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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
2 * libjingle
3 * Copyright 2012, Google Inc.
4 *
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// This file contains the PeerConnection interface as defined in
29// http://dev.w3.org/2011/webrtc/editor/webrtc.html#peer-to-peer-connections.
30// Applications must use this interface to implement peerconnection.
31// PeerConnectionFactory class provides factory methods to create
32// peerconnection, mediastream and media tracks objects.
33//
34// The Following steps are needed to setup a typical call using Jsep.
35// 1. Create a PeerConnectionFactoryInterface. Check constructors for more
36// information about input parameters.
37// 2. Create a PeerConnection object. Provide a configuration string which
38// points either to stun or turn server to generate ICE candidates and provide
39// an object that implements the PeerConnectionObserver interface.
40// 3. Create local MediaStream and MediaTracks using the PeerConnectionFactory
41// and add it to PeerConnection by calling AddStream.
42// 4. Create an offer and serialize it and send it to the remote peer.
43// 5. Once an ice candidate have been found PeerConnection will call the
44// observer function OnIceCandidate. The candidates must also be serialized and
45// sent to the remote peer.
46// 6. Once an answer is received from the remote peer, call
47// SetLocalSessionDescription with the offer and SetRemoteSessionDescription
48// with the remote answer.
49// 7. Once a remote candidate is received from the remote peer, provide it to
50// the peerconnection by calling AddIceCandidate.
51
52
53// The Receiver of a call can decide to accept or reject the call.
54// This decision will be taken by the application not peerconnection.
55// If application decides to accept the call
56// 1. Create PeerConnectionFactoryInterface if it doesn't exist.
57// 2. Create a new PeerConnection.
58// 3. Provide the remote offer to the new PeerConnection object by calling
59// SetRemoteSessionDescription.
60// 4. Generate an answer to the remote offer by calling CreateAnswer and send it
61// back to the remote peer.
62// 5. Provide the local answer to the new PeerConnection by calling
63// SetLocalSessionDescription with the answer.
64// 6. Provide the remote ice candidates by calling AddIceCandidate.
65// 7. Once a candidate have been found PeerConnection will call the observer
66// function OnIceCandidate. Send these candidates to the remote peer.
67
68#ifndef TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_
69#define TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_
70
71#include <string>
72#include <vector>
73
74#include "talk/app/webrtc/datachannelinterface.h"
75#include "talk/app/webrtc/dtmfsenderinterface.h"
76#include "talk/app/webrtc/jsep.h"
77#include "talk/app/webrtc/mediastreaminterface.h"
78#include "talk/app/webrtc/statstypes.h"
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +000079#include "talk/app/webrtc/umametrics.h"
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000080#include "webrtc/base/fileutils.h"
81#include "webrtc/base/socketaddress.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000082
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000083namespace rtc {
henrike@webrtc.org28e20752013-07-10 00:45:36 +000084class Thread;
85}
86
87namespace cricket {
88class PortAllocator;
89class WebRtcVideoDecoderFactory;
90class WebRtcVideoEncoderFactory;
91}
92
93namespace webrtc {
94class AudioDeviceModule;
95class MediaConstraintsInterface;
96
97// MediaStream container interface.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000098class StreamCollectionInterface : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +000099 public:
100 // TODO(ronghuawu): Update the function names to c++ style, e.g. find -> Find.
101 virtual size_t count() = 0;
102 virtual MediaStreamInterface* at(size_t index) = 0;
103 virtual MediaStreamInterface* find(const std::string& label) = 0;
104 virtual MediaStreamTrackInterface* FindAudioTrack(
105 const std::string& id) = 0;
106 virtual MediaStreamTrackInterface* FindVideoTrack(
107 const std::string& id) = 0;
108
109 protected:
110 // Dtor protected as objects shouldn't be deleted via this interface.
111 ~StreamCollectionInterface() {}
112};
113
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000114class StatsObserver : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000115 public:
tommi@webrtc.org5b06b062014-08-15 08:38:30 +0000116 // TODO(tommi): Remove.
117 virtual void OnComplete(const std::vector<StatsReport>& reports) {}
118
119 // TODO(tommi): Make pure virtual and remove implementation.
120 virtual void OnComplete(const StatsReports& reports) {
121 std::vector<StatsReportCopyable> report_copies;
122 for (size_t i = 0; i < reports.size(); ++i)
123 report_copies.push_back(StatsReportCopyable(*reports[i]));
124 std::vector<StatsReport>* r =
125 reinterpret_cast<std::vector<StatsReport>*>(&report_copies);
126 OnComplete(*r);
127 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000128
129 protected:
130 virtual ~StatsObserver() {}
131};
132
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000133class MetricsObserverInterface : public rtc::RefCountInterface {
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +0000134 public:
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000135 virtual void IncrementCounter(PeerConnectionMetricsCounter type) = 0;
136 virtual void AddHistogramSample(PeerConnectionMetricsName type,
mallinath@webrtc.orgd37bcfa2014-05-12 23:10:18 +0000137 int value) = 0;
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +0000138
139 protected:
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000140 virtual ~MetricsObserverInterface() {}
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +0000141};
142
guoweis@webrtc.org7169afd2014-12-04 17:59:29 +0000143typedef MetricsObserverInterface UMAObserver;
144
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000145class PeerConnectionInterface : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000146 public:
147 // See http://dev.w3.org/2011/webrtc/editor/webrtc.html#state-definitions .
148 enum SignalingState {
149 kStable,
150 kHaveLocalOffer,
151 kHaveLocalPrAnswer,
152 kHaveRemoteOffer,
153 kHaveRemotePrAnswer,
154 kClosed,
155 };
156
157 // TODO(bemasc): Remove IceState when callers are changed to
158 // IceConnection/GatheringState.
159 enum IceState {
160 kIceNew,
161 kIceGathering,
162 kIceWaiting,
163 kIceChecking,
164 kIceConnected,
165 kIceCompleted,
166 kIceFailed,
167 kIceClosed,
168 };
169
170 enum IceGatheringState {
171 kIceGatheringNew,
172 kIceGatheringGathering,
173 kIceGatheringComplete
174 };
175
176 enum IceConnectionState {
177 kIceConnectionNew,
178 kIceConnectionChecking,
179 kIceConnectionConnected,
180 kIceConnectionCompleted,
181 kIceConnectionFailed,
182 kIceConnectionDisconnected,
183 kIceConnectionClosed,
184 };
185
186 struct IceServer {
187 std::string uri;
188 std::string username;
189 std::string password;
190 };
191 typedef std::vector<IceServer> IceServers;
192
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000193 enum IceTransportsType {
194 kNone,
195 kRelay,
196 kNoHost,
197 kAll
198 };
199
200 struct RTCConfiguration {
201 IceTransportsType type;
202 IceServers servers;
203
204 RTCConfiguration() : type(kAll) {}
205 explicit RTCConfiguration(IceTransportsType type) : type(type) {}
206 };
207
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000208 struct RTCOfferAnswerOptions {
209 static const int kUndefined = -1;
210 static const int kMaxOfferToReceiveMedia = 1;
211
212 // The default value for constraint offerToReceiveX:true.
213 static const int kOfferToReceiveMediaTrue = 1;
214
215 int offer_to_receive_video;
216 int offer_to_receive_audio;
217 bool voice_activity_detection;
218 bool ice_restart;
219 bool use_rtp_mux;
220
221 RTCOfferAnswerOptions()
222 : offer_to_receive_video(kUndefined),
223 offer_to_receive_audio(kUndefined),
224 voice_activity_detection(true),
225 ice_restart(false),
226 use_rtp_mux(true) {}
227
228 RTCOfferAnswerOptions(int offer_to_receive_video,
229 int offer_to_receive_audio,
230 bool voice_activity_detection,
231 bool ice_restart,
232 bool use_rtp_mux)
233 : offer_to_receive_video(offer_to_receive_video),
234 offer_to_receive_audio(offer_to_receive_audio),
235 voice_activity_detection(voice_activity_detection),
236 ice_restart(ice_restart),
237 use_rtp_mux(use_rtp_mux) {}
238 };
239
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000240 // Used by GetStats to decide which stats to include in the stats reports.
241 // |kStatsOutputLevelStandard| includes the standard stats for Javascript API;
242 // |kStatsOutputLevelDebug| includes both the standard stats and additional
243 // stats for debugging purposes.
244 enum StatsOutputLevel {
245 kStatsOutputLevelStandard,
246 kStatsOutputLevelDebug,
247 };
248
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000249 // Accessor methods to active local streams.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000250 virtual rtc::scoped_refptr<StreamCollectionInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000251 local_streams() = 0;
252
253 // Accessor methods to remote streams.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000254 virtual rtc::scoped_refptr<StreamCollectionInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000255 remote_streams() = 0;
256
257 // Add a new MediaStream to be sent on this PeerConnection.
258 // Note that a SessionDescription negotiation is needed before the
259 // remote peer can receive the stream.
perkj@webrtc.orgfd0efb62014-11-06 12:16:36 +0000260 virtual bool AddStream(MediaStreamInterface* stream) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000261
262 // Remove a MediaStream from this PeerConnection.
263 // Note that a SessionDescription negotiation is need before the
264 // remote peer is notified.
265 virtual void RemoveStream(MediaStreamInterface* stream) = 0;
266
267 // Returns pointer to the created DtmfSender on success.
268 // Otherwise returns NULL.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000269 virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000270 AudioTrackInterface* track) = 0;
271
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000272 virtual bool GetStats(StatsObserver* observer,
273 MediaStreamTrackInterface* track,
274 StatsOutputLevel level) = 0;
275
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000276 virtual rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000277 const std::string& label,
278 const DataChannelInit* config) = 0;
279
280 virtual const SessionDescriptionInterface* local_description() const = 0;
281 virtual const SessionDescriptionInterface* remote_description() const = 0;
282
283 // Create a new offer.
284 // The CreateSessionDescriptionObserver callback will be called when done.
285 virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000286 const MediaConstraintsInterface* constraints) {}
287
288 // TODO(jiayl): remove the default impl and the old interface when chromium
289 // code is updated.
290 virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
291 const RTCOfferAnswerOptions& options) {}
292
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000293 // Create an answer to an offer.
294 // The CreateSessionDescriptionObserver callback will be called when done.
295 virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
296 const MediaConstraintsInterface* constraints) = 0;
297 // Sets the local session description.
298 // JsepInterface takes the ownership of |desc| even if it fails.
299 // The |observer| callback will be called when done.
300 virtual void SetLocalDescription(SetSessionDescriptionObserver* observer,
301 SessionDescriptionInterface* desc) = 0;
302 // Sets the remote session description.
303 // JsepInterface takes the ownership of |desc| even if it fails.
304 // The |observer| callback will be called when done.
305 virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
306 SessionDescriptionInterface* desc) = 0;
307 // Restarts or updates the ICE Agent process of gathering local candidates
308 // and pinging remote candidates.
309 virtual bool UpdateIce(const IceServers& configuration,
310 const MediaConstraintsInterface* constraints) = 0;
311 // Provides a remote candidate to the ICE Agent.
312 // A copy of the |candidate| will be created and added to the remote
313 // description. So the caller of this method still has the ownership of the
314 // |candidate|.
315 // TODO(ronghuawu): Consider to change this so that the AddIceCandidate will
316 // take the ownership of the |candidate|.
317 virtual bool AddIceCandidate(const IceCandidateInterface* candidate) = 0;
318
buildbot@webrtc.org1567b8c2014-05-08 19:54:16 +0000319 virtual void RegisterUMAObserver(UMAObserver* observer) = 0;
320
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000321 // Returns the current SignalingState.
322 virtual SignalingState signaling_state() = 0;
323
324 // TODO(bemasc): Remove ice_state when callers are changed to
325 // IceConnection/GatheringState.
326 // Returns the current IceState.
327 virtual IceState ice_state() = 0;
328 virtual IceConnectionState ice_connection_state() = 0;
329 virtual IceGatheringState ice_gathering_state() = 0;
330
331 // Terminates all media and closes the transport.
332 virtual void Close() = 0;
333
334 protected:
335 // Dtor protected as objects shouldn't be deleted via this interface.
336 ~PeerConnectionInterface() {}
337};
338
339// PeerConnection callback interface. Application should implement these
340// methods.
341class PeerConnectionObserver {
342 public:
343 enum StateType {
344 kSignalingState,
345 kIceState,
346 };
347
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000348 // Triggered when the SignalingState changed.
349 virtual void OnSignalingChange(
350 PeerConnectionInterface::SignalingState new_state) {}
351
352 // Triggered when SignalingState or IceState have changed.
353 // TODO(bemasc): Remove once callers transition to OnSignalingChange.
354 virtual void OnStateChange(StateType state_changed) {}
355
356 // Triggered when media is received on a new stream from remote peer.
357 virtual void OnAddStream(MediaStreamInterface* stream) = 0;
358
359 // Triggered when a remote peer close a stream.
360 virtual void OnRemoveStream(MediaStreamInterface* stream) = 0;
361
362 // Triggered when a remote peer open a data channel.
perkj@webrtc.orgc2dd5ee2014-11-04 11:31:29 +0000363 virtual void OnDataChannel(DataChannelInterface* data_channel) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000364
mallinath@webrtc.org0d92ef62014-01-22 02:21:22 +0000365 // Triggered when renegotiation is needed, for example the ICE has restarted.
fischman@webrtc.orgd7568a02014-01-13 22:04:12 +0000366 virtual void OnRenegotiationNeeded() = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000367
368 // Called any time the IceConnectionState changes
369 virtual void OnIceConnectionChange(
370 PeerConnectionInterface::IceConnectionState new_state) {}
371
372 // Called any time the IceGatheringState changes
373 virtual void OnIceGatheringChange(
374 PeerConnectionInterface::IceGatheringState new_state) {}
375
376 // New Ice candidate have been found.
377 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
378
379 // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
380 // All Ice candidates have been found.
381 virtual void OnIceComplete() {}
382
383 protected:
384 // Dtor protected as objects shouldn't be deleted via this interface.
385 ~PeerConnectionObserver() {}
386};
387
388// Factory class used for creating cricket::PortAllocator that is used
389// for ICE negotiation.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000390class PortAllocatorFactoryInterface : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000391 public:
392 struct StunConfiguration {
393 StunConfiguration(const std::string& address, int port)
394 : server(address, port) {}
395 // STUN server address and port.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000396 rtc::SocketAddress server;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000397 };
398
399 struct TurnConfiguration {
400 TurnConfiguration(const std::string& address,
401 int port,
402 const std::string& username,
403 const std::string& password,
wu@webrtc.org91053e72013-08-10 07:18:04 +0000404 const std::string& transport_type,
405 bool secure)
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000406 : server(address, port),
407 username(username),
408 password(password),
wu@webrtc.org91053e72013-08-10 07:18:04 +0000409 transport_type(transport_type),
410 secure(secure) {}
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000411 rtc::SocketAddress server;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000412 std::string username;
413 std::string password;
414 std::string transport_type;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000415 bool secure;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000416 };
417
418 virtual cricket::PortAllocator* CreatePortAllocator(
419 const std::vector<StunConfiguration>& stun_servers,
420 const std::vector<TurnConfiguration>& turn_configurations) = 0;
421
422 protected:
423 PortAllocatorFactoryInterface() {}
424 ~PortAllocatorFactoryInterface() {}
425};
426
427// Used to receive callbacks of DTLS identity requests.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000428class DTLSIdentityRequestObserver : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000429 public:
430 virtual void OnFailure(int error) = 0;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000431 virtual void OnSuccess(const std::string& der_cert,
432 const std::string& der_private_key) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000433 protected:
434 virtual ~DTLSIdentityRequestObserver() {}
435};
436
437class DTLSIdentityServiceInterface {
438 public:
439 // Asynchronously request a DTLS identity, including a self-signed certificate
440 // and the private key used to sign the certificate, from the identity store
441 // for the given identity name.
442 // DTLSIdentityRequestObserver::OnSuccess will be called with the identity if
443 // the request succeeded; DTLSIdentityRequestObserver::OnFailure will be
444 // called with an error code if the request failed.
445 //
446 // Only one request can be made at a time. If a second request is called
447 // before the first one completes, RequestIdentity will abort and return
448 // false.
449 //
450 // |identity_name| is an internal name selected by the client to identify an
451 // identity within an origin. E.g. an web site may cache the certificates used
452 // to communicate with differnent peers under different identity names.
453 //
454 // |common_name| is the common name used to generate the certificate. If the
455 // certificate already exists in the store, |common_name| is ignored.
456 //
457 // |observer| is the object to receive success or failure callbacks.
458 //
459 // Returns true if either OnFailure or OnSuccess will be called.
460 virtual bool RequestIdentity(
461 const std::string& identity_name,
462 const std::string& common_name,
463 DTLSIdentityRequestObserver* observer) = 0;
wu@webrtc.org91053e72013-08-10 07:18:04 +0000464
465 virtual ~DTLSIdentityServiceInterface() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000466};
467
468// PeerConnectionFactoryInterface is the factory interface use for creating
469// PeerConnection, MediaStream and media tracks.
470// PeerConnectionFactoryInterface will create required libjingle threads,
471// socket and network manager factory classes for networking.
472// If an application decides to provide its own threads and network
473// implementation of these classes it should use the alternate
474// CreatePeerConnectionFactory method which accepts threads as input and use the
475// CreatePeerConnection version that takes a PortAllocatorFactoryInterface as
476// argument.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000477class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000478 public:
wu@webrtc.org97077a32013-10-25 21:18:33 +0000479 class Options {
480 public:
481 Options() :
wu@webrtc.org97077a32013-10-25 21:18:33 +0000482 disable_encryption(false),
483 disable_sctp_data_channels(false) {
484 }
wu@webrtc.org97077a32013-10-25 21:18:33 +0000485 bool disable_encryption;
486 bool disable_sctp_data_channels;
487 };
488
489 virtual void SetOptions(const Options& options) = 0;
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000490
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000491 virtual rtc::scoped_refptr<PeerConnectionInterface>
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000492 CreatePeerConnection(
493 const PeerConnectionInterface::RTCConfiguration& configuration,
494 const MediaConstraintsInterface* constraints,
495 PortAllocatorFactoryInterface* allocator_factory,
496 DTLSIdentityServiceInterface* dtls_identity_service,
497 PeerConnectionObserver* observer) = 0;
498
499 // TODO(mallinath) : Remove below versions after clients are updated
500 // to above method.
501 // In latest W3C WebRTC draft, PC constructor will take RTCConfiguration,
502 // and not IceServers. RTCConfiguration is made up of ice servers and
503 // ice transport type.
504 // http://dev.w3.org/2011/webrtc/editor/webrtc.html
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000505 inline rtc::scoped_refptr<PeerConnectionInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000506 CreatePeerConnection(
507 const PeerConnectionInterface::IceServers& configuration,
508 const MediaConstraintsInterface* constraints,
509 PortAllocatorFactoryInterface* allocator_factory,
510 DTLSIdentityServiceInterface* dtls_identity_service,
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000511 PeerConnectionObserver* observer) {
512 PeerConnectionInterface::RTCConfiguration rtc_config;
513 rtc_config.servers = configuration;
514 return CreatePeerConnection(rtc_config, constraints, allocator_factory,
515 dtls_identity_service, observer);
516 }
517
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000518 virtual rtc::scoped_refptr<MediaStreamInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000519 CreateLocalMediaStream(const std::string& label) = 0;
520
521 // Creates a AudioSourceInterface.
522 // |constraints| decides audio processing settings but can be NULL.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000523 virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000524 const MediaConstraintsInterface* constraints) = 0;
525
526 // Creates a VideoSourceInterface. The new source take ownership of
527 // |capturer|. |constraints| decides video resolution and frame rate but can
528 // be NULL.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000529 virtual rtc::scoped_refptr<VideoSourceInterface> CreateVideoSource(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000530 cricket::VideoCapturer* capturer,
531 const MediaConstraintsInterface* constraints) = 0;
532
533 // Creates a new local VideoTrack. The same |source| can be used in several
534 // tracks.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000535 virtual rtc::scoped_refptr<VideoTrackInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000536 CreateVideoTrack(const std::string& label,
537 VideoSourceInterface* source) = 0;
538
539 // Creates an new AudioTrack. At the moment |source| can be NULL.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000540 virtual rtc::scoped_refptr<AudioTrackInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000541 CreateAudioTrack(const std::string& label,
542 AudioSourceInterface* source) = 0;
543
wu@webrtc.orga9890802013-12-13 00:21:03 +0000544 // Starts AEC dump using existing file. Takes ownership of |file| and passes
545 // it on to VoiceEngine (via other objects) immediately, which will take
wu@webrtc.orga8910d22014-01-23 22:12:45 +0000546 // the ownerhip. If the operation fails, the file will be closed.
wu@webrtc.orga9890802013-12-13 00:21:03 +0000547 // TODO(grunell): Remove when Chromium has started to use AEC in each source.
wu@webrtc.orga8910d22014-01-23 22:12:45 +0000548 // http://crbug.com/264611.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000549 virtual bool StartAecDump(rtc::PlatformFile file) = 0;
wu@webrtc.orga9890802013-12-13 00:21:03 +0000550
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000551 protected:
552 // Dtor and ctor protected as objects shouldn't be created or deleted via
553 // this interface.
554 PeerConnectionFactoryInterface() {}
555 ~PeerConnectionFactoryInterface() {} // NOLINT
556};
557
558// Create a new instance of PeerConnectionFactoryInterface.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000559rtc::scoped_refptr<PeerConnectionFactoryInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000560CreatePeerConnectionFactory();
561
562// Create a new instance of PeerConnectionFactoryInterface.
563// Ownership of |factory|, |default_adm|, and optionally |encoder_factory| and
564// |decoder_factory| transferred to the returned factory.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000565rtc::scoped_refptr<PeerConnectionFactoryInterface>
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000566CreatePeerConnectionFactory(
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000567 rtc::Thread* worker_thread,
568 rtc::Thread* signaling_thread,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000569 AudioDeviceModule* default_adm,
570 cricket::WebRtcVideoEncoderFactory* encoder_factory,
571 cricket::WebRtcVideoDecoderFactory* decoder_factory);
572
573} // namespace webrtc
574
575#endif // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_