);
+ track_->AddSink(sink_adapter_.get());
+ Reconfigure();
return true;
}
-void AudioRtpSender::SetSsrc(uint32_t ssrc) {
- if (stopped_ || ssrc == ssrc_) {
- return;
- }
- // If we are already sending with a particular SSRC, stop sending.
- if (can_send_track()) {
- cricket::AudioOptions options;
- provider_->SetAudioSend(ssrc_, false, options, nullptr);
- if (stats_) {
- stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
- }
- }
- ssrc_ = ssrc;
- if (can_send_track()) {
- SetAudioSend();
- if (stats_) {
- stats_->AddLocalAudioTrack(track_.get(), ssrc_);
- }
- }
-}
-
void AudioRtpSender::Stop() {
// TODO(deadbeef): Need to do more here to fully stop sending packets.
- if (stopped_) {
+ if (!provider_) {
return;
}
- if (track_) {
- track_->RemoveSink(sink_adapter_.get());
- track_->UnregisterObserver(this);
- }
- if (can_send_track()) {
- cricket::AudioOptions options;
- provider_->SetAudioSend(ssrc_, false, options, nullptr);
- if (stats_) {
- stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
- }
- }
- stopped_ = true;
+ cricket::AudioOptions options;
+ provider_->SetAudioSend(ssrc_, false, options, nullptr);
+ provider_ = nullptr;
}
-void AudioRtpSender::SetAudioSend() {
- RTC_DCHECK(!stopped_ && can_send_track());
+void AudioRtpSender::Reconfigure() {
+ if (!provider_) {
+ return;
+ }
cricket::AudioOptions options;
if (track_->enabled() && track_->GetSource()) {
// TODO(xians): Remove this static_cast since we should be able to connect
- // a remote audio track to a peer connection.
+ // a remote audio track to peer connection.
options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
}
@@ -199,42 +136,35 @@
}
VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
- const std::string& stream_id,
+ uint32_t ssrc,
VideoProviderInterface* provider)
: id_(track->id()),
- stream_id_(stream_id),
- provider_(provider),
track_(track),
+ ssrc_(ssrc),
+ provider_(provider),
cached_track_enabled_(track->enabled()) {
- RTC_DCHECK(provider != nullptr);
track_->RegisterObserver(this);
+ VideoSourceInterface* source = track_->GetSource();
+ if (source) {
+ provider_->SetCaptureDevice(ssrc_, source->GetVideoCapturer());
+ }
+ Reconfigure();
}
-VideoRtpSender::VideoRtpSender(VideoProviderInterface* provider)
- : id_(rtc::CreateRandomUuid()),
- stream_id_(rtc::CreateRandomUuid()),
- provider_(provider) {}
-
VideoRtpSender::~VideoRtpSender() {
+ track_->UnregisterObserver(this);
Stop();
}
void VideoRtpSender::OnChanged() {
- RTC_DCHECK(!stopped_);
if (cached_track_enabled_ != track_->enabled()) {
cached_track_enabled_ = track_->enabled();
- if (can_send_track()) {
- SetVideoSend();
- }
+ Reconfigure();
}
}
bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
- if (stopped_) {
- LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
- return false;
- }
- if (track && track->kind() != MediaStreamTrackInterface::kVideoTrackKind) {
+ if (track->kind() != "video") {
LOG(LS_ERROR) << "SetTrack called on video RtpSender with " << track->kind()
<< " track.";
return false;
@@ -242,72 +172,30 @@
VideoTrackInterface* video_track = static_cast<VideoTrackInterface*>(track);
// Detach from old track.
- if (track_) {
- track_->UnregisterObserver(this);
- }
+ track_->UnregisterObserver(this);
// Attach to new track.
- bool prev_can_send_track = can_send_track();
track_ = video_track;
- if (track_) {
- cached_track_enabled_ = track_->enabled();
- track_->RegisterObserver(this);
- }
-
- // Update video provider.
- if (can_send_track()) {
- VideoSourceInterface* source = track_->GetSource();
- // TODO(deadbeef): If SetTrack is called with a disabled track, and the
- // previous track was enabled, this could cause a frame from the new track
- // to slip out. Really, what we need is for SetCaptureDevice and
- // SetVideoSend
- // to be combined into one atomic operation, all the way down to
- // WebRtcVideoSendStream.
- provider_->SetCaptureDevice(ssrc_,
- source ? source->GetVideoCapturer() : nullptr);
- SetVideoSend();
- } else if (prev_can_send_track) {
- provider_->SetCaptureDevice(ssrc_, nullptr);
- provider_->SetVideoSend(ssrc_, false, nullptr);
- }
+ cached_track_enabled_ = track_->enabled();
+ track_->RegisterObserver(this);
+ Reconfigure();
return true;
}
-void VideoRtpSender::SetSsrc(uint32_t ssrc) {
- if (stopped_ || ssrc == ssrc_) {
- return;
- }
- // If we are already sending with a particular SSRC, stop sending.
- if (can_send_track()) {
- provider_->SetCaptureDevice(ssrc_, nullptr);
- provider_->SetVideoSend(ssrc_, false, nullptr);
- }
- ssrc_ = ssrc;
- if (can_send_track()) {
- VideoSourceInterface* source = track_->GetSource();
- provider_->SetCaptureDevice(ssrc_,
- source ? source->GetVideoCapturer() : nullptr);
- SetVideoSend();
- }
-}
-
void VideoRtpSender::Stop() {
// TODO(deadbeef): Need to do more here to fully stop sending packets.
- if (stopped_) {
+ if (!provider_) {
return;
}
- if (track_) {
- track_->UnregisterObserver(this);
- }
- if (can_send_track()) {
- provider_->SetCaptureDevice(ssrc_, nullptr);
- provider_->SetVideoSend(ssrc_, false, nullptr);
- }
- stopped_ = true;
+ provider_->SetCaptureDevice(ssrc_, nullptr);
+ provider_->SetVideoSend(ssrc_, false, nullptr);
+ provider_ = nullptr;
}
-void VideoRtpSender::SetVideoSend() {
- RTC_DCHECK(!stopped_ && can_send_track());
+void VideoRtpSender::Reconfigure() {
+ if (!provider_) {
+ return;
+ }
const cricket::VideoOptions* options = nullptr;
VideoSourceInterface* source = track_->GetSource();
if (track_->enabled() && source) {
diff --git a/talk/app/webrtc/rtpsender.h b/talk/app/webrtc/rtpsender.h
index d5f88a9..3741909 100644
--- a/talk/app/webrtc/rtpsender.h
+++ b/talk/app/webrtc/rtpsender.h
@@ -36,7 +36,6 @@
#include "talk/app/webrtc/mediastreamprovider.h"
#include "talk/app/webrtc/rtpsenderinterface.h"
-#include "talk/app/webrtc/statscollector.h"
#include "talk/media/base/audiorenderer.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/criticalsection.h"
@@ -71,15 +70,9 @@
class AudioRtpSender : public ObserverInterface,
public rtc::RefCountedObject<RtpSenderInterface> {
public:
- // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called
- // at the appropriate times.
AudioRtpSender(AudioTrackInterface* track,
- const std::string& stream_id,
- AudioProviderInterface* provider,
- StatsCollector* stats);
-
- // Randomly generates id and stream_id.
- AudioRtpSender(AudioProviderInterface* provider, StatsCollector* stats);
+ uint32_t ssrc,
+ AudioProviderInterface* provider);
virtual ~AudioRtpSender();
@@ -92,37 +85,18 @@
return track_.get();
}
- void SetSsrc(uint32_t ssrc) override;
-
- uint32_t ssrc() const override { return ssrc_; }
-
- cricket::MediaType media_type() const override {
- return cricket::MEDIA_TYPE_AUDIO;
- }
-
std::string id() const override { return id_; }
- void set_stream_id(const std::string& stream_id) override {
- stream_id_ = stream_id;
- }
- std::string stream_id() const override { return stream_id_; }
-
void Stop() override;
private:
- bool can_send_track() const { return track_ && ssrc_; }
- // Helper function to construct options for
- // AudioProviderInterface::SetAudioSend.
- void SetAudioSend();
+ void Reconfigure();
std::string id_;
- std::string stream_id_;
- AudioProviderInterface* provider_;
- StatsCollector* stats_;
rtc::scoped_refptr<AudioTrackInterface> track_;
- uint32_t ssrc_ = 0;
- bool cached_track_enabled_ = false;
- bool stopped_ = false;
+ uint32_t ssrc_;
+ AudioProviderInterface* provider_;
+ bool cached_track_enabled_;
// Used to pass the data callback from the |track_| to the other end of
// cricket::AudioRenderer.
@@ -133,12 +107,9 @@
public rtc::RefCountedObject<RtpSenderInterface> {
public:
VideoRtpSender(VideoTrackInterface* track,
- const std::string& stream_id,
+ uint32_t ssrc,
VideoProviderInterface* provider);
- // Randomly generates id and stream_id.
- explicit VideoRtpSender(VideoProviderInterface* provider);
-
virtual ~VideoRtpSender();
// ObserverInterface implementation
@@ -150,36 +121,18 @@
return track_.get();
}
- void SetSsrc(uint32_t ssrc) override;
-
- uint32_t ssrc() const override { return ssrc_; }
-
- cricket::MediaType media_type() const override {
- return cricket::MEDIA_TYPE_VIDEO;
- }
-
std::string id() const override { return id_; }
- void set_stream_id(const std::string& stream_id) override {
- stream_id_ = stream_id;
- }
- std::string stream_id() const override { return stream_id_; }
-
void Stop() override;
private:
- bool can_send_track() const { return track_ && ssrc_; }
- // Helper function to construct options for
- // VideoProviderInterface::SetVideoSend.
- void SetVideoSend();
+ void Reconfigure();
std::string id_;
- std::string stream_id_;
- VideoProviderInterface* provider_;
rtc::scoped_refptr<VideoTrackInterface> track_;
- uint32_t ssrc_ = 0;
- bool cached_track_enabled_ = false;
- bool stopped_ = false;
+ uint32_t ssrc_;
+ VideoProviderInterface* provider_;
+ bool cached_track_enabled_;
};
} // namespace webrtc
diff --git a/talk/app/webrtc/rtpsenderinterface.h b/talk/app/webrtc/rtpsenderinterface.h
index f54e8ca..fca98f2 100644
--- a/talk/app/webrtc/rtpsenderinterface.h
+++ b/talk/app/webrtc/rtpsenderinterface.h
@@ -35,7 +35,6 @@
#include "talk/app/webrtc/proxy.h"
#include "talk/app/webrtc/mediastreaminterface.h"
-#include "talk/session/media/mediasession.h"
#include "webrtc/base/refcount.h"
#include "webrtc/base/scoped_ref_ptr.h"
@@ -48,24 +47,10 @@
virtual bool SetTrack(MediaStreamTrackInterface* track) = 0;
virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
- // Used to set the SSRC of the sender, once a local description has been set.
- // If |ssrc| is 0, this indiates that the sender should disconnect from the
- // underlying transport (this occurs if the sender isn't seen in a local
- // description).
- virtual void SetSsrc(uint32_t ssrc) = 0;
- virtual uint32_t ssrc() const = 0;
-
- // Audio or video sender?
- virtual cricket::MediaType media_type() const = 0;
-
// Not to be confused with "mid", this is a field we can temporarily use
// to uniquely identify a receiver until we implement Unified Plan SDP.
virtual std::string id() const = 0;
- // TODO(deadbeef): Support one sender having multiple stream ids.
- virtual void set_stream_id(const std::string& stream_id) = 0;
- virtual std::string stream_id() const = 0;
-
virtual void Stop() = 0;
protected:
@@ -76,12 +61,7 @@
BEGIN_PROXY_MAP(RtpSender)
PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
-PROXY_METHOD1(void, SetSsrc, uint32_t)
-PROXY_CONSTMETHOD0(uint32_t, ssrc)
-PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
PROXY_CONSTMETHOD0(std::string, id)
-PROXY_METHOD1(void, set_stream_id, const std::string&)
-PROXY_CONSTMETHOD0(std::string, stream_id)
PROXY_METHOD0(void, Stop)
END_PROXY()
diff --git a/talk/app/webrtc/rtpsenderreceiver_unittest.cc b/talk/app/webrtc/rtpsenderreceiver_unittest.cc
index c987186..c9d7e00 100644
--- a/talk/app/webrtc/rtpsenderreceiver_unittest.cc
+++ b/talk/app/webrtc/rtpsenderreceiver_unittest.cc
@@ -48,9 +48,7 @@
static const char kVideoTrackId[] = "video_1";
static const char kAudioTrackId[] = "audio_1";
static const uint32_t kVideoSsrc = 98;
-static const uint32_t kVideoSsrc2 = 100;
static const uint32_t kAudioSsrc = 99;
-static const uint32_t kAudioSsrc2 = 101;
namespace webrtc {
@@ -122,10 +120,8 @@
audio_track_ = AudioTrack::Create(kAudioTrackId, NULL);
EXPECT_TRUE(stream_->AddTrack(audio_track_));
EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
- audio_rtp_sender_ =
- new AudioRtpSender(stream_->GetAudioTracks()[0], stream_->label(),
- &audio_provider_, nullptr);
- audio_rtp_sender_->SetSsrc(kAudioSsrc);
+ audio_rtp_sender_ = new AudioRtpSender(stream_->GetAudioTracks()[0],
+ kAudioSsrc, &audio_provider_);
}
void CreateVideoRtpSender() {
@@ -134,8 +130,7 @@
kVideoSsrc, video_track_->GetSource()->GetVideoCapturer()));
EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
video_rtp_sender_ = new VideoRtpSender(stream_->GetVideoTracks()[0],
- stream_->label(), &video_provider_);
- video_rtp_sender_->SetSsrc(kVideoSsrc);
+ kVideoSsrc, &video_provider_);
}
void DestroyAudioRtpSender() {
@@ -285,206 +280,4 @@
DestroyAudioRtpReceiver();
}
-// Test that provider methods aren't called without both a track and an SSRC.
-TEST_F(RtpSenderReceiverTest, AudioSenderWithoutTrackAndSsrc) {
- rtc::scoped_refptr<AudioRtpSender> sender =
- new AudioRtpSender(&audio_provider_, nullptr);
- rtc::scoped_refptr<AudioTrackInterface> track =
- AudioTrack::Create(kAudioTrackId, nullptr);
- EXPECT_TRUE(sender->SetTrack(track));
- EXPECT_TRUE(sender->SetTrack(nullptr));
- sender->SetSsrc(kAudioSsrc);
- sender->SetSsrc(0);
- // Just let it get destroyed and make sure it doesn't call any methods on the
- // provider interface.
-}
-
-// Test that provider methods aren't called without both a track and an SSRC.
-TEST_F(RtpSenderReceiverTest, VideoSenderWithoutTrackAndSsrc) {
- rtc::scoped_refptr<VideoRtpSender> sender =
- new VideoRtpSender(&video_provider_);
- EXPECT_TRUE(sender->SetTrack(video_track_));
- EXPECT_TRUE(sender->SetTrack(nullptr));
- sender->SetSsrc(kVideoSsrc);
- sender->SetSsrc(0);
- // Just let it get destroyed and make sure it doesn't call any methods on the
- // provider interface.
-}
-
-// Test that an audio sender calls the expected methods on the provider once
-// it has a track and SSRC, when the SSRC is set first.
-TEST_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupSsrcThenTrack) {
- rtc::scoped_refptr<AudioRtpSender> sender =
- new AudioRtpSender(&audio_provider_, nullptr);
- rtc::scoped_refptr<AudioTrackInterface> track =
- AudioTrack::Create(kAudioTrackId, nullptr);
- sender->SetSsrc(kAudioSsrc);
- EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
- sender->SetTrack(track);
-
- // Calls expected from destructor.
- EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
-}
-
-// Test that an audio sender calls the expected methods on the provider once
-// it has a track and SSRC, when the SSRC is set last.
-TEST_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupTrackThenSsrc) {
- rtc::scoped_refptr<AudioRtpSender> sender =
- new AudioRtpSender(&audio_provider_, nullptr);
- rtc::scoped_refptr<AudioTrackInterface> track =
- AudioTrack::Create(kAudioTrackId, nullptr);
- sender->SetTrack(track);
- EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
- sender->SetSsrc(kAudioSsrc);
-
- // Calls expected from destructor.
- EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
-}
-
-// Test that a video sender calls the expected methods on the provider once
-// it has a track and SSRC, when the SSRC is set first.
-TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupSsrcThenTrack) {
- rtc::scoped_refptr<VideoRtpSender> sender =
- new VideoRtpSender(&video_provider_);
- sender->SetSsrc(kVideoSsrc);
- EXPECT_CALL(video_provider_,
- SetCaptureDevice(kVideoSsrc,
- video_track_->GetSource()->GetVideoCapturer()));
- EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
- sender->SetTrack(video_track_);
-
- // Calls expected from destructor.
- EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
- EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
-}
-
-// Test that a video sender calls the expected methods on the provider once
-// it has a track and SSRC, when the SSRC is set last.
-TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupTrackThenSsrc) {
- rtc::scoped_refptr<VideoRtpSender> sender =
- new VideoRtpSender(&video_provider_);
- sender->SetTrack(video_track_);
- EXPECT_CALL(video_provider_,
- SetCaptureDevice(kVideoSsrc,
- video_track_->GetSource()->GetVideoCapturer()));
- EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
- sender->SetSsrc(kVideoSsrc);
-
- // Calls expected from destructor.
- EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
- EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
-}
-
-// Test that the sender is disconnected from the provider when its SSRC is
-// set to 0.
-TEST_F(RtpSenderReceiverTest, AudioSenderSsrcSetToZero) {
- rtc::scoped_refptr<AudioTrackInterface> track =
- AudioTrack::Create(kAudioTrackId, nullptr);
- EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
- rtc::scoped_refptr<AudioRtpSender> sender =
- new AudioRtpSender(track, kStreamLabel1, &audio_provider_, nullptr);
- sender->SetSsrc(kAudioSsrc);
-
- EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
- sender->SetSsrc(0);
-
- // Make sure it's SetSsrc that called methods on the provider, and not the
- // destructor.
- EXPECT_CALL(audio_provider_, SetAudioSend(_, _, _, _)).Times(0);
-}
-
-// Test that the sender is disconnected from the provider when its SSRC is
-// set to 0.
-TEST_F(RtpSenderReceiverTest, VideoSenderSsrcSetToZero) {
- EXPECT_CALL(video_provider_,
- SetCaptureDevice(kVideoSsrc,
- video_track_->GetSource()->GetVideoCapturer()));
- EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
- rtc::scoped_refptr<VideoRtpSender> sender =
- new VideoRtpSender(video_track_, kStreamLabel1, &video_provider_);
- sender->SetSsrc(kVideoSsrc);
-
- EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
- EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
- sender->SetSsrc(0);
-
- // Make sure it's SetSsrc that called methods on the provider, and not the
- // destructor.
- EXPECT_CALL(video_provider_, SetCaptureDevice(_, _)).Times(0);
- EXPECT_CALL(video_provider_, SetVideoSend(_, _, _)).Times(0);
-}
-
-TEST_F(RtpSenderReceiverTest, AudioSenderTrackSetToNull) {
- rtc::scoped_refptr<AudioTrackInterface> track =
- AudioTrack::Create(kAudioTrackId, nullptr);
- EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
- rtc::scoped_refptr<AudioRtpSender> sender =
- new AudioRtpSender(track, kStreamLabel1, &audio_provider_, nullptr);
- sender->SetSsrc(kAudioSsrc);
-
- EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
- EXPECT_TRUE(sender->SetTrack(nullptr));
-
- // Make sure it's SetTrack that called methods on the provider, and not the
- // destructor.
- EXPECT_CALL(audio_provider_, SetAudioSend(_, _, _, _)).Times(0);
-}
-
-TEST_F(RtpSenderReceiverTest, VideoSenderTrackSetToNull) {
- EXPECT_CALL(video_provider_,
- SetCaptureDevice(kVideoSsrc,
- video_track_->GetSource()->GetVideoCapturer()));
- EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
- rtc::scoped_refptr<VideoRtpSender> sender =
- new VideoRtpSender(video_track_, kStreamLabel1, &video_provider_);
- sender->SetSsrc(kVideoSsrc);
-
- EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
- EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
- EXPECT_TRUE(sender->SetTrack(nullptr));
-
- // Make sure it's SetTrack that called methods on the provider, and not the
- // destructor.
- EXPECT_CALL(video_provider_, SetCaptureDevice(_, _)).Times(0);
- EXPECT_CALL(video_provider_, SetVideoSend(_, _, _)).Times(0);
-}
-
-TEST_F(RtpSenderReceiverTest, AudioSenderSsrcChanged) {
- rtc::scoped_refptr<AudioTrackInterface> track =
- AudioTrack::Create(kAudioTrackId, nullptr);
- EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
- rtc::scoped_refptr<AudioRtpSender> sender =
- new AudioRtpSender(track, kStreamLabel1, &audio_provider_, nullptr);
- sender->SetSsrc(kAudioSsrc);
-
- EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
- EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc2, true, _, _)).Times(1);
- sender->SetSsrc(kAudioSsrc2);
-
- // Calls expected from destructor.
- EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc2, false, _, _)).Times(1);
-}
-
-TEST_F(RtpSenderReceiverTest, VideoSenderSsrcChanged) {
- EXPECT_CALL(video_provider_,
- SetCaptureDevice(kVideoSsrc,
- video_track_->GetSource()->GetVideoCapturer()));
- EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
- rtc::scoped_refptr<VideoRtpSender> sender =
- new VideoRtpSender(video_track_, kStreamLabel1, &video_provider_);
- sender->SetSsrc(kVideoSsrc);
-
- EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
- EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
- EXPECT_CALL(video_provider_,
- SetCaptureDevice(kVideoSsrc2,
- video_track_->GetSource()->GetVideoCapturer()));
- EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, true, _));
- sender->SetSsrc(kVideoSsrc2);
-
- // Calls expected from destructor.
- EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc2, nullptr)).Times(1);
- EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, false, _)).Times(1);
-}
-
} // namespace webrtc
diff --git a/talk/app/webrtc/test/fakemediastreamsignaling.h b/talk/app/webrtc/test/fakemediastreamsignaling.h
new file mode 100644
index 0000000..562c4ad
--- /dev/null
+++ b/talk/app/webrtc/test/fakemediastreamsignaling.h
@@ -0,0 +1,140 @@
+/*
+ * libjingle
+ * Copyright 2013 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.
+ */
+
+#ifndef TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_
+#define TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_
+
+#include "talk/app/webrtc/audiotrack.h"
+#include "talk/app/webrtc/mediastreamsignaling.h"
+#include "talk/app/webrtc/videotrack.h"
+
+static const char kStream1[] = "stream1";
+static const char kVideoTrack1[] = "video1";
+static const char kAudioTrack1[] = "audio1";
+
+static const char kStream2[] = "stream2";
+static const char kVideoTrack2[] = "video2";
+static const char kAudioTrack2[] = "audio2";
+
+class FakeMediaStreamSignaling : public webrtc::MediaStreamSignaling,
+ public webrtc::MediaStreamSignalingObserver {
+ public:
+ explicit FakeMediaStreamSignaling(cricket::ChannelManager* channel_manager) :
+ webrtc::MediaStreamSignaling(rtc::Thread::Current(), this,
+ channel_manager) {
+ }
+
+ void SendAudioVideoStream1() {
+ ClearLocalStreams();
+ AddLocalStream(CreateStream(kStream1, kAudioTrack1, kVideoTrack1));
+ }
+
+ void SendAudioVideoStream2() {
+ ClearLocalStreams();
+ AddLocalStream(CreateStream(kStream2, kAudioTrack2, kVideoTrack2));
+ }
+
+ void SendAudioVideoStream1And2() {
+ ClearLocalStreams();
+ AddLocalStream(CreateStream(kStream1, kAudioTrack1, kVideoTrack1));
+ AddLocalStream(CreateStream(kStream2, kAudioTrack2, kVideoTrack2));
+ }
+
+ void SendNothing() {
+ ClearLocalStreams();
+ }
+
+ void UseOptionsAudioOnly() {
+ ClearLocalStreams();
+ AddLocalStream(CreateStream(kStream2, kAudioTrack2, ""));
+ }
+
+ void UseOptionsVideoOnly() {
+ ClearLocalStreams();
+ AddLocalStream(CreateStream(kStream2, "", kVideoTrack2));
+ }
+
+ void ClearLocalStreams() {
+ while (local_streams()->count() != 0) {
+ RemoveLocalStream(local_streams()->at(0));
+ }
+ }
+
+ // Implements MediaStreamSignalingObserver.
+ virtual void OnAddRemoteStream(webrtc::MediaStreamInterface* stream) {}
+ virtual void OnRemoveRemoteStream(webrtc::MediaStreamInterface* stream) {}
+ virtual void OnAddDataChannel(webrtc::DataChannelInterface* data_channel) {}
+ virtual void OnAddLocalAudioTrack(webrtc::MediaStreamInterface* stream,
+ webrtc::AudioTrackInterface* audio_track,
+ uint32_t ssrc) {}
+ virtual void OnAddLocalVideoTrack(webrtc::MediaStreamInterface* stream,
+ webrtc::VideoTrackInterface* video_track,
+ uint32_t ssrc) {}
+ virtual void OnAddRemoteAudioTrack(webrtc::MediaStreamInterface* stream,
+ webrtc::AudioTrackInterface* audio_track,
+ uint32_t ssrc) {}
+ virtual void OnAddRemoteVideoTrack(webrtc::MediaStreamInterface* stream,
+ webrtc::VideoTrackInterface* video_track,
+ uint32_t ssrc) {}
+ virtual void OnRemoveRemoteAudioTrack(
+ webrtc::MediaStreamInterface* stream,
+ webrtc::AudioTrackInterface* audio_track) {}
+ virtual void OnRemoveRemoteVideoTrack(
+ webrtc::MediaStreamInterface* stream,
+ webrtc::VideoTrackInterface* video_track) {}
+ virtual void OnRemoveLocalAudioTrack(webrtc::MediaStreamInterface* stream,
+ webrtc::AudioTrackInterface* audio_track,
+ uint32_t ssrc) {}
+ virtual void OnRemoveLocalVideoTrack(
+ webrtc::MediaStreamInterface* stream,
+ webrtc::VideoTrackInterface* video_track) {}
+ virtual void OnRemoveLocalStream(webrtc::MediaStreamInterface* stream) {}
+
+ private:
+ rtc::scoped_refptr<webrtc::MediaStreamInterface> CreateStream(
+ const std::string& stream_label,
+ const std::string& audio_track_id,
+ const std::string& video_track_id) {
+ rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
+ webrtc::MediaStream::Create(stream_label));
+
+ if (!audio_track_id.empty()) {
+ rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
+ webrtc::AudioTrack::Create(audio_track_id, NULL));
+ stream->AddTrack(audio_track);
+ }
+
+ if (!video_track_id.empty()) {
+ rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
+ webrtc::VideoTrack::Create(video_track_id, NULL));
+ stream->AddTrack(video_track);
+ }
+ return stream;
+ }
+};
+
+#endif // TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_
diff --git a/talk/app/webrtc/videotrack.cc b/talk/app/webrtc/videotrack.cc
index 3222735..7c78aea 100644
--- a/talk/app/webrtc/videotrack.cc
+++ b/talk/app/webrtc/videotrack.cc
@@ -31,7 +31,7 @@
namespace webrtc {
-const char MediaStreamTrackInterface::kVideoTrackKind[] = "video";
+static const char kVideoTrackKind[] = "video";
VideoTrack::VideoTrack(const std::string& label,
VideoSourceInterface* video_source)
diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc
index 24454dc..95abeab 100644
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@@ -1747,6 +1747,7 @@
cricket::GetFirstVideoContent(desc);
if ((!video_info || video_info->rejected) && video_channel_) {
SignalVideoChannelDestroyed();
+ const std::string content_name = video_channel_->content_name();
channel_manager_->DestroyVideoChannel(video_channel_.release());
}
@@ -1754,6 +1755,7 @@
cricket::GetFirstAudioContent(desc);
if ((!voice_info || voice_info->rejected) && voice_channel_) {
SignalVoiceChannelDestroyed();
+ const std::string content_name = voice_channel_->content_name();
channel_manager_->DestroyVoiceChannel(voice_channel_.release());
}
@@ -1761,6 +1763,7 @@
cricket::GetFirstDataContent(desc);
if ((!data_info || data_info->rejected) && data_channel_) {
SignalDataChannelDestroyed();
+ const std::string content_name = data_channel_->content_name();
channel_manager_->DestroyDataChannel(data_channel_.release());
}
}
diff --git a/talk/libjingle_tests.gyp b/talk/libjingle_tests.gyp
index a8bf57d..41b38b3 100755
--- a/talk/libjingle_tests.gyp
+++ b/talk/libjingle_tests.gyp
@@ -215,6 +215,7 @@
'app/webrtc/test/fakeconstraints.h',
'app/webrtc/test/fakedatachannelprovider.h',
'app/webrtc/test/fakedtlsidentitystore.h',
+ 'app/webrtc/test/fakemediastreamsignaling.h',
'app/webrtc/test/fakeperiodicvideocapturer.h',
'app/webrtc/test/fakevideotrackrenderer.h',
'app/webrtc/test/mockpeerconnectionobservers.h',
diff --git a/webrtc/base/helpers.cc b/webrtc/base/helpers.cc
index 1ad5d0e..8e59b64 100644
--- a/webrtc/base/helpers.cc
+++ b/webrtc/base/helpers.cc
@@ -164,20 +164,16 @@
int seed_;
};
-namespace {
-
// TODO: Use Base64::Base64Table instead.
-static const char kBase64[64] = {
- 'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H', 'I', 'J', 'K', 'L', 'M',
- 'N', 'O', 'P', 'Q', 'R', 'S', 'T', 'U', 'V', 'W', 'X', 'Y', 'Z',
- 'a', 'b', 'c', 'd', 'e', 'f', 'g', 'h', 'i', 'j', 'k', 'l', 'm',
- 'n', 'o', 'p', 'q', 'r', 's', 't', 'u', 'v', 'w', 'x', 'y', 'z',
- '0', '1', '2', '3', '4', '5', '6', '7', '8', '9', '+', '/'};
+static const char BASE64[64] = {
+ 'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H', 'I', 'J', 'K', 'L', 'M',
+ 'N', 'O', 'P', 'Q', 'R', 'S', 'T', 'U', 'V', 'W', 'X', 'Y', 'Z',
+ 'a', 'b', 'c', 'd', 'e', 'f', 'g', 'h', 'i', 'j', 'k', 'l', 'm',
+ 'n', 'o', 'p', 'q', 'r', 's', 't', 'u', 'v', 'w', 'x', 'y', 'z',
+ '0', '1', '2', '3', '4', '5', '6', '7', '8', '9', '+', '/'
+};
-static const char kHex[16] = {'0', '1', '2', '3', '4', '5', '6', '7',
- '8', '9', 'a', 'b', 'c', 'd', 'e', 'f'};
-
-static const char kUuidDigit17[4] = {'8', '9', 'a', 'b'};
+namespace {
// This round about way of creating a global RNG is to safe-guard against
// indeterminant static initialization order.
@@ -236,7 +232,7 @@
}
bool CreateRandomString(size_t len, std::string* str) {
- return CreateRandomString(len, kBase64, 64, str);
+ return CreateRandomString(len, BASE64, 64, str);
}
bool CreateRandomString(size_t len, const std::string& table,
@@ -245,41 +241,6 @@
static_cast<int>(table.size()), str);
}
-// Version 4 UUID is of the form:
-// xxxxxxxx-xxxx-4xxx-yxxx-xxxxxxxxxxxx
-// Where 'x' is a hex digit, and 'y' is 8, 9, a or b.
-std::string CreateRandomUuid() {
- std::string str;
- scoped_ptr<uint8_t[]> bytes(new uint8_t[31]);
- if (!Rng().Generate(bytes.get(), 31)) {
- LOG(LS_ERROR) << "Failed to generate random string!";
- return str;
- }
- str.reserve(36);
- for (size_t i = 0; i < 8; ++i) {
- str.push_back(kHex[bytes[i] % 16]);
- }
- str.push_back('-');
- for (size_t i = 8; i < 12; ++i) {
- str.push_back(kHex[bytes[i] % 16]);
- }
- str.push_back('-');
- str.push_back('4');
- for (size_t i = 12; i < 15; ++i) {
- str.push_back(kHex[bytes[i] % 16]);
- }
- str.push_back('-');
- str.push_back(kUuidDigit17[bytes[15] % 4]);
- for (size_t i = 16; i < 19; ++i) {
- str.push_back(kHex[bytes[i] % 16]);
- }
- str.push_back('-');
- for (size_t i = 19; i < 31; ++i) {
- str.push_back(kHex[bytes[i] % 16]);
- }
- return str;
-}
-
uint32_t CreateRandomId() {
uint32_t id;
if (!Rng().Generate(&id, sizeof(id))) {
diff --git a/webrtc/base/helpers.h b/webrtc/base/helpers.h
index 0e79373..102c08b 100644
--- a/webrtc/base/helpers.h
+++ b/webrtc/base/helpers.h
@@ -39,9 +39,6 @@
bool CreateRandomString(size_t length, const std::string& table,
std::string* str);
-// Generates a (cryptographically) random UUID version 4 string.
-std::string CreateRandomUuid();
-
// Generates a random id.
uint32_t CreateRandomId();
diff --git a/webrtc/base/helpers_unittest.cc b/webrtc/base/helpers_unittest.cc
index 83cc685..6ea0167 100644
--- a/webrtc/base/helpers_unittest.cc
+++ b/webrtc/base/helpers_unittest.cc
@@ -43,23 +43,16 @@
EXPECT_EQ(256U, random2.size());
}
-TEST_F(RandomTest, TestCreateRandomUuid) {
- std::string random = CreateRandomUuid();
- EXPECT_EQ(36U, random.size());
-}
-
TEST_F(RandomTest, TestCreateRandomForTest) {
// Make sure we get the output we expect.
SetRandomTestMode(true);
EXPECT_EQ(2154761789U, CreateRandomId());
EXPECT_EQ("h0ISP4S5SJKH/9EY", CreateRandomString(16));
- EXPECT_EQ("41706e92-cdd3-46d9-a22d-8ff1737ffb11", CreateRandomUuid());
// Reset and make sure we get the same output.
SetRandomTestMode(true);
EXPECT_EQ(2154761789U, CreateRandomId());
EXPECT_EQ("h0ISP4S5SJKH/9EY", CreateRandomString(16));
- EXPECT_EQ("41706e92-cdd3-46d9-a22d-8ff1737ffb11", CreateRandomUuid());
// Test different character sets.
SetRandomTestMode(true);
