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

Issue 1413983004: Reland of Adding the ability to create an RtpSender without a track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixing merge issue. Created 5 years, 1 month ago
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Index: talk/app/webrtc/peerconnection.cc
diff --git a/talk/app/webrtc/peerconnection.cc b/talk/app/webrtc/peerconnection.cc
index caa892d00b9b04802c15908b04f97bd4ddf3cfb1..f0db1397aa7941612612ea3be8cb010b0b09c854 100644
--- a/talk/app/webrtc/peerconnection.cc
+++ b/talk/app/webrtc/peerconnection.cc
@@ -60,6 +60,7 @@ using webrtc::DataChannel;
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 @@ bool IsValidOfferToReceiveMedia(int value) {
}
// 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 @@ PeerConnection::remote_streams() {
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 @@ bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
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 @@ bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
}
// 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 @@ rtc::scoped_refptr<DtmfSenderInterface> PeerConnection::CreateDtmfSender(
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::DestroyVideoReceiver(MediaStreamInterface* stream,
}
}
-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 @@ bool PeerConnection::GetOptionsForOffer(
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 @@ bool PeerConnection::GetOptionsForAnswer(
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 @@ void PeerConnection::UpdateRemoteStreamsList(
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 @@ void PeerConnection::OnLocalTrackSeen(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) {
- 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 @@ void PeerConnection::OnDataChannelOpenMessage(
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(
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