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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
2 * libjingle
jlmiller@webrtc.org5f93d0a2015-01-20 21:36:13 +00003 * Copyright 2012 Google Inc.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00004 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28#ifndef TALK_APP_WEBRTC_PEERCONNECTION_H_
29#define TALK_APP_WEBRTC_PEERCONNECTION_H_
30
31#include <string>
32
Henrik Boström5e56c592015-08-11 10:33:13 +020033#include "talk/app/webrtc/dtlsidentitystore.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000034#include "talk/app/webrtc/peerconnectionfactory.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000035#include "talk/app/webrtc/peerconnectioninterface.h"
deadbeef70ab1a12015-09-28 16:53:55 -070036#include "talk/app/webrtc/rtpreceiverinterface.h"
37#include "talk/app/webrtc/rtpsenderinterface.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000038#include "talk/app/webrtc/statscollector.h"
39#include "talk/app/webrtc/streamcollection.h"
40#include "talk/app/webrtc/webrtcsession.h"
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000041#include "webrtc/base/scoped_ptr.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000042
43namespace webrtc {
henrike@webrtc.org28e20752013-07-10 00:45:36 +000044
deadbeefeb459812015-12-15 19:24:43 -080045class MediaStreamObserver;
deadbeefab9b2d12015-10-14 11:33:11 -070046class RemoteMediaStreamFactory;
47
henrike@webrtc.org28e20752013-07-10 00:45:36 +000048typedef std::vector<PortAllocatorFactoryInterface::StunConfiguration>
49 StunConfigurations;
50typedef std::vector<PortAllocatorFactoryInterface::TurnConfiguration>
51 TurnConfigurations;
52
deadbeefab9b2d12015-10-14 11:33:11 -070053// Populates |session_options| from |rtc_options|, and returns true if options
54// are valid.
deadbeefab9b2d12015-10-14 11:33:11 -070055bool ConvertRtcOptionsForOffer(
56 const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
57 cricket::MediaSessionOptions* session_options);
58
59// Populates |session_options| from |constraints|, and returns true if all
60// mandatory constraints are satisfied.
61bool ParseConstraintsForAnswer(const MediaConstraintsInterface* constraints,
62 cricket::MediaSessionOptions* session_options);
63
deadbeef0a6c4ca2015-10-06 11:38:28 -070064// Parses the URLs for each server in |servers| to build |stun_config| and
65// |turn_config|.
66bool ParseIceServers(const PeerConnectionInterface::IceServers& servers,
67 StunConfigurations* stun_config,
68 TurnConfigurations* turn_config);
69
deadbeef70ab1a12015-09-28 16:53:55 -070070// PeerConnection implements the PeerConnectionInterface interface.
deadbeefab9b2d12015-10-14 11:33:11 -070071// It uses WebRtcSession to implement the PeerConnection functionality.
henrike@webrtc.org28e20752013-07-10 00:45:36 +000072class PeerConnection : public PeerConnectionInterface,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000073 public IceObserver,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000074 public rtc::MessageHandler,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000075 public sigslot::has_slots<> {
76 public:
77 explicit PeerConnection(PeerConnectionFactory* factory);
78
deadbeef653b8e02015-11-11 12:55:10 -080079 // TODO(deadbeef): Remove this overload of Initialize once everyone is moved
80 // to the new version.
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000081 bool Initialize(
82 const PeerConnectionInterface::RTCConfiguration& configuration,
83 const MediaConstraintsInterface* constraints,
84 PortAllocatorFactoryInterface* allocator_factory,
Henrik Boström5e56c592015-08-11 10:33:13 +020085 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
buildbot@webrtc.org41451d42014-05-03 05:39:45 +000086 PeerConnectionObserver* observer);
deadbeef653b8e02015-11-11 12:55:10 -080087
88 bool Initialize(
89 const PeerConnectionInterface::RTCConfiguration& configuration,
90 const MediaConstraintsInterface* constraints,
91 rtc::scoped_ptr<cricket::PortAllocator> allocator,
92 rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
93 PeerConnectionObserver* observer);
94
deadbeefa67696b2015-09-29 11:56:26 -070095 rtc::scoped_refptr<StreamCollectionInterface> local_streams() override;
96 rtc::scoped_refptr<StreamCollectionInterface> remote_streams() override;
97 bool AddStream(MediaStreamInterface* local_stream) override;
98 void RemoveStream(MediaStreamInterface* local_stream) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000099
deadbeefab9b2d12015-10-14 11:33:11 -0700100 virtual WebRtcSession* session() { return session_.get(); }
101
deadbeefa67696b2015-09-29 11:56:26 -0700102 rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
103 AudioTrackInterface* track) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000104
deadbeeffac06552015-11-25 11:26:01 -0800105 rtc::scoped_refptr<RtpSenderInterface> CreateSender(
106 const std::string& kind) override;
107
deadbeef70ab1a12015-09-28 16:53:55 -0700108 std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
109 const override;
110 std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
111 const override;
112
deadbeefa67696b2015-09-29 11:56:26 -0700113 rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000114 const std::string& label,
deadbeefa67696b2015-09-29 11:56:26 -0700115 const DataChannelInit* config) override;
116 bool GetStats(StatsObserver* observer,
117 webrtc::MediaStreamTrackInterface* track,
118 StatsOutputLevel level) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000119
deadbeefa67696b2015-09-29 11:56:26 -0700120 SignalingState signaling_state() override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000121
122 // TODO(bemasc): Remove ice_state() when callers are removed.
deadbeefa67696b2015-09-29 11:56:26 -0700123 IceState ice_state() override;
124 IceConnectionState ice_connection_state() override;
125 IceGatheringState ice_gathering_state() override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000126
deadbeefa67696b2015-09-29 11:56:26 -0700127 const SessionDescriptionInterface* local_description() const override;
128 const SessionDescriptionInterface* remote_description() const override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000129
130 // JSEP01
deadbeefa67696b2015-09-29 11:56:26 -0700131 void CreateOffer(CreateSessionDescriptionObserver* observer,
132 const MediaConstraintsInterface* constraints) override;
133 void CreateOffer(CreateSessionDescriptionObserver* observer,
134 const RTCOfferAnswerOptions& options) override;
135 void CreateAnswer(CreateSessionDescriptionObserver* observer,
136 const MediaConstraintsInterface* constraints) override;
137 void SetLocalDescription(SetSessionDescriptionObserver* observer,
138 SessionDescriptionInterface* desc) override;
139 void SetRemoteDescription(SetSessionDescriptionObserver* observer,
140 SessionDescriptionInterface* desc) override;
141 bool SetConfiguration(
142 const PeerConnectionInterface::RTCConfiguration& config) override;
143 bool AddIceCandidate(const IceCandidateInterface* candidate) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000144
deadbeefa67696b2015-09-29 11:56:26 -0700145 void RegisterUMAObserver(UMAObserver* observer) override;
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +0000146
deadbeefa67696b2015-09-29 11:56:26 -0700147 void Close() override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000148
deadbeefab9b2d12015-10-14 11:33:11 -0700149 // Virtual for unit tests.
150 virtual const std::vector<rtc::scoped_refptr<DataChannel>>&
151 sctp_data_channels() const {
152 return sctp_data_channels_;
153 };
154
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000155 protected:
deadbeefa67696b2015-09-29 11:56:26 -0700156 ~PeerConnection() override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000157
158 private:
deadbeefab9b2d12015-10-14 11:33:11 -0700159 struct TrackInfo {
160 TrackInfo() : ssrc(0) {}
161 TrackInfo(const std::string& stream_label,
162 const std::string track_id,
163 uint32_t ssrc)
164 : stream_label(stream_label), track_id(track_id), ssrc(ssrc) {}
deadbeefbda7e0b2015-12-08 17:13:40 -0800165 bool operator==(const TrackInfo& other) {
166 return this->stream_label == other.stream_label &&
167 this->track_id == other.track_id && this->ssrc == other.ssrc;
168 }
deadbeefab9b2d12015-10-14 11:33:11 -0700169 std::string stream_label;
170 std::string track_id;
171 uint32_t ssrc;
172 };
173 typedef std::vector<TrackInfo> TrackInfos;
174
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000175 // Implements MessageHandler.
deadbeefa67696b2015-09-29 11:56:26 -0700176 void OnMessage(rtc::Message* msg) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000177
deadbeefab9b2d12015-10-14 11:33:11 -0700178 void CreateAudioReceiver(MediaStreamInterface* stream,
179 AudioTrackInterface* audio_track,
180 uint32_t ssrc);
181 void CreateVideoReceiver(MediaStreamInterface* stream,
182 VideoTrackInterface* video_track,
183 uint32_t ssrc);
184 void DestroyAudioReceiver(MediaStreamInterface* stream,
185 AudioTrackInterface* audio_track);
186 void DestroyVideoReceiver(MediaStreamInterface* stream,
187 VideoTrackInterface* video_track);
deadbeefab9b2d12015-10-14 11:33:11 -0700188 void DestroyAudioSender(MediaStreamInterface* stream,
189 AudioTrackInterface* audio_track,
190 uint32_t ssrc);
191 void DestroyVideoSender(MediaStreamInterface* stream,
192 VideoTrackInterface* video_track);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000193
194 // Implements IceObserver
Peter Thatcher54360512015-07-08 11:08:35 -0700195 void OnIceConnectionChange(IceConnectionState new_state) override;
196 void OnIceGatheringChange(IceGatheringState new_state) override;
197 void OnIceCandidate(const IceCandidateInterface* candidate) override;
198 void OnIceComplete() override;
199 void OnIceConnectionReceivingChange(bool receiving) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000200
201 // Signals from WebRtcSession.
deadbeefd59daf82015-10-14 15:02:44 -0700202 void OnSessionStateChange(WebRtcSession* session, WebRtcSession::State state);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000203 void ChangeSignalingState(SignalingState signaling_state);
204
deadbeefeb459812015-12-15 19:24:43 -0800205 // Signals from MediaStreamObserver.
206 void OnAudioTrackAdded(AudioTrackInterface* track,
207 MediaStreamInterface* stream);
208 void OnAudioTrackRemoved(AudioTrackInterface* track,
209 MediaStreamInterface* stream);
210 void OnVideoTrackAdded(VideoTrackInterface* track,
211 MediaStreamInterface* stream);
212 void OnVideoTrackRemoved(VideoTrackInterface* track,
213 MediaStreamInterface* stream);
214
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000215 rtc::Thread* signaling_thread() const {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000216 return factory_->signaling_thread();
217 }
218
219 void PostSetSessionDescriptionFailure(SetSessionDescriptionObserver* observer,
220 const std::string& error);
deadbeefab9b2d12015-10-14 11:33:11 -0700221 void PostCreateSessionDescriptionFailure(
222 CreateSessionDescriptionObserver* observer,
223 const std::string& error);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000224
225 bool IsClosed() const {
226 return signaling_state_ == PeerConnectionInterface::kClosed;
227 }
228
deadbeefab9b2d12015-10-14 11:33:11 -0700229 // Returns a MediaSessionOptions struct with options decided by |options|,
230 // the local MediaStreams and DataChannels.
231 virtual bool GetOptionsForOffer(
232 const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
233 cricket::MediaSessionOptions* session_options);
234
235 // Returns a MediaSessionOptions struct with options decided by
236 // |constraints|, the local MediaStreams and DataChannels.
237 virtual bool GetOptionsForAnswer(
238 const MediaConstraintsInterface* constraints,
239 cricket::MediaSessionOptions* session_options);
240
deadbeeffaac4972015-11-12 15:33:07 -0800241 // Remove all local and remote tracks of type |media_type|.
242 // Called when a media type is rejected (m-line set to port 0).
243 void RemoveTracks(cricket::MediaType media_type);
244
deadbeefbda7e0b2015-12-08 17:13:40 -0800245 // Makes sure a MediaStreamTrack is created for each StreamParam in |streams|,
246 // and existing MediaStreamTracks are removed if there is no corresponding
247 // StreamParam. If |default_track_needed| is true, a default MediaStreamTrack
248 // is created if it doesn't exist; if false, it's removed if it exists.
249 // |media_type| is the type of the |streams| and can be either audio or video.
deadbeefab9b2d12015-10-14 11:33:11 -0700250 // If a new MediaStream is created it is added to |new_streams|.
251 void UpdateRemoteStreamsList(
252 const std::vector<cricket::StreamParams>& streams,
deadbeefbda7e0b2015-12-08 17:13:40 -0800253 bool default_track_needed,
deadbeefab9b2d12015-10-14 11:33:11 -0700254 cricket::MediaType media_type,
255 StreamCollection* new_streams);
256
257 // Triggered when a remote track has been seen for the first time in a remote
258 // session description. It creates a remote MediaStreamTrackInterface
259 // implementation and triggers CreateAudioReceiver or CreateVideoReceiver.
260 void OnRemoteTrackSeen(const std::string& stream_label,
261 const std::string& track_id,
262 uint32_t ssrc,
263 cricket::MediaType media_type);
264
265 // Triggered when a remote track has been removed from a remote session
266 // description. It removes the remote track with id |track_id| from a remote
267 // MediaStream and triggers DestroyAudioReceiver or DestroyVideoReceiver.
268 void OnRemoteTrackRemoved(const std::string& stream_label,
269 const std::string& track_id,
270 cricket::MediaType media_type);
271
272 // Finds remote MediaStreams without any tracks and removes them from
273 // |remote_streams_| and notifies the observer that the MediaStreams no longer
274 // exist.
275 void UpdateEndedRemoteMediaStreams();
276
deadbeefab9b2d12015-10-14 11:33:11 -0700277 // Set the MediaStreamTrackInterface::TrackState to |kEnded| on all remote
278 // tracks of type |media_type|.
279 void EndRemoteTracks(cricket::MediaType media_type);
280
281 // Loops through the vector of |streams| and finds added and removed
282 // StreamParams since last time this method was called.
283 // For each new or removed StreamParam, OnLocalTrackSeen or
284 // OnLocalTrackRemoved is invoked.
285 void UpdateLocalTracks(const std::vector<cricket::StreamParams>& streams,
286 cricket::MediaType media_type);
287
288 // Triggered when a local track has been seen for the first time in a local
289 // session description.
290 // This method triggers CreateAudioSender or CreateVideoSender if the rtp
291 // streams in the local SessionDescription can be mapped to a MediaStreamTrack
292 // in a MediaStream in |local_streams_|
293 void OnLocalTrackSeen(const std::string& stream_label,
294 const std::string& track_id,
295 uint32_t ssrc,
296 cricket::MediaType media_type);
297
298 // Triggered when a local track has been removed from a local session
299 // description.
300 // This method triggers DestroyAudioSender or DestroyVideoSender if a stream
301 // has been removed from the local SessionDescription and the stream can be
302 // mapped to a MediaStreamTrack in a MediaStream in |local_streams_|.
303 void OnLocalTrackRemoved(const std::string& stream_label,
304 const std::string& track_id,
305 uint32_t ssrc,
306 cricket::MediaType media_type);
307
308 void UpdateLocalRtpDataChannels(const cricket::StreamParamsVec& streams);
309 void UpdateRemoteRtpDataChannels(const cricket::StreamParamsVec& streams);
310 void UpdateClosingRtpDataChannels(
311 const std::vector<std::string>& active_channels,
312 bool is_local_update);
313 void CreateRemoteRtpDataChannel(const std::string& label,
314 uint32_t remote_ssrc);
315
316 // Creates channel and adds it to the collection of DataChannels that will
317 // be offered in a SessionDescription.
318 rtc::scoped_refptr<DataChannel> InternalCreateDataChannel(
319 const std::string& label,
320 const InternalDataChannelInit* config);
321
322 // Checks if any data channel has been added.
323 bool HasDataChannels() const;
324
325 void AllocateSctpSids(rtc::SSLRole role);
326 void OnSctpDataChannelClosed(DataChannel* channel);
327
328 // Notifications from WebRtcSession relating to BaseChannels.
329 void OnVoiceChannelDestroyed();
330 void OnVideoChannelDestroyed();
331 void OnDataChannelCreated();
332 void OnDataChannelDestroyed();
333 // Called when the cricket::DataChannel receives a message indicating that a
334 // webrtc::DataChannel should be opened.
335 void OnDataChannelOpenMessage(const std::string& label,
336 const InternalDataChannelInit& config);
337
deadbeeffac06552015-11-25 11:26:01 -0800338 RtpSenderInterface* FindSenderById(const std::string& id);
339
deadbeef70ab1a12015-09-28 16:53:55 -0700340 std::vector<rtc::scoped_refptr<RtpSenderInterface>>::iterator
341 FindSenderForTrack(MediaStreamTrackInterface* track);
342 std::vector<rtc::scoped_refptr<RtpReceiverInterface>>::iterator
343 FindReceiverForTrack(MediaStreamTrackInterface* track);
344
deadbeefab9b2d12015-10-14 11:33:11 -0700345 TrackInfos* GetRemoteTracks(cricket::MediaType media_type);
346 TrackInfos* GetLocalTracks(cricket::MediaType media_type);
347 const TrackInfo* FindTrackInfo(const TrackInfos& infos,
348 const std::string& stream_label,
349 const std::string track_id) const;
350
351 // Returns the specified SCTP DataChannel in sctp_data_channels_,
352 // or nullptr if not found.
353 DataChannel* FindDataChannelBySid(int sid) const;
354
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000355 // Storing the factory as a scoped reference pointer ensures that the memory
356 // in the PeerConnectionFactoryImpl remains available as long as the
357 // PeerConnection is running. It is passed to PeerConnection as a raw pointer.
358 // However, since the reference counting is done in the
deadbeefab9b2d12015-10-14 11:33:11 -0700359 // PeerConnectionFactoryInterface all instances created using the raw pointer
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000360 // will refer to the same reference count.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000361 rtc::scoped_refptr<PeerConnectionFactory> factory_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000362 PeerConnectionObserver* observer_;
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +0000363 UMAObserver* uma_observer_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000364 SignalingState signaling_state_;
365 // TODO(bemasc): Remove ice_state_.
366 IceState ice_state_;
367 IceConnectionState ice_connection_state_;
368 IceGatheringState ice_gathering_state_;
369
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000370 rtc::scoped_ptr<cricket::PortAllocator> port_allocator_;
stefanc1aeaf02015-10-15 07:26:07 -0700371 rtc::scoped_ptr<MediaControllerInterface> media_controller_;
deadbeefab9b2d12015-10-14 11:33:11 -0700372
373 // Streams added via AddStream.
374 rtc::scoped_refptr<StreamCollection> local_streams_;
375 // Streams created as a result of SetRemoteDescription.
376 rtc::scoped_refptr<StreamCollection> remote_streams_;
377
deadbeefeb459812015-12-15 19:24:43 -0800378 std::vector<rtc::scoped_ptr<MediaStreamObserver>> stream_observers_;
379
deadbeefab9b2d12015-10-14 11:33:11 -0700380 // These lists store track info seen in local/remote descriptions.
381 TrackInfos remote_audio_tracks_;
382 TrackInfos remote_video_tracks_;
383 TrackInfos local_audio_tracks_;
384 TrackInfos local_video_tracks_;
385
386 SctpSidAllocator sid_allocator_;
387 // label -> DataChannel
388 std::map<std::string, rtc::scoped_refptr<DataChannel>> rtp_data_channels_;
389 std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_;
deadbeefbd292462015-12-14 18:15:29 -0800390 std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_to_free_;
deadbeefab9b2d12015-10-14 11:33:11 -0700391
deadbeefbda7e0b2015-12-08 17:13:40 -0800392 bool remote_peer_supports_msid_ = false;
deadbeefab9b2d12015-10-14 11:33:11 -0700393 rtc::scoped_ptr<RemoteMediaStreamFactory> remote_stream_factory_;
deadbeef70ab1a12015-09-28 16:53:55 -0700394
395 std::vector<rtc::scoped_refptr<RtpSenderInterface>> senders_;
396 std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers_;
deadbeefab9b2d12015-10-14 11:33:11 -0700397
398 // The session_ scoped_ptr is declared at the bottom of PeerConnection
399 // because its destruction fires signals (such as VoiceChannelDestroyed)
400 // which will trigger some final actions in PeerConnection...
401 rtc::scoped_ptr<WebRtcSession> session_;
402 // ... But stats_ depends on session_ so it should be destroyed even earlier.
403 rtc::scoped_ptr<StatsCollector> stats_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000404};
405
406} // namespace webrtc
407
408#endif // TALK_APP_WEBRTC_PEERCONNECTION_H_