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henrike@webrtc.org28e20752013-07-10 00:45:361/*
kjellanderb24317b2016-02-10 15:54:432 * Copyright 2012 The WebRTC project authors. All Rights Reserved.
henrike@webrtc.org28e20752013-07-10 00:45:363 *
kjellanderb24317b2016-02-10 15:54:434 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
henrike@webrtc.org28e20752013-07-10 00:45:369 */
10
Mirko Bonadei92ea95e2017-09-15 04:47:3111#ifndef PC_PEERCONNECTION_H_
12#define PC_PEERCONNECTION_H_
henrike@webrtc.org28e20752013-07-10 00:45:3613
14#include <string>
perkjd61bf802016-03-24 10:16:1915#include <map>
kwibergd1fe2812016-04-27 13:47:2916#include <memory>
perkjd61bf802016-03-24 10:16:1917#include <vector>
henrike@webrtc.org28e20752013-07-10 00:45:3618
Mirko Bonadei92ea95e2017-09-15 04:47:3119#include "api/peerconnectioninterface.h"
20#include "pc/iceserverparsing.h"
21#include "pc/peerconnectionfactory.h"
22#include "pc/rtcstatscollector.h"
23#include "pc/rtpreceiver.h"
24#include "pc/rtpsender.h"
25#include "pc/statscollector.h"
26#include "pc/streamcollection.h"
27#include "pc/webrtcsession.h"
henrike@webrtc.org28e20752013-07-10 00:45:3628
29namespace webrtc {
henrike@webrtc.org28e20752013-07-10 00:45:3630
deadbeefeb459812015-12-16 03:24:4331class MediaStreamObserver;
perkjf0dcfe22016-03-10 17:32:0032class VideoRtpReceiver;
skvlad11a9cbf2016-10-07 18:53:0533class RtcEventLog;
deadbeefab9b2d12015-10-14 18:33:1134
zhihuang1c378ed2017-08-17 21:10:5035// TODO(zhihuang): Remove this declaration when the WebRtcSession tests don't
36// need it.
37void ExtractSharedMediaSessionOptions(
deadbeefab9b2d12015-10-14 18:33:1138 const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
39 cricket::MediaSessionOptions* session_options);
40
deadbeef70ab1a12015-09-28 23:53:5541// PeerConnection implements the PeerConnectionInterface interface.
deadbeefab9b2d12015-10-14 18:33:1142// It uses WebRtcSession to implement the PeerConnection functionality.
henrike@webrtc.org28e20752013-07-10 00:45:3643class PeerConnection : public PeerConnectionInterface,
henrike@webrtc.org28e20752013-07-10 00:45:3644 public IceObserver,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:5245 public rtc::MessageHandler,
henrike@webrtc.org28e20752013-07-10 00:45:3646 public sigslot::has_slots<> {
47 public:
zhihuang38ede132017-06-15 19:52:3248 explicit PeerConnection(PeerConnectionFactory* factory,
49 std::unique_ptr<RtcEventLog> event_log,
50 std::unique_ptr<Call> call);
henrike@webrtc.org28e20752013-07-10 00:45:3651
deadbeef653b8e02015-11-11 20:55:1052 bool Initialize(
53 const PeerConnectionInterface::RTCConfiguration& configuration,
kwibergd1fe2812016-04-27 13:47:2954 std::unique_ptr<cricket::PortAllocator> allocator,
Henrik Boströmd03c23b2016-06-01 09:44:1855 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
deadbeef653b8e02015-11-11 20:55:1056 PeerConnectionObserver* observer);
57
deadbeefa67696b2015-09-29 18:56:2658 rtc::scoped_refptr<StreamCollectionInterface> local_streams() override;
59 rtc::scoped_refptr<StreamCollectionInterface> remote_streams() override;
60 bool AddStream(MediaStreamInterface* local_stream) override;
61 void RemoveStream(MediaStreamInterface* local_stream) override;
henrike@webrtc.org28e20752013-07-10 00:45:3662
deadbeefe1f9d832016-01-14 23:35:4263 rtc::scoped_refptr<RtpSenderInterface> AddTrack(
64 MediaStreamTrackInterface* track,
65 std::vector<MediaStreamInterface*> streams) override;
66 bool RemoveTrack(RtpSenderInterface* sender) override;
67
Steve Anton978b8762017-09-29 19:15:0268 // TODO(steveanton): Remove this once all clients have switched to using the
69 // PeerConnection shims for WebRtcSession methods instead of the methods
70 // directly via this getter.
71 virtual WebRtcSession* session() { return session_; }
Alex Loikobf667942017-09-29 10:44:3172
Steve Anton8c0f7a72017-10-03 17:03:1073 // Gets the DTLS SSL certificate associated with the audio transport on the
74 // remote side. This will become populated once the DTLS connection with the
75 // peer has been completed, as indicated by the ICE connection state
76 // transitioning to kIceConnectionCompleted.
77 // Note that this will be removed once we implement RTCDtlsTransport which
78 // has standardized method for getting this information.
79 // See https://www.w3.org/TR/webrtc/#rtcdtlstransport-interface
80 std::unique_ptr<rtc::SSLCertificate> GetRemoteAudioSSLCertificate();
81
deadbeefa67696b2015-09-29 18:56:2682 rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
83 AudioTrackInterface* track) override;
henrike@webrtc.org28e20752013-07-10 00:45:3684
deadbeeffac06552015-11-25 19:26:0185 rtc::scoped_refptr<RtpSenderInterface> CreateSender(
deadbeefbd7d8f72015-12-19 00:58:4486 const std::string& kind,
87 const std::string& stream_id) override;
deadbeeffac06552015-11-25 19:26:0188
deadbeef70ab1a12015-09-28 23:53:5589 std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
90 const override;
91 std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
92 const override;
93
deadbeefa67696b2015-09-29 18:56:2694 rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
henrike@webrtc.org28e20752013-07-10 00:45:3695 const std::string& label,
deadbeefa67696b2015-09-29 18:56:2696 const DataChannelInit* config) override;
97 bool GetStats(StatsObserver* observer,
98 webrtc::MediaStreamTrackInterface* track,
99 StatsOutputLevel level) override;
hbos74e1a4f2016-09-16 06:33:01100 void GetStats(RTCStatsCollectorCallback* callback) override;
henrike@webrtc.org28e20752013-07-10 00:45:36101
deadbeefa67696b2015-09-29 18:56:26102 SignalingState signaling_state() override;
henrike@webrtc.org28e20752013-07-10 00:45:36103
deadbeefa67696b2015-09-29 18:56:26104 IceConnectionState ice_connection_state() override;
105 IceGatheringState ice_gathering_state() override;
henrike@webrtc.org28e20752013-07-10 00:45:36106
deadbeefa67696b2015-09-29 18:56:26107 const SessionDescriptionInterface* local_description() const override;
108 const SessionDescriptionInterface* remote_description() const override;
deadbeeffe4a8a42016-12-21 01:56:17109 const SessionDescriptionInterface* current_local_description() const override;
110 const SessionDescriptionInterface* current_remote_description()
111 const override;
112 const SessionDescriptionInterface* pending_local_description() const override;
113 const SessionDescriptionInterface* pending_remote_description()
114 const override;
henrike@webrtc.org28e20752013-07-10 00:45:36115
116 // JSEP01
htaa2a49d92016-03-04 10:51:39117 // Deprecated, use version without constraints.
deadbeefa67696b2015-09-29 18:56:26118 void CreateOffer(CreateSessionDescriptionObserver* observer,
119 const MediaConstraintsInterface* constraints) override;
120 void CreateOffer(CreateSessionDescriptionObserver* observer,
121 const RTCOfferAnswerOptions& options) override;
htaa2a49d92016-03-04 10:51:39122 // Deprecated, use version without constraints.
deadbeefa67696b2015-09-29 18:56:26123 void CreateAnswer(CreateSessionDescriptionObserver* observer,
124 const MediaConstraintsInterface* constraints) override;
htaa2a49d92016-03-04 10:51:39125 void CreateAnswer(CreateSessionDescriptionObserver* observer,
126 const RTCOfferAnswerOptions& options) override;
deadbeefa67696b2015-09-29 18:56:26127 void SetLocalDescription(SetSessionDescriptionObserver* observer,
128 SessionDescriptionInterface* desc) override;
129 void SetRemoteDescription(SetSessionDescriptionObserver* observer,
130 SessionDescriptionInterface* desc) override;
deadbeef46c73892016-11-17 03:42:04131 PeerConnectionInterface::RTCConfiguration GetConfiguration() override;
deadbeefa67696b2015-09-29 18:56:26132 bool SetConfiguration(
deadbeef293e9262017-01-11 20:28:30133 const PeerConnectionInterface::RTCConfiguration& configuration,
134 RTCError* error) override;
135 bool SetConfiguration(
136 const PeerConnectionInterface::RTCConfiguration& configuration) override {
137 return SetConfiguration(configuration, nullptr);
138 }
deadbeefa67696b2015-09-29 18:56:26139 bool AddIceCandidate(const IceCandidateInterface* candidate) override;
Honghai Zhang7fb69db2016-03-14 18:59:18140 bool RemoveIceCandidates(
141 const std::vector<cricket::Candidate>& candidates) override;
henrike@webrtc.org28e20752013-07-10 00:45:36142
deadbeefa67696b2015-09-29 18:56:26143 void RegisterUMAObserver(UMAObserver* observer) override;
buildbot@webrtc.org1567b8c2014-05-08 19:54:16144
zstein4b979802017-06-02 21:37:37145 RTCError SetBitrate(const BitrateParameters& bitrate) override;
146
ivoc14d5dbe2016-07-04 14:06:55147 bool StartRtcEventLog(rtc::PlatformFile file,
148 int64_t max_size_bytes) override;
149 void StopRtcEventLog() override;
150
deadbeefa67696b2015-09-29 18:56:26151 void Close() override;
henrike@webrtc.org28e20752013-07-10 00:45:36152
hbos82ebe022016-11-14 09:41:09153 sigslot::signal1<DataChannel*> SignalDataChannelCreated;
154
deadbeefab9b2d12015-10-14 18:33:11155 // Virtual for unit tests.
156 virtual const std::vector<rtc::scoped_refptr<DataChannel>>&
157 sctp_data_channels() const {
158 return sctp_data_channels_;
perkjd61bf802016-03-24 10:16:19159 }
deadbeefab9b2d12015-10-14 18:33:11160
Steve Anton978b8762017-09-29 19:15:02161 // TODO(steveanton): These methods are temporarily added to facilitate work
162 // towards merging WebRtcSession into PeerConnection. To make this easier, we
163 // want only PeerConnection to interact with WebRtcSession so they can be
164 // merged easily. A few outside classes still access WebRtcSession methods
165 // directly, so these have been added to PeerConnection to remove the
166 // dependency from WebRtcSession.
167
168 rtc::Thread* network_thread() const { return factory_->network_thread(); }
169 rtc::Thread* worker_thread() const { return factory_->worker_thread(); }
170 rtc::Thread* signaling_thread() const { return factory_->signaling_thread(); }
171 virtual const std::string& session_id() const { return session_->id(); }
172 virtual bool session_created() const { return session_ != nullptr; }
173 virtual bool initial_offerer() const { return session_->initial_offerer(); }
174 virtual std::unique_ptr<SessionStats> GetSessionStats_s() {
175 return session_->GetStats_s();
176 }
177 virtual std::unique_ptr<SessionStats> GetSessionStats(
178 const ChannelNamePairs& channel_name_pairs) {
179 return session_->GetStats(channel_name_pairs);
180 }
181 virtual bool GetLocalCertificate(
182 const std::string& transport_name,
183 rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
184 return session_->GetLocalCertificate(transport_name, certificate);
185 }
186 virtual std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate(
187 const std::string& transport_name) {
188 return session_->GetRemoteSSLCertificate(transport_name);
189 }
190 virtual Call::Stats GetCallStats() { return session_->GetCallStats(); }
191 virtual cricket::VoiceChannel* voice_channel() {
192 return session_->voice_channel();
193 }
194 virtual cricket::VideoChannel* video_channel() {
195 return session_->video_channel();
196 }
197 virtual cricket::RtpDataChannel* rtp_data_channel() {
198 return session_->rtp_data_channel();
199 }
200 virtual rtc::Optional<std::string> sctp_content_name() const {
201 return session_->sctp_content_name();
202 }
203 virtual rtc::Optional<std::string> sctp_transport_name() const {
204 return session_->sctp_transport_name();
205 }
206 virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id) {
207 return session_->GetLocalTrackIdBySsrc(ssrc, track_id);
208 }
209 virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id) {
210 return session_->GetRemoteTrackIdBySsrc(ssrc, track_id);
211 }
212
213 // This is needed for stats tests to inject a MockWebRtcSession. Once
214 // WebRtcSession has been merged in, this will no longer be needed.
215 void set_session_for_testing(WebRtcSession* session) {
216 session_ = session;
217 }
218
henrike@webrtc.org28e20752013-07-10 00:45:36219 protected:
deadbeefa67696b2015-09-29 18:56:26220 ~PeerConnection() override;
henrike@webrtc.org28e20752013-07-10 00:45:36221
222 private:
deadbeefab9b2d12015-10-14 18:33:11223 struct TrackInfo {
224 TrackInfo() : ssrc(0) {}
225 TrackInfo(const std::string& stream_label,
226 const std::string track_id,
227 uint32_t ssrc)
228 : stream_label(stream_label), track_id(track_id), ssrc(ssrc) {}
deadbeefbda7e0b2015-12-09 01:13:40229 bool operator==(const TrackInfo& other) {
230 return this->stream_label == other.stream_label &&
231 this->track_id == other.track_id && this->ssrc == other.ssrc;
232 }
deadbeefab9b2d12015-10-14 18:33:11233 std::string stream_label;
234 std::string track_id;
235 uint32_t ssrc;
236 };
237 typedef std::vector<TrackInfo> TrackInfos;
238
henrike@webrtc.org28e20752013-07-10 00:45:36239 // Implements MessageHandler.
deadbeefa67696b2015-09-29 18:56:26240 void OnMessage(rtc::Message* msg) override;
henrike@webrtc.org28e20752013-07-10 00:45:36241
deadbeefab9b2d12015-10-14 18:33:11242 void CreateAudioReceiver(MediaStreamInterface* stream,
perkjd61bf802016-03-24 10:16:19243 const std::string& track_id,
deadbeefab9b2d12015-10-14 18:33:11244 uint32_t ssrc);
perkjf0dcfe22016-03-10 17:32:00245
deadbeefab9b2d12015-10-14 18:33:11246 void CreateVideoReceiver(MediaStreamInterface* stream,
perkjf0dcfe22016-03-10 17:32:00247 const std::string& track_id,
deadbeefab9b2d12015-10-14 18:33:11248 uint32_t ssrc);
Henrik Boström933d8b02017-10-10 17:05:16249 rtc::scoped_refptr<RtpReceiverInterface> RemoveAndStopReceiver(
250 const std::string& track_id);
korniltsev.anatolyec390b52017-07-25 00:00:25251
252 // May be called either by AddStream/RemoveStream, or when a track is
253 // added/removed from a stream previously added via AddStream.
254 void AddAudioTrack(AudioTrackInterface* track, MediaStreamInterface* stream);
255 void RemoveAudioTrack(AudioTrackInterface* track,
256 MediaStreamInterface* stream);
257 void AddVideoTrack(VideoTrackInterface* track, MediaStreamInterface* stream);
258 void RemoveVideoTrack(VideoTrackInterface* track,
259 MediaStreamInterface* stream);
henrike@webrtc.org28e20752013-07-10 00:45:36260
261 // Implements IceObserver
zstein6dfd53a2017-03-06 21:49:03262 void OnIceConnectionStateChange(IceConnectionState new_state) override;
Peter Thatcher54360512015-07-08 18:08:35263 void OnIceGatheringChange(IceGatheringState new_state) override;
jbauch81bf7b02017-03-25 15:31:12264 void OnIceCandidate(
265 std::unique_ptr<IceCandidateInterface> candidate) override;
Honghai Zhang7fb69db2016-03-14 18:59:18266 void OnIceCandidatesRemoved(
267 const std::vector<cricket::Candidate>& candidates) override;
Peter Thatcher54360512015-07-08 18:08:35268 void OnIceConnectionReceivingChange(bool receiving) override;
henrike@webrtc.org28e20752013-07-10 00:45:36269
270 // Signals from WebRtcSession.
deadbeefd59daf82015-10-14 22:02:44271 void OnSessionStateChange(WebRtcSession* session, WebRtcSession::State state);
henrike@webrtc.org28e20752013-07-10 00:45:36272 void ChangeSignalingState(SignalingState signaling_state);
273
deadbeefeb459812015-12-16 03:24:43274 // Signals from MediaStreamObserver.
275 void OnAudioTrackAdded(AudioTrackInterface* track,
276 MediaStreamInterface* stream);
277 void OnAudioTrackRemoved(AudioTrackInterface* track,
278 MediaStreamInterface* stream);
279 void OnVideoTrackAdded(VideoTrackInterface* track,
280 MediaStreamInterface* stream);
281 void OnVideoTrackRemoved(VideoTrackInterface* track,
282 MediaStreamInterface* stream);
283
henrike@webrtc.org28e20752013-07-10 00:45:36284 void PostSetSessionDescriptionFailure(SetSessionDescriptionObserver* observer,
285 const std::string& error);
deadbeefab9b2d12015-10-14 18:33:11286 void PostCreateSessionDescriptionFailure(
287 CreateSessionDescriptionObserver* observer,
288 const std::string& error);
henrike@webrtc.org28e20752013-07-10 00:45:36289
290 bool IsClosed() const {
291 return signaling_state_ == PeerConnectionInterface::kClosed;
292 }
293
deadbeefab9b2d12015-10-14 18:33:11294 // Returns a MediaSessionOptions struct with options decided by |options|,
295 // the local MediaStreams and DataChannels.
zhihuang1c378ed2017-08-17 21:10:50296 void GetOptionsForOffer(
deadbeefab9b2d12015-10-14 18:33:11297 const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
298 cricket::MediaSessionOptions* session_options);
299
300 // Returns a MediaSessionOptions struct with options decided by
301 // |constraints|, the local MediaStreams and DataChannels.
zhihuang1c378ed2017-08-17 21:10:50302 void GetOptionsForAnswer(const RTCOfferAnswerOptions& options,
303 cricket::MediaSessionOptions* session_options);
htaa2a49d92016-03-04 10:51:39304
zhihuang1c378ed2017-08-17 21:10:50305 // Generates MediaDescriptionOptions for the |session_opts| based on existing
306 // local description or remote description.
307 void GenerateMediaDescriptionOptions(
308 const SessionDescriptionInterface* session_desc,
309 cricket::RtpTransceiverDirection audio_direction,
310 cricket::RtpTransceiverDirection video_direction,
311 rtc::Optional<size_t>* audio_index,
312 rtc::Optional<size_t>* video_index,
313 rtc::Optional<size_t>* data_index,
htaa2a49d92016-03-04 10:51:39314 cricket::MediaSessionOptions* session_options);
deadbeefab9b2d12015-10-14 18:33:11315
deadbeeffaac4972015-11-12 23:33:07316 // Remove all local and remote tracks of type |media_type|.
317 // Called when a media type is rejected (m-line set to port 0).
318 void RemoveTracks(cricket::MediaType media_type);
319
deadbeefbda7e0b2015-12-09 01:13:40320 // Makes sure a MediaStreamTrack is created for each StreamParam in |streams|,
321 // and existing MediaStreamTracks are removed if there is no corresponding
322 // StreamParam. If |default_track_needed| is true, a default MediaStreamTrack
323 // is created if it doesn't exist; if false, it's removed if it exists.
324 // |media_type| is the type of the |streams| and can be either audio or video.
deadbeefab9b2d12015-10-14 18:33:11325 // If a new MediaStream is created it is added to |new_streams|.
326 void UpdateRemoteStreamsList(
327 const std::vector<cricket::StreamParams>& streams,
deadbeefbda7e0b2015-12-09 01:13:40328 bool default_track_needed,
deadbeefab9b2d12015-10-14 18:33:11329 cricket::MediaType media_type,
330 StreamCollection* new_streams);
331
332 // Triggered when a remote track has been seen for the first time in a remote
333 // session description. It creates a remote MediaStreamTrackInterface
334 // implementation and triggers CreateAudioReceiver or CreateVideoReceiver.
335 void OnRemoteTrackSeen(const std::string& stream_label,
336 const std::string& track_id,
337 uint32_t ssrc,
338 cricket::MediaType media_type);
339
340 // Triggered when a remote track has been removed from a remote session
341 // description. It removes the remote track with id |track_id| from a remote
342 // MediaStream and triggers DestroyAudioReceiver or DestroyVideoReceiver.
343 void OnRemoteTrackRemoved(const std::string& stream_label,
344 const std::string& track_id,
345 cricket::MediaType media_type);
346
347 // Finds remote MediaStreams without any tracks and removes them from
348 // |remote_streams_| and notifies the observer that the MediaStreams no longer
349 // exist.
350 void UpdateEndedRemoteMediaStreams();
351
deadbeefab9b2d12015-10-14 18:33:11352 // Loops through the vector of |streams| and finds added and removed
353 // StreamParams since last time this method was called.
354 // For each new or removed StreamParam, OnLocalTrackSeen or
355 // OnLocalTrackRemoved is invoked.
356 void UpdateLocalTracks(const std::vector<cricket::StreamParams>& streams,
357 cricket::MediaType media_type);
358
359 // Triggered when a local track has been seen for the first time in a local
360 // session description.
361 // This method triggers CreateAudioSender or CreateVideoSender if the rtp
362 // streams in the local SessionDescription can be mapped to a MediaStreamTrack
363 // in a MediaStream in |local_streams_|
364 void OnLocalTrackSeen(const std::string& stream_label,
365 const std::string& track_id,
366 uint32_t ssrc,
367 cricket::MediaType media_type);
368
369 // Triggered when a local track has been removed from a local session
370 // description.
371 // This method triggers DestroyAudioSender or DestroyVideoSender if a stream
372 // has been removed from the local SessionDescription and the stream can be
373 // mapped to a MediaStreamTrack in a MediaStream in |local_streams_|.
374 void OnLocalTrackRemoved(const std::string& stream_label,
375 const std::string& track_id,
376 uint32_t ssrc,
377 cricket::MediaType media_type);
378
379 void UpdateLocalRtpDataChannels(const cricket::StreamParamsVec& streams);
380 void UpdateRemoteRtpDataChannels(const cricket::StreamParamsVec& streams);
381 void UpdateClosingRtpDataChannels(
382 const std::vector<std::string>& active_channels,
383 bool is_local_update);
384 void CreateRemoteRtpDataChannel(const std::string& label,
385 uint32_t remote_ssrc);
386
387 // Creates channel and adds it to the collection of DataChannels that will
388 // be offered in a SessionDescription.
389 rtc::scoped_refptr<DataChannel> InternalCreateDataChannel(
390 const std::string& label,
391 const InternalDataChannelInit* config);
392
393 // Checks if any data channel has been added.
394 bool HasDataChannels() const;
395
396 void AllocateSctpSids(rtc::SSLRole role);
397 void OnSctpDataChannelClosed(DataChannel* channel);
398
399 // Notifications from WebRtcSession relating to BaseChannels.
Taylor Brandstetterba29c6a2016-06-27 23:30:35400 void OnVoiceChannelCreated();
deadbeefab9b2d12015-10-14 18:33:11401 void OnVoiceChannelDestroyed();
Taylor Brandstetterba29c6a2016-06-27 23:30:35402 void OnVideoChannelCreated();
deadbeefab9b2d12015-10-14 18:33:11403 void OnVideoChannelDestroyed();
404 void OnDataChannelCreated();
405 void OnDataChannelDestroyed();
406 // Called when the cricket::DataChannel receives a message indicating that a
407 // webrtc::DataChannel should be opened.
408 void OnDataChannelOpenMessage(const std::string& label,
409 const InternalDataChannelInit& config);
410
zhihuang1c378ed2017-08-17 21:10:50411 bool HasRtpSender(cricket::MediaType type) const;
deadbeefa601f5c2016-06-06 21:27:39412 RtpSenderInternal* FindSenderById(const std::string& id);
deadbeeffac06552015-11-25 19:26:01413
deadbeefa601f5c2016-06-06 21:27:39414 std::vector<rtc::scoped_refptr<
415 RtpSenderProxyWithInternal<RtpSenderInternal>>>::iterator
deadbeef70ab1a12015-09-28 23:53:55416 FindSenderForTrack(MediaStreamTrackInterface* track);
deadbeefa601f5c2016-06-06 21:27:39417 std::vector<rtc::scoped_refptr<
418 RtpReceiverProxyWithInternal<RtpReceiverInternal>>>::iterator
perkjd61bf802016-03-24 10:16:19419 FindReceiverForTrack(const std::string& track_id);
deadbeef70ab1a12015-09-28 23:53:55420
deadbeefab9b2d12015-10-14 18:33:11421 TrackInfos* GetRemoteTracks(cricket::MediaType media_type);
422 TrackInfos* GetLocalTracks(cricket::MediaType media_type);
423 const TrackInfo* FindTrackInfo(const TrackInfos& infos,
424 const std::string& stream_label,
425 const std::string track_id) const;
426
427 // Returns the specified SCTP DataChannel in sctp_data_channels_,
428 // or nullptr if not found.
429 DataChannel* FindDataChannelBySid(int sid) const;
430
Taylor Brandstettera1c30352016-05-13 15:15:11431 // Called when first configuring the port allocator.
deadbeef91dd5672016-05-18 23:55:30432 bool InitializePortAllocator_n(const RTCConfiguration& configuration);
deadbeef293e9262017-01-11 20:28:30433 // Called when SetConfiguration is called to apply the supported subset
434 // of the configuration on the network thread.
435 bool ReconfigurePortAllocator_n(
436 const cricket::ServerAddresses& stun_servers,
437 const std::vector<cricket::RelayServerConfig>& turn_servers,
438 IceTransportsType type,
439 int candidate_pool_size,
Guido Urdaneta604427b2017-10-09 09:53:49440 bool prune_turn_ports);
Taylor Brandstettera1c30352016-05-13 15:15:11441
Elad Alonacb24172017-10-06 12:32:13442 // Starts recording an RTC event log using the supplied platform file.
ivoc14d5dbe2016-07-04 14:06:55443 // This function should only be called from the worker thread.
444 bool StartRtcEventLog_w(rtc::PlatformFile file, int64_t max_size_bytes);
Elad Alonacb24172017-10-06 12:32:13445 // Stops recording an RTC event log.
ivoc14d5dbe2016-07-04 14:06:55446 // This function should only be called from the worker thread.
447 void StopRtcEventLog_w();
448
Steve Anton038834f2017-07-14 22:59:59449 // Ensures the configuration doesn't have any parameters with invalid values,
450 // or values that conflict with other parameters.
451 //
452 // Returns RTCError::OK() if there are no issues.
453 RTCError ValidateConfiguration(const RTCConfiguration& config) const;
454
henrike@webrtc.org28e20752013-07-10 00:45:36455 // Storing the factory as a scoped reference pointer ensures that the memory
456 // in the PeerConnectionFactoryImpl remains available as long as the
457 // PeerConnection is running. It is passed to PeerConnection as a raw pointer.
458 // However, since the reference counting is done in the
deadbeefab9b2d12015-10-14 18:33:11459 // PeerConnectionFactoryInterface all instances created using the raw pointer
henrike@webrtc.org28e20752013-07-10 00:45:36460 // will refer to the same reference count.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52461 rtc::scoped_refptr<PeerConnectionFactory> factory_;
henrike@webrtc.org28e20752013-07-10 00:45:36462 PeerConnectionObserver* observer_;
buildbot@webrtc.org1567b8c2014-05-08 19:54:16463 UMAObserver* uma_observer_;
terelius33860252017-05-13 06:37:18464
465 // The EventLog needs to outlive |call_| (and any other object that uses it).
466 std::unique_ptr<RtcEventLog> event_log_;
467
henrike@webrtc.org28e20752013-07-10 00:45:36468 SignalingState signaling_state_;
henrike@webrtc.org28e20752013-07-10 00:45:36469 IceConnectionState ice_connection_state_;
470 IceGatheringState ice_gathering_state_;
deadbeef46c73892016-11-17 03:42:04471 PeerConnectionInterface::RTCConfiguration configuration_;
henrike@webrtc.org28e20752013-07-10 00:45:36472
kwibergd1fe2812016-04-27 13:47:29473 std::unique_ptr<cricket::PortAllocator> port_allocator_;
deadbeefab9b2d12015-10-14 18:33:11474
zhihuang8f65cdf2016-05-07 01:40:30475 // One PeerConnection has only one RTCP CNAME.
476 // https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-26#section-4.9
477 std::string rtcp_cname_;
478
deadbeefab9b2d12015-10-14 18:33:11479 // Streams added via AddStream.
480 rtc::scoped_refptr<StreamCollection> local_streams_;
481 // Streams created as a result of SetRemoteDescription.
482 rtc::scoped_refptr<StreamCollection> remote_streams_;
483
kwibergd1fe2812016-04-27 13:47:29484 std::vector<std::unique_ptr<MediaStreamObserver>> stream_observers_;
deadbeefeb459812015-12-16 03:24:43485
deadbeefab9b2d12015-10-14 18:33:11486 // These lists store track info seen in local/remote descriptions.
487 TrackInfos remote_audio_tracks_;
488 TrackInfos remote_video_tracks_;
489 TrackInfos local_audio_tracks_;
490 TrackInfos local_video_tracks_;
491
492 SctpSidAllocator sid_allocator_;
493 // label -> DataChannel
494 std::map<std::string, rtc::scoped_refptr<DataChannel>> rtp_data_channels_;
495 std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_;
deadbeefbd292462015-12-15 02:15:29496 std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_to_free_;
deadbeefab9b2d12015-10-14 18:33:11497
deadbeefbda7e0b2015-12-09 01:13:40498 bool remote_peer_supports_msid_ = false;
deadbeef70ab1a12015-09-28 23:53:55499
terelius33860252017-05-13 06:37:18500 std::unique_ptr<Call> call_;
Steve Anton978b8762017-09-29 19:15:02501 WebRtcSession* session_;
502 std::unique_ptr<WebRtcSession> owned_session_;
terelius33860252017-05-13 06:37:18503 std::unique_ptr<StatsCollector> stats_; // A pointer is passed to senders_
504 rtc::scoped_refptr<RTCStatsCollector> stats_collector_;
505
deadbeefa601f5c2016-06-06 21:27:39506 std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
507 senders_;
508 std::vector<
509 rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
510 receivers_;
henrike@webrtc.org28e20752013-07-10 00:45:36511};
512
513} // namespace webrtc
514
Mirko Bonadei92ea95e2017-09-15 04:47:31515#endif // PC_PEERCONNECTION_H_