Reland of Adding the ability to create an RtpSender without a track.
(patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
Relanding after fixing CallAndModifyStream to account for new
procedures for adding/removing a track from a stream.
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}
Review URL: https://codereview.webrtc.org/1468113002
Cr-Commit-Position: refs/heads/master@{#10790}
diff --git a/talk/app/webrtc/audiotrack.cc b/talk/app/webrtc/audiotrack.cc
index b0c9129..63bd87c 100644
--- a/talk/app/webrtc/audiotrack.cc
+++ b/talk/app/webrtc/audiotrack.cc
@@ -31,7 +31,7 @@
namespace webrtc {
-static const char kAudioTrackKind[] = "audio";
+const char MediaStreamTrackInterface::kAudioKind[] = "audio";
AudioTrack::AudioTrack(const std::string& label,
AudioSourceInterface* audio_source)
@@ -40,7 +40,7 @@
}
std::string AudioTrack::kind() const {
- return kAudioTrackKind;
+ return kAudioKind;
}
rtc::scoped_refptr<AudioTrack> AudioTrack::Create(
diff --git a/talk/app/webrtc/mediastreaminterface.h b/talk/app/webrtc/mediastreaminterface.h
index 5911e85..00782cb 100644
--- a/talk/app/webrtc/mediastreaminterface.h
+++ b/talk/app/webrtc/mediastreaminterface.h
@@ -100,6 +100,9 @@
kFailed = 3, // Track negotiation failed.
};
+ static const char kAudioKind[];
+ static const char kVideoKind[];
+
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 1c62daf..a78b55a 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 AudioTrackHandler classes in mediastreamhandler.h
-// to change the settings of an audio track connected to certain PeerConnection.
+// This interface is called by AudioRtpSender/Receivers 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|.
@@ -71,9 +71,8 @@
virtual ~AudioProviderInterface() {}
};
-// This interface is called by VideoTrackHandler classes in mediastreamhandler.h
-// to change the settings of a video track connected to a certain
-// PeerConnection.
+// This interface is called by VideoRtpSender/Receivers 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 caa892d..f0db139 100644
--- a/talk/app/webrtc/peerconnection.cc
+++ b/talk/app/webrtc/peerconnection.cc
@@ -60,6 +60,7 @@
using webrtc::MediaConstraintsInterface;
using webrtc::MediaStreamInterface;
using webrtc::PeerConnectionInterface;
+using webrtc::RtpSenderInterface;
using webrtc::StreamCollection;
using webrtc::StunConfigurations;
using webrtc::TurnConfigurations;
@@ -400,25 +401,15 @@
}
// Add the stream and RTP data channel info to |session_options|.
-void SetStreams(cricket::MediaSessionOptions* session_options,
- rtc::scoped_refptr<StreamCollection> streams,
- const std::map<std::string, rtc::scoped_refptr<DataChannel>>&
- rtp_data_channels) {
+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) {
session_options->streams.clear();
- 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());
- }
- }
+ for (const auto& sender : senders) {
+ session_options->AddSendStream(sender->media_type(), sender->id(),
+ sender->stream_id());
}
// Check for data channels.
@@ -733,8 +724,6 @@
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;
@@ -745,23 +734,46 @@
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()) {
- const TrackInfo* track_info =
- FindTrackInfo(local_audio_tracks_, local_stream->label(), track->id());
- if (track_info) {
- CreateAudioSender(local_stream, track.get(), track_info->ssrc);
+ 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());
}
}
for (const auto& track : local_stream->GetVideoTracks()) {
- const TrackInfo* track_info =
- FindTrackInfo(local_video_tracks_, local_stream->label(), track->id());
- if (track_info) {
- CreateVideoSender(local_stream, track.get(), track_info->ssrc);
+ 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());
}
}
@@ -771,21 +783,27 @@
}
// TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
-// indefinitely.
+// indefinitely, when we have unified plan SDP.
void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
for (const auto& track : local_stream->GetAudioTracks()) {
- const TrackInfo* track_info =
- FindTrackInfo(local_audio_tracks_, local_stream->label(), track->id());
- if (track_info) {
- DestroyAudioSender(local_stream, track.get(), track_info->ssrc);
+ auto sender = FindSenderForTrack(track.get());
+ if (sender == senders_.end()) {
+ LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
+ << " doesn't exist.";
+ continue;
}
+ (*sender)->Stop();
+ senders_.erase(sender);
}
for (const auto& track : local_stream->GetVideoTracks()) {
- const TrackInfo* track_info =
- FindTrackInfo(local_video_tracks_, local_stream->label(), track->id());
- if (track_info) {
- DestroyVideoSender(local_stream, track.get());
+ auto sender = FindSenderForTrack(track.get());
+ if (sender == senders_.end()) {
+ LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
+ << " doesn't exist.";
+ continue;
}
+ (*sender)->Stop();
+ senders_.erase(sender);
}
local_streams_->RemoveStream(local_stream);
@@ -816,6 +834,21 @@
return DtmfSenderProxy::Create(signaling_thread(), sender.get());
}
+rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
+ const std::string& kind) {
+ RtpSenderInterface* new_sender;
+ if (kind == MediaStreamTrackInterface::kAudioKind) {
+ new_sender = new AudioRtpSender(session_.get(), stats_.get());
+ } else if (kind == MediaStreamTrackInterface::kVideoKind) {
+ 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;
@@ -1324,49 +1357,6 @@
}
}
-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());
@@ -1442,7 +1432,7 @@
return false;
}
- SetStreams(session_options, local_streams_, rtp_data_channels_);
+ AddSendStreams(session_options, senders_, 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) {
@@ -1475,7 +1465,7 @@
return false;
}
- SetStreams(session_options, local_streams_, rtp_data_channels_);
+ AddSendStreams(session_options, senders_, rtp_data_channels_);
session_options->bundle_enabled =
session_options->bundle_enabled &&
(session_options->has_audio() || session_options->has_video() ||
@@ -1503,8 +1493,7 @@
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;
@@ -1706,62 +1695,44 @@
const std::string& track_id,
uint32_t ssrc,
cricket::MediaType media_type) {
- MediaStreamInterface* stream = local_streams_->find(stream_label);
- if (!stream) {
- LOG(LS_WARNING) << "An unknown local MediaStream with label "
- << stream_label << " has been configured.";
+ RtpSenderInterface* sender = FindSenderById(track_id);
+ if (!sender) {
+ LOG(LS_WARNING) << "An unknown RtpSender with id " << track_id
+ << " has been configured in the local description.";
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");
+ if (sender->media_type() != media_type) {
+ LOG(LS_WARNING) << "An RtpSender has been configured in the local"
+ << " description with an unexpected media type.";
+ return;
}
+
+ 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) {
- MediaStreamInterface* stream = local_streams_->find(stream_label);
- if (!stream) {
- // This is the normal case. I.e., RemoveLocalStream has been called and the
+ RtpSenderInterface* sender = FindSenderById(track_id);
+ if (!sender) {
+ // This is the normal case. I.e., RemoveStream has been called and the
// SessionDescriptions has been renegotiated.
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.");
+
+ // 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;
}
+
+ sender->SetSsrc(0);
}
void PeerConnection::UpdateLocalRtpDataChannels(
@@ -1979,6 +1950,15 @@
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 cd6f671..2588020 100644
--- a/talk/app/webrtc/peerconnection.h
+++ b/talk/app/webrtc/peerconnection.h
@@ -101,6 +101,9 @@
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()
@@ -197,12 +200,6 @@
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);
@@ -342,6 +339,8 @@
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 6dc83a5..9ce6c5e 100644
--- a/talk/app/webrtc/peerconnection_unittest.cc
+++ b/talk/app/webrtc/peerconnection_unittest.cc
@@ -98,6 +98,7 @@
#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;
@@ -288,18 +289,7 @@
peer_connection_factory_->CreateLocalMediaStream(stream_label);
if (audio && can_receive_audio()) {
- 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);
+ stream->AddTrack(CreateLocalAudioTrack(stream_label));
}
if (video && can_receive_video()) {
stream->AddTrack(CreateLocalVideoTrack(stream_label));
@@ -401,6 +391,35 @@
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();
@@ -715,22 +734,6 @@
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) {
@@ -1011,7 +1014,6 @@
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);
@@ -1683,6 +1685,34 @@
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(MAYBE_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 89dd5f1..f0a79a6 100644
--- a/talk/app/webrtc/peerconnectioninterface.h
+++ b/talk/app/webrtc/peerconnectioninterface.h
@@ -334,6 +334,12 @@
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 13fcbc8..e3935f7 100644
--- a/talk/app/webrtc/peerconnectioninterface_unittest.cc
+++ b/talk/app/webrtc/peerconnectioninterface_unittest.cc
@@ -2037,8 +2037,10 @@
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 d207fbb..9b446c0 100644
--- a/talk/app/webrtc/peerconnectionproxy.h
+++ b/talk/app/webrtc/peerconnectionproxy.h
@@ -43,6 +43,9 @@
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 3a78f45..ea10b7b 100644
--- a/talk/app/webrtc/rtpsender.cc
+++ b/talk/app/webrtc/rtpsender.cc
@@ -29,6 +29,7 @@
#include "talk/app/webrtc/localaudiosource.h"
#include "talk/app/webrtc/videosourceinterface.h"
+#include "webrtc/base/helpers.h"
namespace webrtc {
@@ -59,34 +60,49 @@
}
AudioRtpSender::AudioRtpSender(AudioTrackInterface* track,
- uint32_t ssrc,
- AudioProviderInterface* provider)
+ const std::string& stream_id,
+ AudioProviderInterface* provider,
+ StatsCollector* stats)
: id_(track->id()),
- track_(track),
- ssrc_(ssrc),
+ stream_id_(stream_id),
provider_(provider),
+ stats_(stats),
+ track_(track),
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();
- Reconfigure();
+ if (can_send_track()) {
+ SetAudioSend();
+ }
}
}
bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
- if (track->kind() != "audio") {
+ if (stopped_) {
+ LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
+ return false;
+ }
+ if (track && track->kind() != MediaStreamTrackInterface::kAudioKind) {
LOG(LS_ERROR) << "SetTrack called on audio RtpSender with " << track->kind()
<< " track.";
return false;
@@ -94,36 +110,83 @@
AudioTrackInterface* audio_track = static_cast<AudioTrackInterface*>(track);
// Detach from old track.
- track_->RemoveSink(sink_adapter_.get());
- track_->UnregisterObserver(this);
+ if (track_) {
+ track_->RemoveSink(sink_adapter_.get());
+ track_->UnregisterObserver(this);
+ }
+
+ if (can_send_track() && stats_) {
+ stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
+ }
// Attach to new track.
+ bool prev_can_send_track = can_send_track();
track_ = audio_track;
- cached_track_enabled_ = track_->enabled();
- track_->RegisterObserver(this);
- track_->AddSink(sink_adapter_.get());
- Reconfigure();
+ 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);
+ }
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 (!provider_) {
+ if (stopped_) {
return;
}
- cricket::AudioOptions options;
- provider_->SetAudioSend(ssrc_, false, options, nullptr);
- provider_ = nullptr;
+ 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;
}
-void AudioRtpSender::Reconfigure() {
- if (!provider_) {
- return;
- }
+void AudioRtpSender::SetAudioSend() {
+ RTC_DCHECK(!stopped_ && can_send_track());
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 peer connection.
+ // a remote audio track to a peer connection.
options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
}
@@ -136,35 +199,42 @@
}
VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
- uint32_t ssrc,
+ const std::string& stream_id,
VideoProviderInterface* provider)
: id_(track->id()),
- track_(track),
- ssrc_(ssrc),
+ stream_id_(stream_id),
provider_(provider),
+ track_(track),
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();
- Reconfigure();
+ if (can_send_track()) {
+ SetVideoSend();
+ }
}
}
bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
- if (track->kind() != "video") {
+ if (stopped_) {
+ LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
+ return false;
+ }
+ if (track && track->kind() != MediaStreamTrackInterface::kVideoKind) {
LOG(LS_ERROR) << "SetTrack called on video RtpSender with " << track->kind()
<< " track.";
return false;
@@ -172,30 +242,72 @@
VideoTrackInterface* video_track = static_cast<VideoTrackInterface*>(track);
// Detach from old track.
- track_->UnregisterObserver(this);
+ if (track_) {
+ track_->UnregisterObserver(this);
+ }
// Attach to new track.
+ bool prev_can_send_track = can_send_track();
track_ = video_track;
- cached_track_enabled_ = track_->enabled();
- track_->RegisterObserver(this);
- Reconfigure();
+ 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);
+ }
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 (!provider_) {
+ if (stopped_) {
return;
}
- provider_->SetCaptureDevice(ssrc_, nullptr);
- provider_->SetVideoSend(ssrc_, false, nullptr);
- provider_ = nullptr;
+ if (track_) {
+ track_->UnregisterObserver(this);
+ }
+ if (can_send_track()) {
+ provider_->SetCaptureDevice(ssrc_, nullptr);
+ provider_->SetVideoSend(ssrc_, false, nullptr);
+ }
+ stopped_ = true;
}
-void VideoRtpSender::Reconfigure() {
- if (!provider_) {
- return;
- }
+void VideoRtpSender::SetVideoSend() {
+ RTC_DCHECK(!stopped_ && can_send_track());
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 3741909..d5f88a9 100644
--- a/talk/app/webrtc/rtpsender.h
+++ b/talk/app/webrtc/rtpsender.h
@@ -36,6 +36,7 @@
#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"
@@ -70,9 +71,15 @@
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,
- uint32_t ssrc,
- AudioProviderInterface* provider);
+ const std::string& stream_id,
+ AudioProviderInterface* provider,
+ StatsCollector* stats);
+
+ // Randomly generates id and stream_id.
+ AudioRtpSender(AudioProviderInterface* provider, StatsCollector* stats);
virtual ~AudioRtpSender();
@@ -85,18 +92,37 @@
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:
- void Reconfigure();
+ bool can_send_track() const { return track_ && ssrc_; }
+ // Helper function to construct options for
+ // AudioProviderInterface::SetAudioSend.
+ void SetAudioSend();
std::string id_;
- rtc::scoped_refptr<AudioTrackInterface> track_;
- uint32_t ssrc_;
+ std::string stream_id_;
AudioProviderInterface* provider_;
- bool cached_track_enabled_;
+ StatsCollector* stats_;
+ rtc::scoped_refptr<AudioTrackInterface> track_;
+ uint32_t ssrc_ = 0;
+ bool cached_track_enabled_ = false;
+ bool stopped_ = false;
// Used to pass the data callback from the |track_| to the other end of
// cricket::AudioRenderer.
@@ -107,9 +133,12 @@
public rtc::RefCountedObject<RtpSenderInterface> {
public:
VideoRtpSender(VideoTrackInterface* track,
- uint32_t ssrc,
+ const std::string& stream_id,
VideoProviderInterface* provider);
+ // Randomly generates id and stream_id.
+ explicit VideoRtpSender(VideoProviderInterface* provider);
+
virtual ~VideoRtpSender();
// ObserverInterface implementation
@@ -121,18 +150,36 @@
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:
- void Reconfigure();
+ bool can_send_track() const { return track_ && ssrc_; }
+ // Helper function to construct options for
+ // VideoProviderInterface::SetVideoSend.
+ void SetVideoSend();
std::string id_;
- rtc::scoped_refptr<VideoTrackInterface> track_;
- uint32_t ssrc_;
+ std::string stream_id_;
VideoProviderInterface* provider_;
- bool cached_track_enabled_;
+ rtc::scoped_refptr<VideoTrackInterface> track_;
+ uint32_t ssrc_ = 0;
+ bool cached_track_enabled_ = false;
+ bool stopped_ = false;
};
} // namespace webrtc
diff --git a/talk/app/webrtc/rtpsenderinterface.h b/talk/app/webrtc/rtpsenderinterface.h
index fca98f2..f54e8ca 100644
--- a/talk/app/webrtc/rtpsenderinterface.h
+++ b/talk/app/webrtc/rtpsenderinterface.h
@@ -35,6 +35,7 @@
#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"
@@ -47,10 +48,24 @@
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:
@@ -61,7 +76,12 @@
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 c9d7e00..c987186 100644
--- a/talk/app/webrtc/rtpsenderreceiver_unittest.cc
+++ b/talk/app/webrtc/rtpsenderreceiver_unittest.cc
@@ -48,7 +48,9 @@
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 {
@@ -120,8 +122,10 @@
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],
- kAudioSsrc, &audio_provider_);
+ audio_rtp_sender_ =
+ new AudioRtpSender(stream_->GetAudioTracks()[0], stream_->label(),
+ &audio_provider_, nullptr);
+ audio_rtp_sender_->SetSsrc(kAudioSsrc);
}
void CreateVideoRtpSender() {
@@ -130,7 +134,8 @@
kVideoSsrc, video_track_->GetSource()->GetVideoCapturer()));
EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
video_rtp_sender_ = new VideoRtpSender(stream_->GetVideoTracks()[0],
- kVideoSsrc, &video_provider_);
+ stream_->label(), &video_provider_);
+ video_rtp_sender_->SetSsrc(kVideoSsrc);
}
void DestroyAudioRtpSender() {
@@ -280,4 +285,206 @@
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/videotrack.cc b/talk/app/webrtc/videotrack.cc
index 7c78aea..f138240 100644
--- a/talk/app/webrtc/videotrack.cc
+++ b/talk/app/webrtc/videotrack.cc
@@ -31,7 +31,7 @@
namespace webrtc {
-static const char kVideoTrackKind[] = "video";
+const char MediaStreamTrackInterface::kVideoKind[] = "video";
VideoTrack::VideoTrack(const std::string& label,
VideoSourceInterface* video_source)
@@ -47,7 +47,7 @@
}
std::string VideoTrack::kind() const {
- return kVideoTrackKind;
+ return kVideoKind;
}
void VideoTrack::AddRenderer(VideoRendererInterface* renderer) {
diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc
index 737379e..111a76c 100644
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@@ -1741,7 +1741,6 @@
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());
}
@@ -1749,7 +1748,6 @@
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());
}
@@ -1757,7 +1755,6 @@
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/webrtc/base/helpers.cc b/webrtc/base/helpers.cc
index 8e59b64..1ad5d0e 100644
--- a/webrtc/base/helpers.cc
+++ b/webrtc/base/helpers.cc
@@ -164,17 +164,21 @@
int seed_;
};
-// TODO: Use Base64::Base64Table instead.
-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', '+', '/'
-};
-
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 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'};
+
// This round about way of creating a global RNG is to safe-guard against
// indeterminant static initialization order.
scoped_ptr<RandomGenerator>& GetGlobalRng() {
@@ -232,7 +236,7 @@
}
bool CreateRandomString(size_t len, std::string* str) {
- return CreateRandomString(len, BASE64, 64, str);
+ return CreateRandomString(len, kBase64, 64, str);
}
bool CreateRandomString(size_t len, const std::string& table,
@@ -241,6 +245,41 @@
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 102c08b..0e79373 100644
--- a/webrtc/base/helpers.h
+++ b/webrtc/base/helpers.h
@@ -39,6 +39,9 @@
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 6ea0167..83cc685 100644
--- a/webrtc/base/helpers_unittest.cc
+++ b/webrtc/base/helpers_unittest.cc
@@ -43,16 +43,23 @@
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);
