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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
kjellanderb24317b2016-02-10 07:54:43 -08002 * Copyright 2012 The WebRTC project authors. All Rights Reserved.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00003 *
kjellanderb24317b2016-02-10 07:54:43 -08004 * 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 00:45:36 +00009 */
10
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020011#ifndef PC_PEERCONNECTION_H_
12#define PC_PEERCONNECTION_H_
henrike@webrtc.org28e20752013-07-10 00:45:36 +000013
14#include <string>
perkjd61bf802016-03-24 03:16:19 -070015#include <map>
kwibergd1fe2812016-04-27 06:47:29 -070016#include <memory>
perkjd61bf802016-03-24 03:16:19 -070017#include <vector>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000018
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020019#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 00:45:36 +000028
29namespace webrtc {
henrike@webrtc.org28e20752013-07-10 00:45:36 +000030
deadbeefeb459812015-12-15 19:24:43 -080031class MediaStreamObserver;
perkjf0dcfe22016-03-10 18:32:00 +010032class VideoRtpReceiver;
skvlad11a9cbf2016-10-07 11:53:05 -070033class RtcEventLog;
deadbeefab9b2d12015-10-14 11:33:11 -070034
zhihuang1c378ed2017-08-17 14:10:50 -070035// TODO(zhihuang): Remove this declaration when the WebRtcSession tests don't
36// need it.
37void ExtractSharedMediaSessionOptions(
deadbeefab9b2d12015-10-14 11:33:11 -070038 const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
39 cricket::MediaSessionOptions* session_options);
40
deadbeef70ab1a12015-09-28 16:53:55 -070041// PeerConnection implements the PeerConnectionInterface interface.
deadbeefab9b2d12015-10-14 11:33:11 -070042// It uses WebRtcSession to implement the PeerConnection functionality.
henrike@webrtc.org28e20752013-07-10 00:45:36 +000043class PeerConnection : public PeerConnectionInterface,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000044 public IceObserver,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000045 public rtc::MessageHandler,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000046 public sigslot::has_slots<> {
47 public:
zhihuang38ede132017-06-15 12:52:32 -070048 explicit PeerConnection(PeerConnectionFactory* factory,
49 std::unique_ptr<RtcEventLog> event_log,
50 std::unique_ptr<Call> call);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000051
deadbeef653b8e02015-11-11 12:55:10 -080052 bool Initialize(
53 const PeerConnectionInterface::RTCConfiguration& configuration,
kwibergd1fe2812016-04-27 06:47:29 -070054 std::unique_ptr<cricket::PortAllocator> allocator,
Henrik Boströmd03c23b2016-06-01 11:44:18 +020055 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
deadbeef653b8e02015-11-11 12:55:10 -080056 PeerConnectionObserver* observer);
57
deadbeefa67696b2015-09-29 11:56:26 -070058 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 00:45:36 +000062
deadbeefe1f9d832016-01-14 15:35:42 -080063 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 12:15:02 -070068 // 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 10:44:31 +000072
Steve Anton8c0f7a72017-10-03 10:03:10 -070073 // 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 11:56:26 -070082 rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
83 AudioTrackInterface* track) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000084
deadbeeffac06552015-11-25 11:26:01 -080085 rtc::scoped_refptr<RtpSenderInterface> CreateSender(
deadbeefbd7d8f72015-12-18 16:58:44 -080086 const std::string& kind,
87 const std::string& stream_id) override;
deadbeeffac06552015-11-25 11:26:01 -080088
deadbeef70ab1a12015-09-28 16:53:55 -070089 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 11:56:26 -070094 rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
henrike@webrtc.org28e20752013-07-10 00:45:36 +000095 const std::string& label,
deadbeefa67696b2015-09-29 11:56:26 -070096 const DataChannelInit* config) override;
97 bool GetStats(StatsObserver* observer,
98 webrtc::MediaStreamTrackInterface* track,
99 StatsOutputLevel level) override;
hbos74e1a4f2016-09-15 23:33:01 -0700100 void GetStats(RTCStatsCollectorCallback* callback) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000101
deadbeefa67696b2015-09-29 11:56:26 -0700102 SignalingState signaling_state() override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000103
deadbeefa67696b2015-09-29 11:56:26 -0700104 IceConnectionState ice_connection_state() override;
105 IceGatheringState ice_gathering_state() override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000106
deadbeefa67696b2015-09-29 11:56:26 -0700107 const SessionDescriptionInterface* local_description() const override;
108 const SessionDescriptionInterface* remote_description() const override;
deadbeeffe4a8a42016-12-20 17:56:17 -0800109 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 00:45:36 +0000115
116 // JSEP01
htaa2a49d92016-03-04 02:51:39 -0800117 // Deprecated, use version without constraints.
deadbeefa67696b2015-09-29 11:56:26 -0700118 void CreateOffer(CreateSessionDescriptionObserver* observer,
119 const MediaConstraintsInterface* constraints) override;
120 void CreateOffer(CreateSessionDescriptionObserver* observer,
121 const RTCOfferAnswerOptions& options) override;
htaa2a49d92016-03-04 02:51:39 -0800122 // Deprecated, use version without constraints.
deadbeefa67696b2015-09-29 11:56:26 -0700123 void CreateAnswer(CreateSessionDescriptionObserver* observer,
124 const MediaConstraintsInterface* constraints) override;
htaa2a49d92016-03-04 02:51:39 -0800125 void CreateAnswer(CreateSessionDescriptionObserver* observer,
126 const RTCOfferAnswerOptions& options) override;
deadbeefa67696b2015-09-29 11:56:26 -0700127 void SetLocalDescription(SetSessionDescriptionObserver* observer,
128 SessionDescriptionInterface* desc) override;
129 void SetRemoteDescription(SetSessionDescriptionObserver* observer,
130 SessionDescriptionInterface* desc) override;
deadbeef46c73892016-11-16 19:42:04 -0800131 PeerConnectionInterface::RTCConfiguration GetConfiguration() override;
deadbeefa67696b2015-09-29 11:56:26 -0700132 bool SetConfiguration(
deadbeef293e9262017-01-11 12:28:30 -0800133 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 11:56:26 -0700139 bool AddIceCandidate(const IceCandidateInterface* candidate) override;
Honghai Zhang7fb69db2016-03-14 11:59:18 -0700140 bool RemoveIceCandidates(
141 const std::vector<cricket::Candidate>& candidates) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000142
deadbeefa67696b2015-09-29 11:56:26 -0700143 void RegisterUMAObserver(UMAObserver* observer) override;
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +0000144
zstein4b979802017-06-02 14:37:37 -0700145 RTCError SetBitrate(const BitrateParameters& bitrate) override;
146
ivoc14d5dbe2016-07-04 07:06:55 -0700147 bool StartRtcEventLog(rtc::PlatformFile file,
148 int64_t max_size_bytes) override;
149 void StopRtcEventLog() override;
150
deadbeefa67696b2015-09-29 11:56:26 -0700151 void Close() override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000152
hbos82ebe022016-11-14 01:41:09 -0800153 sigslot::signal1<DataChannel*> SignalDataChannelCreated;
154
deadbeefab9b2d12015-10-14 11:33:11 -0700155 // 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 03:16:19 -0700159 }
deadbeefab9b2d12015-10-14 11:33:11 -0700160
Steve Anton978b8762017-09-29 12:15:02 -0700161 // 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 00:45:36 +0000219 protected:
deadbeefa67696b2015-09-29 11:56:26 -0700220 ~PeerConnection() override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000221
222 private:
deadbeefab9b2d12015-10-14 11:33:11 -0700223 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) {}
deadbeefbda7e0b2015-12-08 17:13:40 -0800229 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 11:33:11 -0700233 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 00:45:36 +0000239 // Implements MessageHandler.
deadbeefa67696b2015-09-29 11:56:26 -0700240 void OnMessage(rtc::Message* msg) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000241
deadbeefab9b2d12015-10-14 11:33:11 -0700242 void CreateAudioReceiver(MediaStreamInterface* stream,
perkjd61bf802016-03-24 03:16:19 -0700243 const std::string& track_id,
deadbeefab9b2d12015-10-14 11:33:11 -0700244 uint32_t ssrc);
perkjf0dcfe22016-03-10 18:32:00 +0100245
deadbeefab9b2d12015-10-14 11:33:11 -0700246 void CreateVideoReceiver(MediaStreamInterface* stream,
perkjf0dcfe22016-03-10 18:32:00 +0100247 const std::string& track_id,
deadbeefab9b2d12015-10-14 11:33:11 -0700248 uint32_t ssrc);
perkjd61bf802016-03-24 03:16:19 -0700249 void DestroyReceiver(const std::string& track_id);
korniltsev.anatolyec390b52017-07-24 17:00:25 -0700250
251 // May be called either by AddStream/RemoveStream, or when a track is
252 // added/removed from a stream previously added via AddStream.
253 void AddAudioTrack(AudioTrackInterface* track, MediaStreamInterface* stream);
254 void RemoveAudioTrack(AudioTrackInterface* track,
255 MediaStreamInterface* stream);
256 void AddVideoTrack(VideoTrackInterface* track, MediaStreamInterface* stream);
257 void RemoveVideoTrack(VideoTrackInterface* track,
258 MediaStreamInterface* stream);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000259
260 // Implements IceObserver
zstein6dfd53a2017-03-06 13:49:03 -0800261 void OnIceConnectionStateChange(IceConnectionState new_state) override;
Peter Thatcher54360512015-07-08 11:08:35 -0700262 void OnIceGatheringChange(IceGatheringState new_state) override;
jbauch81bf7b02017-03-25 08:31:12 -0700263 void OnIceCandidate(
264 std::unique_ptr<IceCandidateInterface> candidate) override;
Honghai Zhang7fb69db2016-03-14 11:59:18 -0700265 void OnIceCandidatesRemoved(
266 const std::vector<cricket::Candidate>& candidates) override;
Peter Thatcher54360512015-07-08 11:08:35 -0700267 void OnIceConnectionReceivingChange(bool receiving) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000268
269 // Signals from WebRtcSession.
deadbeefd59daf82015-10-14 15:02:44 -0700270 void OnSessionStateChange(WebRtcSession* session, WebRtcSession::State state);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000271 void ChangeSignalingState(SignalingState signaling_state);
272
deadbeefeb459812015-12-15 19:24:43 -0800273 // Signals from MediaStreamObserver.
274 void OnAudioTrackAdded(AudioTrackInterface* track,
275 MediaStreamInterface* stream);
276 void OnAudioTrackRemoved(AudioTrackInterface* track,
277 MediaStreamInterface* stream);
278 void OnVideoTrackAdded(VideoTrackInterface* track,
279 MediaStreamInterface* stream);
280 void OnVideoTrackRemoved(VideoTrackInterface* track,
281 MediaStreamInterface* stream);
282
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000283 void PostSetSessionDescriptionFailure(SetSessionDescriptionObserver* observer,
284 const std::string& error);
deadbeefab9b2d12015-10-14 11:33:11 -0700285 void PostCreateSessionDescriptionFailure(
286 CreateSessionDescriptionObserver* observer,
287 const std::string& error);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000288
289 bool IsClosed() const {
290 return signaling_state_ == PeerConnectionInterface::kClosed;
291 }
292
deadbeefab9b2d12015-10-14 11:33:11 -0700293 // Returns a MediaSessionOptions struct with options decided by |options|,
294 // the local MediaStreams and DataChannels.
zhihuang1c378ed2017-08-17 14:10:50 -0700295 void GetOptionsForOffer(
deadbeefab9b2d12015-10-14 11:33:11 -0700296 const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
297 cricket::MediaSessionOptions* session_options);
298
299 // Returns a MediaSessionOptions struct with options decided by
300 // |constraints|, the local MediaStreams and DataChannels.
zhihuang1c378ed2017-08-17 14:10:50 -0700301 void GetOptionsForAnswer(const RTCOfferAnswerOptions& options,
302 cricket::MediaSessionOptions* session_options);
htaa2a49d92016-03-04 02:51:39 -0800303
zhihuang1c378ed2017-08-17 14:10:50 -0700304 // Generates MediaDescriptionOptions for the |session_opts| based on existing
305 // local description or remote description.
306 void GenerateMediaDescriptionOptions(
307 const SessionDescriptionInterface* session_desc,
308 cricket::RtpTransceiverDirection audio_direction,
309 cricket::RtpTransceiverDirection video_direction,
310 rtc::Optional<size_t>* audio_index,
311 rtc::Optional<size_t>* video_index,
312 rtc::Optional<size_t>* data_index,
htaa2a49d92016-03-04 02:51:39 -0800313 cricket::MediaSessionOptions* session_options);
deadbeefab9b2d12015-10-14 11:33:11 -0700314
deadbeeffaac4972015-11-12 15:33:07 -0800315 // Remove all local and remote tracks of type |media_type|.
316 // Called when a media type is rejected (m-line set to port 0).
317 void RemoveTracks(cricket::MediaType media_type);
318
deadbeefbda7e0b2015-12-08 17:13:40 -0800319 // Makes sure a MediaStreamTrack is created for each StreamParam in |streams|,
320 // and existing MediaStreamTracks are removed if there is no corresponding
321 // StreamParam. If |default_track_needed| is true, a default MediaStreamTrack
322 // is created if it doesn't exist; if false, it's removed if it exists.
323 // |media_type| is the type of the |streams| and can be either audio or video.
deadbeefab9b2d12015-10-14 11:33:11 -0700324 // If a new MediaStream is created it is added to |new_streams|.
325 void UpdateRemoteStreamsList(
326 const std::vector<cricket::StreamParams>& streams,
deadbeefbda7e0b2015-12-08 17:13:40 -0800327 bool default_track_needed,
deadbeefab9b2d12015-10-14 11:33:11 -0700328 cricket::MediaType media_type,
329 StreamCollection* new_streams);
330
331 // Triggered when a remote track has been seen for the first time in a remote
332 // session description. It creates a remote MediaStreamTrackInterface
333 // implementation and triggers CreateAudioReceiver or CreateVideoReceiver.
334 void OnRemoteTrackSeen(const std::string& stream_label,
335 const std::string& track_id,
336 uint32_t ssrc,
337 cricket::MediaType media_type);
338
339 // Triggered when a remote track has been removed from a remote session
340 // description. It removes the remote track with id |track_id| from a remote
341 // MediaStream and triggers DestroyAudioReceiver or DestroyVideoReceiver.
342 void OnRemoteTrackRemoved(const std::string& stream_label,
343 const std::string& track_id,
344 cricket::MediaType media_type);
345
346 // Finds remote MediaStreams without any tracks and removes them from
347 // |remote_streams_| and notifies the observer that the MediaStreams no longer
348 // exist.
349 void UpdateEndedRemoteMediaStreams();
350
deadbeefab9b2d12015-10-14 11:33:11 -0700351 // Loops through the vector of |streams| and finds added and removed
352 // StreamParams since last time this method was called.
353 // For each new or removed StreamParam, OnLocalTrackSeen or
354 // OnLocalTrackRemoved is invoked.
355 void UpdateLocalTracks(const std::vector<cricket::StreamParams>& streams,
356 cricket::MediaType media_type);
357
358 // Triggered when a local track has been seen for the first time in a local
359 // session description.
360 // This method triggers CreateAudioSender or CreateVideoSender if the rtp
361 // streams in the local SessionDescription can be mapped to a MediaStreamTrack
362 // in a MediaStream in |local_streams_|
363 void OnLocalTrackSeen(const std::string& stream_label,
364 const std::string& track_id,
365 uint32_t ssrc,
366 cricket::MediaType media_type);
367
368 // Triggered when a local track has been removed from a local session
369 // description.
370 // This method triggers DestroyAudioSender or DestroyVideoSender if a stream
371 // has been removed from the local SessionDescription and the stream can be
372 // mapped to a MediaStreamTrack in a MediaStream in |local_streams_|.
373 void OnLocalTrackRemoved(const std::string& stream_label,
374 const std::string& track_id,
375 uint32_t ssrc,
376 cricket::MediaType media_type);
377
378 void UpdateLocalRtpDataChannels(const cricket::StreamParamsVec& streams);
379 void UpdateRemoteRtpDataChannels(const cricket::StreamParamsVec& streams);
380 void UpdateClosingRtpDataChannels(
381 const std::vector<std::string>& active_channels,
382 bool is_local_update);
383 void CreateRemoteRtpDataChannel(const std::string& label,
384 uint32_t remote_ssrc);
385
386 // Creates channel and adds it to the collection of DataChannels that will
387 // be offered in a SessionDescription.
388 rtc::scoped_refptr<DataChannel> InternalCreateDataChannel(
389 const std::string& label,
390 const InternalDataChannelInit* config);
391
392 // Checks if any data channel has been added.
393 bool HasDataChannels() const;
394
395 void AllocateSctpSids(rtc::SSLRole role);
396 void OnSctpDataChannelClosed(DataChannel* channel);
397
398 // Notifications from WebRtcSession relating to BaseChannels.
Taylor Brandstetterba29c6a2016-06-27 16:30:35 -0700399 void OnVoiceChannelCreated();
deadbeefab9b2d12015-10-14 11:33:11 -0700400 void OnVoiceChannelDestroyed();
Taylor Brandstetterba29c6a2016-06-27 16:30:35 -0700401 void OnVideoChannelCreated();
deadbeefab9b2d12015-10-14 11:33:11 -0700402 void OnVideoChannelDestroyed();
403 void OnDataChannelCreated();
404 void OnDataChannelDestroyed();
405 // Called when the cricket::DataChannel receives a message indicating that a
406 // webrtc::DataChannel should be opened.
407 void OnDataChannelOpenMessage(const std::string& label,
408 const InternalDataChannelInit& config);
409
zhihuang1c378ed2017-08-17 14:10:50 -0700410 bool HasRtpSender(cricket::MediaType type) const;
deadbeefa601f5c2016-06-06 14:27:39 -0700411 RtpSenderInternal* FindSenderById(const std::string& id);
deadbeeffac06552015-11-25 11:26:01 -0800412
deadbeefa601f5c2016-06-06 14:27:39 -0700413 std::vector<rtc::scoped_refptr<
414 RtpSenderProxyWithInternal<RtpSenderInternal>>>::iterator
deadbeef70ab1a12015-09-28 16:53:55 -0700415 FindSenderForTrack(MediaStreamTrackInterface* track);
deadbeefa601f5c2016-06-06 14:27:39 -0700416 std::vector<rtc::scoped_refptr<
417 RtpReceiverProxyWithInternal<RtpReceiverInternal>>>::iterator
perkjd61bf802016-03-24 03:16:19 -0700418 FindReceiverForTrack(const std::string& track_id);
deadbeef70ab1a12015-09-28 16:53:55 -0700419
deadbeefab9b2d12015-10-14 11:33:11 -0700420 TrackInfos* GetRemoteTracks(cricket::MediaType media_type);
421 TrackInfos* GetLocalTracks(cricket::MediaType media_type);
422 const TrackInfo* FindTrackInfo(const TrackInfos& infos,
423 const std::string& stream_label,
424 const std::string track_id) const;
425
426 // Returns the specified SCTP DataChannel in sctp_data_channels_,
427 // or nullptr if not found.
428 DataChannel* FindDataChannelBySid(int sid) const;
429
Taylor Brandstettera1c30352016-05-13 08:15:11 -0700430 // Called when first configuring the port allocator.
deadbeef91dd5672016-05-18 16:55:30 -0700431 bool InitializePortAllocator_n(const RTCConfiguration& configuration);
deadbeef293e9262017-01-11 12:28:30 -0800432 // Called when SetConfiguration is called to apply the supported subset
433 // of the configuration on the network thread.
434 bool ReconfigurePortAllocator_n(
435 const cricket::ServerAddresses& stun_servers,
436 const std::vector<cricket::RelayServerConfig>& turn_servers,
437 IceTransportsType type,
438 int candidate_pool_size,
439 bool prune_turn_ports);
Taylor Brandstettera1c30352016-05-13 08:15:11 -0700440
Elad Alonacb24172017-10-06 14:32:13 +0200441 // Starts recording an RTC event log using the supplied platform file.
ivoc14d5dbe2016-07-04 07:06:55 -0700442 // This function should only be called from the worker thread.
443 bool StartRtcEventLog_w(rtc::PlatformFile file, int64_t max_size_bytes);
Elad Alonacb24172017-10-06 14:32:13 +0200444 // Stops recording an RTC event log.
ivoc14d5dbe2016-07-04 07:06:55 -0700445 // This function should only be called from the worker thread.
446 void StopRtcEventLog_w();
447
Steve Anton038834f2017-07-14 15:59:59 -0700448 // Ensures the configuration doesn't have any parameters with invalid values,
449 // or values that conflict with other parameters.
450 //
451 // Returns RTCError::OK() if there are no issues.
452 RTCError ValidateConfiguration(const RTCConfiguration& config) const;
453
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000454 // Storing the factory as a scoped reference pointer ensures that the memory
455 // in the PeerConnectionFactoryImpl remains available as long as the
456 // PeerConnection is running. It is passed to PeerConnection as a raw pointer.
457 // However, since the reference counting is done in the
deadbeefab9b2d12015-10-14 11:33:11 -0700458 // PeerConnectionFactoryInterface all instances created using the raw pointer
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000459 // will refer to the same reference count.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000460 rtc::scoped_refptr<PeerConnectionFactory> factory_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000461 PeerConnectionObserver* observer_;
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +0000462 UMAObserver* uma_observer_;
terelius33860252017-05-12 23:37:18 -0700463
464 // The EventLog needs to outlive |call_| (and any other object that uses it).
465 std::unique_ptr<RtcEventLog> event_log_;
466
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000467 SignalingState signaling_state_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000468 IceConnectionState ice_connection_state_;
469 IceGatheringState ice_gathering_state_;
deadbeef46c73892016-11-16 19:42:04 -0800470 PeerConnectionInterface::RTCConfiguration configuration_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000471
kwibergd1fe2812016-04-27 06:47:29 -0700472 std::unique_ptr<cricket::PortAllocator> port_allocator_;
deadbeefab9b2d12015-10-14 11:33:11 -0700473
zhihuang8f65cdf2016-05-06 18:40:30 -0700474 // One PeerConnection has only one RTCP CNAME.
475 // https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-26#section-4.9
476 std::string rtcp_cname_;
477
deadbeefab9b2d12015-10-14 11:33:11 -0700478 // Streams added via AddStream.
479 rtc::scoped_refptr<StreamCollection> local_streams_;
480 // Streams created as a result of SetRemoteDescription.
481 rtc::scoped_refptr<StreamCollection> remote_streams_;
482
kwibergd1fe2812016-04-27 06:47:29 -0700483 std::vector<std::unique_ptr<MediaStreamObserver>> stream_observers_;
deadbeefeb459812015-12-15 19:24:43 -0800484
deadbeefab9b2d12015-10-14 11:33:11 -0700485 // These lists store track info seen in local/remote descriptions.
486 TrackInfos remote_audio_tracks_;
487 TrackInfos remote_video_tracks_;
488 TrackInfos local_audio_tracks_;
489 TrackInfos local_video_tracks_;
490
491 SctpSidAllocator sid_allocator_;
492 // label -> DataChannel
493 std::map<std::string, rtc::scoped_refptr<DataChannel>> rtp_data_channels_;
494 std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_;
deadbeefbd292462015-12-14 18:15:29 -0800495 std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_to_free_;
deadbeefab9b2d12015-10-14 11:33:11 -0700496
deadbeefbda7e0b2015-12-08 17:13:40 -0800497 bool remote_peer_supports_msid_ = false;
deadbeef70ab1a12015-09-28 16:53:55 -0700498
terelius33860252017-05-12 23:37:18 -0700499 std::unique_ptr<Call> call_;
Steve Anton978b8762017-09-29 12:15:02 -0700500 WebRtcSession* session_;
501 std::unique_ptr<WebRtcSession> owned_session_;
terelius33860252017-05-12 23:37:18 -0700502 std::unique_ptr<StatsCollector> stats_; // A pointer is passed to senders_
503 rtc::scoped_refptr<RTCStatsCollector> stats_collector_;
504
deadbeefa601f5c2016-06-06 14:27:39 -0700505 std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
506 senders_;
507 std::vector<
508 rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
509 receivers_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000510};
511
512} // namespace webrtc
513
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200514#endif // PC_PEERCONNECTION_H_