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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
11// This file contains the PeerConnection interface as defined in
Steve Antonab6ea6b2018-02-26 14:23:09 -080012// https://w3c.github.io/webrtc-pc/#peer-to-peer-connections
henrike@webrtc.org28e20752013-07-10 00:45:36 +000013//
deadbeefb10f32f2017-02-08 01:38:21 -080014// The PeerConnectionFactory class provides factory methods to create
15// PeerConnection, MediaStream and MediaStreamTrack objects.
16//
17// The following steps are needed to setup a typical call using WebRTC:
18//
henrike@webrtc.org28e20752013-07-10 00:45:36 +000019// 1. Create a PeerConnectionFactoryInterface. Check constructors for more
20// information about input parameters.
deadbeefb10f32f2017-02-08 01:38:21 -080021//
22// 2. Create a PeerConnection object. Provide a configuration struct which
23// points to STUN and/or TURN servers used to generate ICE candidates, and
24// provide an object that implements the PeerConnectionObserver interface,
25// which is used to receive callbacks from the PeerConnection.
26//
27// 3. Create local MediaStreamTracks using the PeerConnectionFactory and add
28// them to PeerConnection by calling AddTrack (or legacy method, AddStream).
29//
30// 4. Create an offer, call SetLocalDescription with it, serialize it, and send
31// it to the remote peer
32//
33// 5. Once an ICE candidate has been gathered, the PeerConnection will call the
henrike@webrtc.org28e20752013-07-10 00:45:36 +000034// observer function OnIceCandidate. The candidates must also be serialized and
35// sent to the remote peer.
deadbeefb10f32f2017-02-08 01:38:21 -080036//
henrike@webrtc.org28e20752013-07-10 00:45:36 +000037// 6. Once an answer is received from the remote peer, call
deadbeefb10f32f2017-02-08 01:38:21 -080038// SetRemoteDescription with the remote answer.
39//
henrike@webrtc.org28e20752013-07-10 00:45:36 +000040// 7. Once a remote candidate is received from the remote peer, provide it to
deadbeefb10f32f2017-02-08 01:38:21 -080041// the PeerConnection by calling AddIceCandidate.
42//
43// The receiver of a call (assuming the application is "call"-based) can decide
44// to accept or reject the call; this decision will be taken by the application,
45// not the PeerConnection.
46//
47// If the application decides to accept the call, it should:
48//
henrike@webrtc.org28e20752013-07-10 00:45:36 +000049// 1. Create PeerConnectionFactoryInterface if it doesn't exist.
deadbeefb10f32f2017-02-08 01:38:21 -080050//
henrike@webrtc.org28e20752013-07-10 00:45:36 +000051// 2. Create a new PeerConnection.
deadbeefb10f32f2017-02-08 01:38:21 -080052//
henrike@webrtc.org28e20752013-07-10 00:45:36 +000053// 3. Provide the remote offer to the new PeerConnection object by calling
deadbeefb10f32f2017-02-08 01:38:21 -080054// SetRemoteDescription.
55//
henrike@webrtc.org28e20752013-07-10 00:45:36 +000056// 4. Generate an answer to the remote offer by calling CreateAnswer and send it
57// back to the remote peer.
deadbeefb10f32f2017-02-08 01:38:21 -080058//
henrike@webrtc.org28e20752013-07-10 00:45:36 +000059// 5. Provide the local answer to the new PeerConnection by calling
deadbeefb10f32f2017-02-08 01:38:21 -080060// SetLocalDescription with the answer.
61//
62// 6. Provide the remote ICE candidates by calling AddIceCandidate.
63//
64// 7. Once a candidate has been gathered, the PeerConnection will call the
65// observer function OnIceCandidate. Send these candidates to the remote peer.
henrike@webrtc.org28e20752013-07-10 00:45:36 +000066
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020067#ifndef API_PEERCONNECTIONINTERFACE_H_
68#define API_PEERCONNECTIONINTERFACE_H_
henrike@webrtc.org28e20752013-07-10 00:45:36 +000069
kwibergd1fe2812016-04-27 06:47:29 -070070#include <memory>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000071#include <string>
72#include <vector>
73
Zach Steine20867f2018-08-02 13:20:15 -070074#include "api/asyncresolverfactory.h"
Niels Möllerd377f042018-02-13 15:03:43 +010075#include "api/audio/audio_mixer.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020076#include "api/audio_codecs/audio_decoder_factory.h"
77#include "api/audio_codecs/audio_encoder_factory.h"
Niels Möllera6fe2612018-01-19 11:28:54 +010078#include "api/audio_options.h"
Niels Möller8366e172018-02-14 12:20:13 +010079#include "api/call/callfactoryinterface.h"
Benjamin Wrighta54daf12018-10-11 15:33:17 -070080#include "api/crypto/cryptooptions.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020081#include "api/datachannelinterface.h"
Ying Wang0dd1b0a2018-02-20 12:50:27 +010082#include "api/fec_controller.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020083#include "api/jsep.h"
Piotr (Peter) Slatalae0c2e972018-10-08 09:43:21 -070084#include "api/media_transport_interface.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020085#include "api/mediastreaminterface.h"
86#include "api/rtcerror.h"
Elad Alon99c3fe52017-10-13 16:29:40 +020087#include "api/rtceventlogoutput.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020088#include "api/rtpreceiverinterface.h"
89#include "api/rtpsenderinterface.h"
Steve Anton9158ef62017-11-27 13:01:52 -080090#include "api/rtptransceiverinterface.h"
Henrik Boström31638672017-11-23 17:48:32 +010091#include "api/setremotedescriptionobserverinterface.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020092#include "api/stats/rtcstatscollectorcallback.h"
93#include "api/statstypes.h"
Niels Möller0c4f7be2018-05-07 14:01:37 +020094#include "api/transport/bitrate_settings.h"
Sebastian Janssondfce03a2018-05-18 18:05:10 +020095#include "api/transport/network_control.h"
Jonas Orelandbdcee282017-10-10 14:01:40 +020096#include "api/turncustomizer.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020097#include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
Niels Möller6daa2782018-01-23 10:37:42 +010098#include "media/base/mediaconfig.h"
Niels Möller8366e172018-02-14 12:20:13 +010099// TODO(bugs.webrtc.org/6353): cricket::VideoCapturer is deprecated and should
100// be deleted from the PeerConnection api.
101#include "media/base/videocapturer.h" // nogncheck
102// TODO(bugs.webrtc.org/7447): We plan to provide a way to let applications
103// inject a PacketSocketFactory and/or NetworkManager, and not expose
104// PortAllocator in the PeerConnection api.
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +0200105#include "media/base/mediaengine.h" // nogncheck
Niels Möller8366e172018-02-14 12:20:13 +0100106#include "p2p/base/portallocator.h" // nogncheck
107// TODO(nisse): The interface for bitrate allocation strategy belongs in api/.
108#include "rtc_base/bitrateallocationstrategy.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200109#include "rtc_base/network.h"
Niels Möller8366e172018-02-14 12:20:13 +0100110#include "rtc_base/platform_file.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200111#include "rtc_base/rtccertificate.h"
112#include "rtc_base/rtccertificategenerator.h"
113#include "rtc_base/socketaddress.h"
Benjamin Wrightd6f86e82018-05-08 13:12:25 -0700114#include "rtc_base/sslcertificate.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200115#include "rtc_base/sslstreamadapter.h"
Mirko Bonadei276827c2018-10-16 14:13:50 +0200116#include "rtc_base/system/rtc_export.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000117
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000118namespace rtc {
jiayl@webrtc.org61e00b02015-03-04 22:17:38 +0000119class SSLIdentity;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000120class Thread;
Yves Gerey665174f2018-06-19 15:03:05 +0200121} // namespace rtc
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000122
123namespace cricket {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000124class WebRtcVideoDecoderFactory;
125class WebRtcVideoEncoderFactory;
Yves Gerey665174f2018-06-19 15:03:05 +0200126} // namespace cricket
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000127
128namespace webrtc {
129class AudioDeviceModule;
gyzhou95aa9642016-12-13 14:06:26 -0800130class AudioMixer;
Niels Möller8366e172018-02-14 12:20:13 +0100131class AudioProcessing;
Harald Alvestrandad88c882018-11-28 16:47:46 +0100132class DtlsTransportInterface;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000133class MediaConstraintsInterface;
Magnus Jedvert58b03162017-09-15 19:02:47 +0200134class VideoDecoderFactory;
135class VideoEncoderFactory;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000136
137// MediaStream container interface.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000138class StreamCollectionInterface : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000139 public:
140 // TODO(ronghuawu): Update the function names to c++ style, e.g. find -> Find.
141 virtual size_t count() = 0;
142 virtual MediaStreamInterface* at(size_t index) = 0;
143 virtual MediaStreamInterface* find(const std::string& label) = 0;
Yves Gerey665174f2018-06-19 15:03:05 +0200144 virtual MediaStreamTrackInterface* FindAudioTrack(const std::string& id) = 0;
145 virtual MediaStreamTrackInterface* FindVideoTrack(const std::string& id) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000146
147 protected:
148 // Dtor protected as objects shouldn't be deleted via this interface.
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +0200149 ~StreamCollectionInterface() override = default;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000150};
151
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000152class StatsObserver : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000153 public:
nissee8abe3e2017-01-18 05:00:34 -0800154 virtual void OnComplete(const StatsReports& reports) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000155
156 protected:
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +0200157 ~StatsObserver() override = default;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000158};
159
Steve Anton3acffc32018-04-12 17:21:03 -0700160enum class SdpSemantics { kPlanB, kUnifiedPlan };
Steve Anton79e79602017-11-20 10:25:56 -0800161
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000162class PeerConnectionInterface : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000163 public:
Jonas Olsson635474e2018-10-18 15:58:17 +0200164 // See https://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000165 enum SignalingState {
166 kStable,
167 kHaveLocalOffer,
168 kHaveLocalPrAnswer,
169 kHaveRemoteOffer,
170 kHaveRemotePrAnswer,
171 kClosed,
172 };
173
Jonas Olsson635474e2018-10-18 15:58:17 +0200174 // See https://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000175 enum IceGatheringState {
176 kIceGatheringNew,
177 kIceGatheringGathering,
178 kIceGatheringComplete
179 };
180
Jonas Olsson635474e2018-10-18 15:58:17 +0200181 // See https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate
182 enum class PeerConnectionState {
183 kNew,
184 kConnecting,
185 kConnected,
186 kDisconnected,
187 kFailed,
188 kClosed,
189 };
190
191 // See https://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000192 enum IceConnectionState {
193 kIceConnectionNew,
194 kIceConnectionChecking,
195 kIceConnectionConnected,
196 kIceConnectionCompleted,
197 kIceConnectionFailed,
198 kIceConnectionDisconnected,
199 kIceConnectionClosed,
Guo-wei Shieh3d564c12015-08-19 16:51:15 -0700200 kIceConnectionMax,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000201 };
202
hnsl04833622017-01-09 08:35:45 -0800203 // TLS certificate policy.
204 enum TlsCertPolicy {
205 // For TLS based protocols, ensure the connection is secure by not
206 // circumventing certificate validation.
207 kTlsCertPolicySecure,
208 // For TLS based protocols, disregard security completely by skipping
209 // certificate validation. This is insecure and should never be used unless
210 // security is irrelevant in that particular context.
211 kTlsCertPolicyInsecureNoCheck,
212 };
213
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000214 struct IceServer {
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +0200215 IceServer();
216 IceServer(const IceServer&);
217 ~IceServer();
218
Joachim Bauch7c4e7452015-05-28 23:06:30 +0200219 // TODO(jbauch): Remove uri when all code using it has switched to urls.
Emad Omaradab1d2d2017-06-16 15:43:11 -0700220 // List of URIs associated with this server. Valid formats are described
221 // in RFC7064 and RFC7065, and more may be added in the future. The "host"
222 // part of the URI may contain either an IP address or a hostname.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000223 std::string uri;
Joachim Bauch7c4e7452015-05-28 23:06:30 +0200224 std::vector<std::string> urls;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000225 std::string username;
226 std::string password;
hnsl04833622017-01-09 08:35:45 -0800227 TlsCertPolicy tls_cert_policy = kTlsCertPolicySecure;
Emad Omaradab1d2d2017-06-16 15:43:11 -0700228 // If the URIs in |urls| only contain IP addresses, this field can be used
229 // to indicate the hostname, which may be necessary for TLS (using the SNI
230 // extension). If |urls| itself contains the hostname, this isn't
231 // necessary.
232 std::string hostname;
Diogo Real1dca9d52017-08-29 12:18:32 -0700233 // List of protocols to be used in the TLS ALPN extension.
234 std::vector<std::string> tls_alpn_protocols;
Diogo Real7bd1f1b2017-09-08 12:50:41 -0700235 // List of elliptic curves to be used in the TLS elliptic curves extension.
236 std::vector<std::string> tls_elliptic_curves;
hnsl04833622017-01-09 08:35:45 -0800237
deadbeefd1a38b52016-12-10 13:15:33 -0800238 bool operator==(const IceServer& o) const {
239 return uri == o.uri && urls == o.urls && username == o.username &&
Emad Omaradab1d2d2017-06-16 15:43:11 -0700240 password == o.password && tls_cert_policy == o.tls_cert_policy &&
Diogo Real1dca9d52017-08-29 12:18:32 -0700241 hostname == o.hostname &&
Diogo Real7bd1f1b2017-09-08 12:50:41 -0700242 tls_alpn_protocols == o.tls_alpn_protocols &&
Sergey Silkin9c147dd2018-09-12 10:45:38 +0000243 tls_elliptic_curves == o.tls_elliptic_curves;
deadbeefd1a38b52016-12-10 13:15:33 -0800244 }
245 bool operator!=(const IceServer& o) const { return !(*this == o); }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000246 };
247 typedef std::vector<IceServer> IceServers;
248
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000249 enum IceTransportsType {
pthatcher@webrtc.orgfd630a52015-01-14 23:19:06 +0000250 // TODO(pthatcher): Rename these kTransporTypeXXX, but update
251 // Chromium at the same time.
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000252 kNone,
253 kRelay,
254 kNoHost,
255 kAll
256 };
257
Steve Antonab6ea6b2018-02-26 14:23:09 -0800258 // https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
pthatcher@webrtc.orgfd630a52015-01-14 23:19:06 +0000259 enum BundlePolicy {
260 kBundlePolicyBalanced,
261 kBundlePolicyMaxBundle,
262 kBundlePolicyMaxCompat
263 };
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000264
Steve Antonab6ea6b2018-02-26 14:23:09 -0800265 // https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
Peter Thatcheraf55ccc2015-05-21 07:48:41 -0700266 enum RtcpMuxPolicy {
267 kRtcpMuxPolicyNegotiate,
268 kRtcpMuxPolicyRequire,
269 };
270
Jiayang Liucac1b382015-04-30 12:35:24 -0700271 enum TcpCandidatePolicy {
272 kTcpCandidatePolicyEnabled,
273 kTcpCandidatePolicyDisabled
274 };
275
honghaiz60347052016-05-31 18:29:12 -0700276 enum CandidateNetworkPolicy {
277 kCandidateNetworkPolicyAll,
278 kCandidateNetworkPolicyLowCost
279 };
280
Yves Gerey665174f2018-06-19 15:03:05 +0200281 enum ContinualGatheringPolicy { GATHER_ONCE, GATHER_CONTINUALLY };
honghaiz1f429e32015-09-28 07:57:34 -0700282
Honghai Zhangf7ddc062016-09-01 15:34:01 -0700283 enum class RTCConfigurationType {
284 // A configuration that is safer to use, despite not having the best
285 // performance. Currently this is the default configuration.
286 kSafe,
287 // An aggressive configuration that has better performance, although it
288 // may be riskier and may need extra support in the application.
289 kAggressive
290 };
291
Henrik Boström87713d02015-08-25 09:53:21 +0200292 // TODO(hbos): Change into class with private data and public getters.
nissec36b31b2016-04-11 23:25:29 -0700293 // TODO(nisse): In particular, accessing fields directly from an
294 // application is brittle, since the organization mirrors the
295 // organization of the implementation, which isn't stable. So we
296 // need getters and setters at least for fields which applications
297 // are interested in.
Mirko Bonadeiac194142018-10-22 17:08:37 +0200298 struct RTC_EXPORT RTCConfiguration {
Niels Möller71bdda02016-03-31 12:59:59 +0200299 // This struct is subject to reorganization, both for naming
300 // consistency, and to group settings to match where they are used
301 // in the implementation. To do that, we need getter and setter
302 // methods for all settings which are of interest to applications,
303 // Chrome in particular.
304
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +0200305 RTCConfiguration();
306 RTCConfiguration(const RTCConfiguration&);
307 explicit RTCConfiguration(RTCConfigurationType type);
308 ~RTCConfiguration();
Honghai Zhangbfd398c2016-08-30 22:07:42 -0700309
deadbeef293e9262017-01-11 12:28:30 -0800310 bool operator==(const RTCConfiguration& o) const;
311 bool operator!=(const RTCConfiguration& o) const;
312
Niels Möller6539f692018-01-18 08:58:50 +0100313 bool dscp() const { return media_config.enable_dscp; }
nissec36b31b2016-04-11 23:25:29 -0700314 void set_dscp(bool enable) { media_config.enable_dscp = enable; }
Niels Möller71bdda02016-03-31 12:59:59 +0200315
Niels Möller6539f692018-01-18 08:58:50 +0100316 bool cpu_adaptation() const {
Niels Möller1d7ecd22018-01-18 15:25:12 +0100317 return media_config.video.enable_cpu_adaptation;
nissec36b31b2016-04-11 23:25:29 -0700318 }
Niels Möller71bdda02016-03-31 12:59:59 +0200319 void set_cpu_adaptation(bool enable) {
Niels Möller1d7ecd22018-01-18 15:25:12 +0100320 media_config.video.enable_cpu_adaptation = enable;
Niels Möller71bdda02016-03-31 12:59:59 +0200321 }
322
Niels Möller6539f692018-01-18 08:58:50 +0100323 bool suspend_below_min_bitrate() const {
nissec36b31b2016-04-11 23:25:29 -0700324 return media_config.video.suspend_below_min_bitrate;
325 }
Niels Möller71bdda02016-03-31 12:59:59 +0200326 void set_suspend_below_min_bitrate(bool enable) {
nissec36b31b2016-04-11 23:25:29 -0700327 media_config.video.suspend_below_min_bitrate = enable;
Niels Möller71bdda02016-03-31 12:59:59 +0200328 }
329
Niels Möller6539f692018-01-18 08:58:50 +0100330 bool prerenderer_smoothing() const {
Niels Möller1d7ecd22018-01-18 15:25:12 +0100331 return media_config.video.enable_prerenderer_smoothing;
nissec36b31b2016-04-11 23:25:29 -0700332 }
Niels Möller71bdda02016-03-31 12:59:59 +0200333 void set_prerenderer_smoothing(bool enable) {
Niels Möller1d7ecd22018-01-18 15:25:12 +0100334 media_config.video.enable_prerenderer_smoothing = enable;
Niels Möller71bdda02016-03-31 12:59:59 +0200335 }
336
Niels Möller6539f692018-01-18 08:58:50 +0100337 bool experiment_cpu_load_estimator() const {
338 return media_config.video.experiment_cpu_load_estimator;
339 }
340 void set_experiment_cpu_load_estimator(bool enable) {
341 media_config.video.experiment_cpu_load_estimator = enable;
342 }
Ilya Nikolaevskiy97b4ee52018-05-28 10:24:22 +0200343
Jiawei Ou55718122018-11-09 13:17:39 -0800344 int audio_rtcp_report_interval_ms() const {
345 return media_config.audio.rtcp_report_interval_ms;
346 }
347 void set_audio_rtcp_report_interval_ms(int audio_rtcp_report_interval_ms) {
348 media_config.audio.rtcp_report_interval_ms =
349 audio_rtcp_report_interval_ms;
350 }
351
352 int video_rtcp_report_interval_ms() const {
353 return media_config.video.rtcp_report_interval_ms;
354 }
355 void set_video_rtcp_report_interval_ms(int video_rtcp_report_interval_ms) {
356 media_config.video.rtcp_report_interval_ms =
357 video_rtcp_report_interval_ms;
358 }
359
honghaiz4edc39c2015-09-01 09:53:56 -0700360 static const int kUndefined = -1;
361 // Default maximum number of packets in the audio jitter buffer.
362 static const int kAudioJitterBufferMaxPackets = 50;
Honghai Zhangaecd9822016-09-02 16:58:17 -0700363 // ICE connection receiving timeout for aggressive configuration.
364 static const int kAggressiveIceConnectionReceivingTimeout = 1000;
deadbeefb10f32f2017-02-08 01:38:21 -0800365
366 ////////////////////////////////////////////////////////////////////////
367 // The below few fields mirror the standard RTCConfiguration dictionary:
Steve Antonab6ea6b2018-02-26 14:23:09 -0800368 // https://w3c.github.io/webrtc-pc/#rtcconfiguration-dictionary
deadbeefb10f32f2017-02-08 01:38:21 -0800369 ////////////////////////////////////////////////////////////////////////
370
pthatcher@webrtc.orgfd630a52015-01-14 23:19:06 +0000371 // TODO(pthatcher): Rename this ice_servers, but update Chromium
372 // at the same time.
373 IceServers servers;
deadbeefb10f32f2017-02-08 01:38:21 -0800374 // TODO(pthatcher): Rename this ice_transport_type, but update
375 // Chromium at the same time.
376 IceTransportsType type = kAll;
Taylor Brandstettera1c30352016-05-13 08:15:11 -0700377 BundlePolicy bundle_policy = kBundlePolicyBalanced;
zhihuang4dfb8ce2016-11-23 10:30:12 -0800378 RtcpMuxPolicy rtcp_mux_policy = kRtcpMuxPolicyRequire;
deadbeefb10f32f2017-02-08 01:38:21 -0800379 std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
380 int ice_candidate_pool_size = 0;
381
382 //////////////////////////////////////////////////////////////////////////
383 // The below fields correspond to constraints from the deprecated
384 // constraints interface for constructing a PeerConnection.
385 //
Danil Chapovalov0bc58cf2018-06-21 13:32:56 +0200386 // absl::optional fields can be "missing", in which case the implementation
deadbeefb10f32f2017-02-08 01:38:21 -0800387 // default will be used.
388 //////////////////////////////////////////////////////////////////////////
389
390 // If set to true, don't gather IPv6 ICE candidates.
391 // TODO(deadbeef): Remove this? IPv6 support has long stopped being
392 // experimental
393 bool disable_ipv6 = false;
394
zhihuangb09b3f92017-03-07 14:40:51 -0800395 // If set to true, don't gather IPv6 ICE candidates on Wi-Fi.
396 // Only intended to be used on specific devices. Certain phones disable IPv6
397 // when the screen is turned off and it would be better to just disable the
398 // IPv6 ICE candidates on Wi-Fi in those cases.
399 bool disable_ipv6_on_wifi = false;
400
deadbeefd21eab32017-07-26 16:50:11 -0700401 // By default, the PeerConnection will use a limited number of IPv6 network
402 // interfaces, in order to avoid too many ICE candidate pairs being created
403 // and delaying ICE completion.
404 //
405 // Can be set to INT_MAX to effectively disable the limit.
406 int max_ipv6_networks = cricket::kDefaultMaxIPv6Networks;
407
Daniel Lazarenko2870b0a2018-01-25 10:30:22 +0100408 // Exclude link-local network interfaces
409 // from considertaion for gathering ICE candidates.
410 bool disable_link_local_networks = false;
411
deadbeefb10f32f2017-02-08 01:38:21 -0800412 // If set to true, use RTP data channels instead of SCTP.
413 // TODO(deadbeef): Remove this. We no longer commit to supporting RTP data
414 // channels, though some applications are still working on moving off of
415 // them.
416 bool enable_rtp_data_channel = false;
417
418 // Minimum bitrate at which screencast video tracks will be encoded at.
419 // This means adding padding bits up to this bitrate, which can help
420 // when switching from a static scene to one with motion.
Danil Chapovalov0bc58cf2018-06-21 13:32:56 +0200421 absl::optional<int> screencast_min_bitrate;
deadbeefb10f32f2017-02-08 01:38:21 -0800422
423 // Use new combined audio/video bandwidth estimation?
Danil Chapovalov0bc58cf2018-06-21 13:32:56 +0200424 absl::optional<bool> combined_audio_video_bwe;
deadbeefb10f32f2017-02-08 01:38:21 -0800425
Benjamin Wright8c27cca2018-10-25 10:16:44 -0700426 // TODO(bugs.webrtc.org/9891) - Move to crypto_options
deadbeefb10f32f2017-02-08 01:38:21 -0800427 // Can be used to disable DTLS-SRTP. This should never be done, but can be
428 // useful for testing purposes, for example in setting up a loopback call
429 // with a single PeerConnection.
Danil Chapovalov0bc58cf2018-06-21 13:32:56 +0200430 absl::optional<bool> enable_dtls_srtp;
deadbeefb10f32f2017-02-08 01:38:21 -0800431
432 /////////////////////////////////////////////////
433 // The below fields are not part of the standard.
434 /////////////////////////////////////////////////
435
436 // Can be used to disable TCP candidate generation.
Taylor Brandstettera1c30352016-05-13 08:15:11 -0700437 TcpCandidatePolicy tcp_candidate_policy = kTcpCandidatePolicyEnabled;
deadbeefb10f32f2017-02-08 01:38:21 -0800438
439 // Can be used to avoid gathering candidates for a "higher cost" network,
440 // if a lower cost one exists. For example, if both Wi-Fi and cellular
441 // interfaces are available, this could be used to avoid using the cellular
442 // interface.
honghaiz60347052016-05-31 18:29:12 -0700443 CandidateNetworkPolicy candidate_network_policy =
444 kCandidateNetworkPolicyAll;
deadbeefb10f32f2017-02-08 01:38:21 -0800445
446 // The maximum number of packets that can be stored in the NetEq audio
447 // jitter buffer. Can be reduced to lower tolerated audio latency.
Taylor Brandstettera1c30352016-05-13 08:15:11 -0700448 int audio_jitter_buffer_max_packets = kAudioJitterBufferMaxPackets;
deadbeefb10f32f2017-02-08 01:38:21 -0800449
450 // Whether to use the NetEq "fast mode" which will accelerate audio quicker
451 // if it falls behind.
Taylor Brandstettera1c30352016-05-13 08:15:11 -0700452 bool audio_jitter_buffer_fast_accelerate = false;
deadbeefb10f32f2017-02-08 01:38:21 -0800453
Jakob Ivarsson10403ae2018-11-27 15:45:20 +0100454 // The minimum delay in milliseconds for the audio jitter buffer.
455 int audio_jitter_buffer_min_delay_ms = 0;
456
deadbeefb10f32f2017-02-08 01:38:21 -0800457 // Timeout in milliseconds before an ICE candidate pair is considered to be
458 // "not receiving", after which a lower priority candidate pair may be
459 // selected.
460 int ice_connection_receiving_timeout = kUndefined;
461
462 // Interval in milliseconds at which an ICE "backup" candidate pair will be
463 // pinged. This is a candidate pair which is not actively in use, but may
464 // be switched to if the active candidate pair becomes unusable.
465 //
466 // This is relevant mainly to Wi-Fi/cell handoff; the application may not
467 // want this backup cellular candidate pair pinged frequently, since it
468 // consumes data/battery.
469 int ice_backup_candidate_pair_ping_interval = kUndefined;
470
471 // Can be used to enable continual gathering, which means new candidates
472 // will be gathered as network interfaces change. Note that if continual
473 // gathering is used, the candidate removal API should also be used, to
474 // avoid an ever-growing list of candidates.
Taylor Brandstettera1c30352016-05-13 08:15:11 -0700475 ContinualGatheringPolicy continual_gathering_policy = GATHER_ONCE;
deadbeefb10f32f2017-02-08 01:38:21 -0800476
477 // If set to true, candidate pairs will be pinged in order of most likely
478 // to work (which means using a TURN server, generally), rather than in
479 // standard priority order.
Taylor Brandstettera1c30352016-05-13 08:15:11 -0700480 bool prioritize_most_likely_ice_candidate_pairs = false;
deadbeefb10f32f2017-02-08 01:38:21 -0800481
Niels Möller6daa2782018-01-23 10:37:42 +0100482 // Implementation defined settings. A public member only for the benefit of
483 // the implementation. Applications must not access it directly, and should
484 // instead use provided accessor methods, e.g., set_cpu_adaptation.
nissec36b31b2016-04-11 23:25:29 -0700485 struct cricket::MediaConfig media_config;
deadbeefb10f32f2017-02-08 01:38:21 -0800486
deadbeefb10f32f2017-02-08 01:38:21 -0800487 // If set to true, only one preferred TURN allocation will be used per
488 // network interface. UDP is preferred over TCP and IPv6 over IPv4. This
489 // can be used to cut down on the number of candidate pairings.
Honghai Zhangb9e7b4a2016-06-30 20:52:02 -0700490 bool prune_turn_ports = false;
deadbeefb10f32f2017-02-08 01:38:21 -0800491
Taylor Brandstettere9851112016-07-01 11:11:13 -0700492 // If set to true, this means the ICE transport should presume TURN-to-TURN
493 // candidate pairs will succeed, even before a binding response is received.
deadbeefb10f32f2017-02-08 01:38:21 -0800494 // This can be used to optimize the initial connection time, since the DTLS
495 // handshake can begin immediately.
Taylor Brandstettere9851112016-07-01 11:11:13 -0700496 bool presume_writable_when_fully_relayed = false;
deadbeefb10f32f2017-02-08 01:38:21 -0800497
Honghai Zhang4cedf2b2016-08-31 08:18:11 -0700498 // If true, "renomination" will be added to the ice options in the transport
499 // description.
deadbeefb10f32f2017-02-08 01:38:21 -0800500 // See: https://tools.ietf.org/html/draft-thatcher-ice-renomination-00
Honghai Zhang4cedf2b2016-08-31 08:18:11 -0700501 bool enable_ice_renomination = false;
deadbeefb10f32f2017-02-08 01:38:21 -0800502
503 // If true, the ICE role is re-determined when the PeerConnection sets a
504 // local transport description that indicates an ICE restart.
505 //
506 // This is standard RFC5245 ICE behavior, but causes unnecessary role
507 // thrashing, so an application may wish to avoid it. This role
508 // re-determining was removed in ICEbis (ICE v2).
Honghai Zhangbfd398c2016-08-30 22:07:42 -0700509 bool redetermine_role_on_ice_restart = true;
deadbeefb10f32f2017-02-08 01:38:21 -0800510
Qingsi Wange6826d22018-03-08 14:55:14 -0800511 // The following fields define intervals in milliseconds at which ICE
512 // connectivity checks are sent.
513 //
514 // We consider ICE is "strongly connected" for an agent when there is at
515 // least one candidate pair that currently succeeds in connectivity check
516 // from its direction i.e. sending a STUN ping and receives a STUN ping
517 // response, AND all candidate pairs have sent a minimum number of pings for
518 // connectivity (this number is implementation-specific). Otherwise, ICE is
519 // considered in "weak connectivity".
520 //
521 // Note that the above notion of strong and weak connectivity is not defined
522 // in RFC 5245, and they apply to our current ICE implementation only.
523 //
524 // 1) ice_check_interval_strong_connectivity defines the interval applied to
525 // ALL candidate pairs when ICE is strongly connected, and it overrides the
526 // default value of this interval in the ICE implementation;
527 // 2) ice_check_interval_weak_connectivity defines the counterpart for ALL
528 // pairs when ICE is weakly connected, and it overrides the default value of
529 // this interval in the ICE implementation;
530 // 3) ice_check_min_interval defines the minimal interval (equivalently the
531 // maximum rate) that overrides the above two intervals when either of them
532 // is less.
Danil Chapovalov0bc58cf2018-06-21 13:32:56 +0200533 absl::optional<int> ice_check_interval_strong_connectivity;
534 absl::optional<int> ice_check_interval_weak_connectivity;
535 absl::optional<int> ice_check_min_interval;
deadbeefb10f32f2017-02-08 01:38:21 -0800536
Qingsi Wang22e623a2018-03-13 10:53:57 -0700537 // The min time period for which a candidate pair must wait for response to
538 // connectivity checks before it becomes unwritable. This parameter
539 // overrides the default value in the ICE implementation if set.
Danil Chapovalov0bc58cf2018-06-21 13:32:56 +0200540 absl::optional<int> ice_unwritable_timeout;
Qingsi Wang22e623a2018-03-13 10:53:57 -0700541
542 // The min number of connectivity checks that a candidate pair must sent
543 // without receiving response before it becomes unwritable. This parameter
544 // overrides the default value in the ICE implementation if set.
Danil Chapovalov0bc58cf2018-06-21 13:32:56 +0200545 absl::optional<int> ice_unwritable_min_checks;
Qingsi Wang22e623a2018-03-13 10:53:57 -0700546
Jiawei Ou9d4fd5552018-12-06 23:30:17 -0800547 // The min time period for which a candidate pair must wait for response to
548 // connectivity checks it becomes inactive. This parameter overrides the
549 // default value in the ICE implementation if set.
550 absl::optional<int> ice_inactive_timeout;
551
Qingsi Wangdb53f8e2018-02-20 14:45:49 -0800552 // The interval in milliseconds at which STUN candidates will resend STUN
553 // binding requests to keep NAT bindings open.
Danil Chapovalov0bc58cf2018-06-21 13:32:56 +0200554 absl::optional<int> stun_candidate_keepalive_interval;
Qingsi Wangdb53f8e2018-02-20 14:45:49 -0800555
Steve Anton300bf8e2017-07-14 10:13:10 -0700556 // ICE Periodic Regathering
557 // If set, WebRTC will periodically create and propose candidates without
558 // starting a new ICE generation. The regathering happens continuously with
559 // interval specified in milliseconds by the uniform distribution [a, b].
Danil Chapovalov0bc58cf2018-06-21 13:32:56 +0200560 absl::optional<rtc::IntervalRange> ice_regather_interval_range;
Steve Anton300bf8e2017-07-14 10:13:10 -0700561
Jonas Orelandbdcee282017-10-10 14:01:40 +0200562 // Optional TurnCustomizer.
563 // With this class one can modify outgoing TURN messages.
564 // The object passed in must remain valid until PeerConnection::Close() is
565 // called.
566 webrtc::TurnCustomizer* turn_customizer = nullptr;
567
Qingsi Wang9a5c6f82018-02-01 10:38:40 -0800568 // Preferred network interface.
569 // A candidate pair on a preferred network has a higher precedence in ICE
570 // than one on an un-preferred network, regardless of priority or network
571 // cost.
Danil Chapovalov0bc58cf2018-06-21 13:32:56 +0200572 absl::optional<rtc::AdapterType> network_preference;
Qingsi Wang9a5c6f82018-02-01 10:38:40 -0800573
Steve Anton79e79602017-11-20 10:25:56 -0800574 // Configure the SDP semantics used by this PeerConnection. Note that the
575 // WebRTC 1.0 specification requires kUnifiedPlan semantics. The
576 // RtpTransceiver API is only available with kUnifiedPlan semantics.
577 //
578 // kPlanB will cause PeerConnection to create offers and answers with at
579 // most one audio and one video m= section with multiple RtpSenders and
580 // RtpReceivers specified as multiple a=ssrc lines within the section. This
Steve Antonab6ea6b2018-02-26 14:23:09 -0800581 // will also cause PeerConnection to ignore all but the first m= section of
582 // the same media type.
Steve Anton79e79602017-11-20 10:25:56 -0800583 //
584 // kUnifiedPlan will cause PeerConnection to create offers and answers with
585 // multiple m= sections where each m= section maps to one RtpSender and one
Steve Antonab6ea6b2018-02-26 14:23:09 -0800586 // RtpReceiver (an RtpTransceiver), either both audio or both video. This
587 // will also cause PeerConnection to ignore all but the first a=ssrc lines
588 // that form a Plan B stream.
Steve Anton79e79602017-11-20 10:25:56 -0800589 //
Steve Anton79e79602017-11-20 10:25:56 -0800590 // For users who wish to send multiple audio/video streams and need to stay
Steve Anton3acffc32018-04-12 17:21:03 -0700591 // interoperable with legacy WebRTC implementations or use legacy APIs,
592 // specify kPlanB.
Steve Anton79e79602017-11-20 10:25:56 -0800593 //
Steve Anton3acffc32018-04-12 17:21:03 -0700594 // For all other users, specify kUnifiedPlan.
595 SdpSemantics sdp_semantics = SdpSemantics::kPlanB;
Steve Anton79e79602017-11-20 10:25:56 -0800596
Benjamin Wright8c27cca2018-10-25 10:16:44 -0700597 // TODO(bugs.webrtc.org/9891) - Move to crypto_options or remove.
Zhi Huangb57e1692018-06-12 11:41:11 -0700598 // Actively reset the SRTP parameters whenever the DTLS transports
599 // underneath are reset for every offer/answer negotiation.
600 // This is only intended to be a workaround for crbug.com/835958
601 // WARNING: This would cause RTP/RTCP packets decryption failure if not used
602 // correctly. This flag will be deprecated soon. Do not rely on it.
603 bool active_reset_srtp_params = false;
604
Piotr (Peter) Slatalae0c2e972018-10-08 09:43:21 -0700605 // If MediaTransportFactory is provided in PeerConnectionFactory, this flag
606 // informs PeerConnection that it should use the MediaTransportInterface.
607 // It's invalid to set it to |true| if the MediaTransportFactory wasn't
608 // provided.
609 bool use_media_transport = false;
610
Bjorn Mellema9bbd862018-11-02 09:07:48 -0700611 // If MediaTransportFactory is provided in PeerConnectionFactory, this flag
612 // informs PeerConnection that it should use the MediaTransportInterface for
613 // data channels. It's invalid to set it to |true| if the
614 // MediaTransportFactory wasn't provided. Data channels over media
615 // transport are not compatible with RTP or SCTP data channels. Setting
616 // both |use_media_transport_for_data_channels| and
617 // |enable_rtp_data_channel| is invalid.
618 bool use_media_transport_for_data_channels = false;
619
Benjamin Wright8c27cca2018-10-25 10:16:44 -0700620 // Defines advanced optional cryptographic settings related to SRTP and
621 // frame encryption for native WebRTC. Setting this will overwrite any
622 // settings set in PeerConnectionFactory (which is deprecated).
623 absl::optional<CryptoOptions> crypto_options;
624
Johannes Kron89f874e2018-11-12 10:25:48 +0100625 // Configure if we should include the SDP attribute extmap-allow-mixed in
626 // our offer. Although we currently do support this, it's not included in
627 // our offer by default due to a previous bug that caused the SDP parser to
628 // abort parsing if this attribute was present. This is fixed in Chrome 71.
629 // TODO(webrtc:9985): Change default to true once sufficient time has
630 // passed.
631 bool offer_extmap_allow_mixed = false;
632
deadbeef293e9262017-01-11 12:28:30 -0800633 //
634 // Don't forget to update operator== if adding something.
635 //
buildbot@webrtc.org41451d42014-05-03 05:39:45 +0000636 };
637
deadbeefb10f32f2017-02-08 01:38:21 -0800638 // See: https://www.w3.org/TR/webrtc/#idl-def-rtcofferansweroptions
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000639 struct RTCOfferAnswerOptions {
640 static const int kUndefined = -1;
641 static const int kMaxOfferToReceiveMedia = 1;
642
643 // The default value for constraint offerToReceiveX:true.
644 static const int kOfferToReceiveMediaTrue = 1;
645
Steve Antonab6ea6b2018-02-26 14:23:09 -0800646 // These options are left as backwards compatibility for clients who need
647 // "Plan B" semantics. Clients who have switched to "Unified Plan" semantics
648 // should use the RtpTransceiver API (AddTransceiver) instead.
deadbeefb10f32f2017-02-08 01:38:21 -0800649 //
650 // offer_to_receive_X set to 1 will cause a media description to be
651 // generated in the offer, even if no tracks of that type have been added.
652 // Values greater than 1 are treated the same.
653 //
654 // If set to 0, the generated directional attribute will not include the
655 // "recv" direction (meaning it will be "sendonly" or "inactive".
Honghai Zhang4cedf2b2016-08-31 08:18:11 -0700656 int offer_to_receive_video = kUndefined;
657 int offer_to_receive_audio = kUndefined;
deadbeefb10f32f2017-02-08 01:38:21 -0800658
Honghai Zhang4cedf2b2016-08-31 08:18:11 -0700659 bool voice_activity_detection = true;
660 bool ice_restart = false;
deadbeefb10f32f2017-02-08 01:38:21 -0800661
662 // If true, will offer to BUNDLE audio/video/data together. Not to be
663 // confused with RTCP mux (multiplexing RTP and RTCP together).
Honghai Zhang4cedf2b2016-08-31 08:18:11 -0700664 bool use_rtp_mux = true;
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000665
Jonas Orelandfc1acd22018-08-24 10:58:37 +0200666 // This will apply to all video tracks with a Plan B SDP offer/answer.
667 int num_simulcast_layers = 1;
668
Honghai Zhang4cedf2b2016-08-31 08:18:11 -0700669 RTCOfferAnswerOptions() = default;
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000670
671 RTCOfferAnswerOptions(int offer_to_receive_video,
672 int offer_to_receive_audio,
673 bool voice_activity_detection,
674 bool ice_restart,
675 bool use_rtp_mux)
676 : offer_to_receive_video(offer_to_receive_video),
677 offer_to_receive_audio(offer_to_receive_audio),
678 voice_activity_detection(voice_activity_detection),
679 ice_restart(ice_restart),
680 use_rtp_mux(use_rtp_mux) {}
681 };
682
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000683 // Used by GetStats to decide which stats to include in the stats reports.
684 // |kStatsOutputLevelStandard| includes the standard stats for Javascript API;
685 // |kStatsOutputLevelDebug| includes both the standard stats and additional
686 // stats for debugging purposes.
687 enum StatsOutputLevel {
688 kStatsOutputLevelStandard,
689 kStatsOutputLevelDebug,
690 };
691
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000692 // Accessor methods to active local streams.
Steve Antonab6ea6b2018-02-26 14:23:09 -0800693 // This method is not supported with kUnifiedPlan semantics. Please use
694 // GetSenders() instead.
Yves Gerey665174f2018-06-19 15:03:05 +0200695 virtual rtc::scoped_refptr<StreamCollectionInterface> local_streams() = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000696
697 // Accessor methods to remote streams.
Steve Antonab6ea6b2018-02-26 14:23:09 -0800698 // This method is not supported with kUnifiedPlan semantics. Please use
699 // GetReceivers() instead.
Yves Gerey665174f2018-06-19 15:03:05 +0200700 virtual rtc::scoped_refptr<StreamCollectionInterface> remote_streams() = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000701
702 // Add a new MediaStream to be sent on this PeerConnection.
703 // Note that a SessionDescription negotiation is needed before the
704 // remote peer can receive the stream.
deadbeefb10f32f2017-02-08 01:38:21 -0800705 //
706 // This has been removed from the standard in favor of a track-based API. So,
707 // this is equivalent to simply calling AddTrack for each track within the
708 // stream, with the one difference that if "stream->AddTrack(...)" is called
709 // later, the PeerConnection will automatically pick up the new track. Though
710 // this functionality will be deprecated in the future.
Steve Antonab6ea6b2018-02-26 14:23:09 -0800711 //
712 // This method is not supported with kUnifiedPlan semantics. Please use
713 // AddTrack instead.
perkj@webrtc.orgfd0efb62014-11-06 12:16:36 +0000714 virtual bool AddStream(MediaStreamInterface* stream) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000715
716 // Remove a MediaStream from this PeerConnection.
deadbeefb10f32f2017-02-08 01:38:21 -0800717 // Note that a SessionDescription negotiation is needed before the
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000718 // remote peer is notified.
Steve Antonab6ea6b2018-02-26 14:23:09 -0800719 //
720 // This method is not supported with kUnifiedPlan semantics. Please use
721 // RemoveTrack instead.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000722 virtual void RemoveStream(MediaStreamInterface* stream) = 0;
723
deadbeefb10f32f2017-02-08 01:38:21 -0800724 // Add a new MediaStreamTrack to be sent on this PeerConnection, and return
Steve Antonf9381f02017-12-14 10:23:57 -0800725 // the newly created RtpSender. The RtpSender will be associated with the
Seth Hampson845e8782018-03-02 11:34:10 -0800726 // streams specified in the |stream_ids| list.
deadbeefb10f32f2017-02-08 01:38:21 -0800727 //
Steve Antonf9381f02017-12-14 10:23:57 -0800728 // Errors:
729 // - INVALID_PARAMETER: |track| is null, has a kind other than audio or video,
730 // or a sender already exists for the track.
731 // - INVALID_STATE: The PeerConnection is closed.
Steve Anton2d6c76a2018-01-05 17:10:52 -0800732 virtual RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrack(
733 rtc::scoped_refptr<MediaStreamTrackInterface> track,
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +0200734 const std::vector<std::string>& stream_ids);
deadbeefe1f9d832016-01-14 15:35:42 -0800735
736 // Remove an RtpSender from this PeerConnection.
737 // Returns true on success.
Steve Anton24db5732018-07-23 10:27:33 -0700738 // TODO(steveanton): Replace with signature that returns RTCError.
739 virtual bool RemoveTrack(RtpSenderInterface* sender);
740
741 // Plan B semantics: Removes the RtpSender from this PeerConnection.
742 // Unified Plan semantics: Stop sending on the RtpSender and mark the
743 // corresponding RtpTransceiver direction as no longer sending.
744 //
745 // Errors:
746 // - INVALID_PARAMETER: |sender| is null or (Plan B only) the sender is not
747 // associated with this PeerConnection.
748 // - INVALID_STATE: PeerConnection is closed.
749 // TODO(bugs.webrtc.org/9534): Rename to RemoveTrack once the other signature
750 // is removed.
751 virtual RTCError RemoveTrackNew(
752 rtc::scoped_refptr<RtpSenderInterface> sender);
deadbeefe1f9d832016-01-14 15:35:42 -0800753
Steve Anton9158ef62017-11-27 13:01:52 -0800754 // AddTransceiver creates a new RtpTransceiver and adds it to the set of
755 // transceivers. Adding a transceiver will cause future calls to CreateOffer
756 // to add a media description for the corresponding transceiver.
757 //
758 // The initial value of |mid| in the returned transceiver is null. Setting a
759 // new session description may change it to a non-null value.
760 //
761 // https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtransceiver
762 //
763 // Optionally, an RtpTransceiverInit structure can be specified to configure
764 // the transceiver from construction. If not specified, the transceiver will
765 // default to having a direction of kSendRecv and not be part of any streams.
766 //
767 // These methods are only available when Unified Plan is enabled (see
768 // RTCConfiguration).
769 //
770 // Common errors:
771 // - INTERNAL_ERROR: The configuration does not have Unified Plan enabled.
772 // TODO(steveanton): Make these pure virtual once downstream projects have
773 // updated.
774
775 // Adds a transceiver with a sender set to transmit the given track. The kind
776 // of the transceiver (and sender/receiver) will be derived from the kind of
777 // the track.
778 // Errors:
779 // - INVALID_PARAMETER: |track| is null.
780 virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +0200781 AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track);
Steve Anton9158ef62017-11-27 13:01:52 -0800782 virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
783 AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track,
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +0200784 const RtpTransceiverInit& init);
Steve Anton9158ef62017-11-27 13:01:52 -0800785
786 // Adds a transceiver with the given kind. Can either be MEDIA_TYPE_AUDIO or
787 // MEDIA_TYPE_VIDEO.
788 // Errors:
789 // - INVALID_PARAMETER: |media_type| is not MEDIA_TYPE_AUDIO or
790 // MEDIA_TYPE_VIDEO.
791 virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +0200792 AddTransceiver(cricket::MediaType media_type);
Steve Anton9158ef62017-11-27 13:01:52 -0800793 virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +0200794 AddTransceiver(cricket::MediaType media_type, const RtpTransceiverInit& init);
Steve Anton9158ef62017-11-27 13:01:52 -0800795
deadbeef70ab1a12015-09-28 16:53:55 -0700796 // TODO(deadbeef): Make these pure virtual once all subclasses implement them.
deadbeefb10f32f2017-02-08 01:38:21 -0800797
798 // Creates a sender without a track. Can be used for "early media"/"warmup"
799 // use cases, where the application may want to negotiate video attributes
800 // before a track is available to send.
801 //
802 // The standard way to do this would be through "addTransceiver", but we
803 // don't support that API yet.
804 //
deadbeeffac06552015-11-25 11:26:01 -0800805 // |kind| must be "audio" or "video".
deadbeefb10f32f2017-02-08 01:38:21 -0800806 //
deadbeefbd7d8f72015-12-18 16:58:44 -0800807 // |stream_id| is used to populate the msid attribute; if empty, one will
808 // be generated automatically.
Steve Antonab6ea6b2018-02-26 14:23:09 -0800809 //
810 // This method is not supported with kUnifiedPlan semantics. Please use
811 // AddTransceiver instead.
deadbeeffac06552015-11-25 11:26:01 -0800812 virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
deadbeefbd7d8f72015-12-18 16:58:44 -0800813 const std::string& kind,
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +0200814 const std::string& stream_id);
deadbeeffac06552015-11-25 11:26:01 -0800815
Steve Antonab6ea6b2018-02-26 14:23:09 -0800816 // If Plan B semantics are specified, gets all RtpSenders, created either
817 // through AddStream, AddTrack, or CreateSender. All senders of a specific
818 // media type share the same media description.
819 //
820 // If Unified Plan semantics are specified, gets the RtpSender for each
821 // RtpTransceiver.
deadbeef70ab1a12015-09-28 16:53:55 -0700822 virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +0200823 const;
deadbeef70ab1a12015-09-28 16:53:55 -0700824
Steve Antonab6ea6b2018-02-26 14:23:09 -0800825 // If Plan B semantics are specified, gets all RtpReceivers created when a
826 // remote description is applied. All receivers of a specific media type share
827 // the same media description. It is also possible to have a media description
828 // with no associated RtpReceivers, if the directional attribute does not
829 // indicate that the remote peer is sending any media.
deadbeefb10f32f2017-02-08 01:38:21 -0800830 //
Steve Antonab6ea6b2018-02-26 14:23:09 -0800831 // If Unified Plan semantics are specified, gets the RtpReceiver for each
832 // RtpTransceiver.
deadbeef70ab1a12015-09-28 16:53:55 -0700833 virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +0200834 const;
deadbeef70ab1a12015-09-28 16:53:55 -0700835
Steve Anton9158ef62017-11-27 13:01:52 -0800836 // Get all RtpTransceivers, created either through AddTransceiver, AddTrack or
837 // by a remote description applied with SetRemoteDescription.
Steve Antonab6ea6b2018-02-26 14:23:09 -0800838 //
Steve Anton9158ef62017-11-27 13:01:52 -0800839 // Note: This method is only available when Unified Plan is enabled (see
840 // RTCConfiguration).
841 virtual std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +0200842 GetTransceivers() const;
Steve Anton9158ef62017-11-27 13:01:52 -0800843
Henrik Boström1df1bf82018-03-20 13:24:20 +0100844 // The legacy non-compliant GetStats() API. This correspond to the
845 // callback-based version of getStats() in JavaScript. The returned metrics
846 // are UNDOCUMENTED and many of them rely on implementation-specific details.
847 // The goal is to DELETE THIS VERSION but we can't today because it is heavily
848 // relied upon by third parties. See https://crbug.com/822696.
849 //
850 // This version is wired up into Chrome. Any stats implemented are
851 // automatically exposed to the Web Platform. This has BYPASSED the Chrome
852 // release processes for years and lead to cross-browser incompatibility
853 // issues and web application reliance on Chrome-only behavior.
854 //
855 // This API is in "maintenance mode", serious regressions should be fixed but
856 // adding new stats is highly discouraged.
857 //
858 // TODO(hbos): Deprecate and remove this when third parties have migrated to
859 // the spec-compliant GetStats() API. https://crbug.com/822696
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000860 virtual bool GetStats(StatsObserver* observer,
Henrik Boström1df1bf82018-03-20 13:24:20 +0100861 MediaStreamTrackInterface* track, // Optional
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000862 StatsOutputLevel level) = 0;
Henrik Boström1df1bf82018-03-20 13:24:20 +0100863 // The spec-compliant GetStats() API. This correspond to the promise-based
864 // version of getStats() in JavaScript. Implementation status is described in
865 // api/stats/rtcstats_objects.h. For more details on stats, see spec:
866 // https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-getstats
867 // TODO(hbos): Takes shared ownership, use rtc::scoped_refptr<> instead. This
868 // requires stop overriding the current version in third party or making third
869 // party calls explicit to avoid ambiguity during switch. Make the future
870 // version abstract as soon as third party projects implement it.
hbose3810152016-12-13 02:35:19 -0800871 virtual void GetStats(RTCStatsCollectorCallback* callback) {}
Henrik Boström1df1bf82018-03-20 13:24:20 +0100872 // Spec-compliant getStats() performing the stats selection algorithm with the
873 // sender. https://w3c.github.io/webrtc-pc/#dom-rtcrtpsender-getstats
874 // TODO(hbos): Make abstract as soon as third party projects implement it.
875 virtual void GetStats(
876 rtc::scoped_refptr<RtpSenderInterface> selector,
877 rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {}
878 // Spec-compliant getStats() performing the stats selection algorithm with the
879 // receiver. https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-getstats
880 // TODO(hbos): Make abstract as soon as third party projects implement it.
881 virtual void GetStats(
882 rtc::scoped_refptr<RtpReceiverInterface> selector,
883 rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {}
Steve Antonab6ea6b2018-02-26 14:23:09 -0800884 // Clear cached stats in the RTCStatsCollector.
Harald Alvestrand89061872018-01-02 14:08:34 +0100885 // Exposed for testing while waiting for automatic cache clear to work.
886 // https://bugs.webrtc.org/8693
887 virtual void ClearStatsCache() {}
wu@webrtc.orgb9a088b2014-02-13 23:18:49 +0000888
deadbeefb10f32f2017-02-08 01:38:21 -0800889 // Create a data channel with the provided config, or default config if none
890 // is provided. Note that an offer/answer negotiation is still necessary
891 // before the data channel can be used.
892 //
893 // Also, calling CreateDataChannel is the only way to get a data "m=" section
894 // in SDP, so it should be done before CreateOffer is called, if the
895 // application plans to use data channels.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000896 virtual rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000897 const std::string& label,
898 const DataChannelInit* config) = 0;
899
deadbeefb10f32f2017-02-08 01:38:21 -0800900 // Returns the more recently applied description; "pending" if it exists, and
901 // otherwise "current". See below.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000902 virtual const SessionDescriptionInterface* local_description() const = 0;
903 virtual const SessionDescriptionInterface* remote_description() const = 0;
deadbeefb10f32f2017-02-08 01:38:21 -0800904
deadbeeffe4a8a42016-12-20 17:56:17 -0800905 // A "current" description the one currently negotiated from a complete
906 // offer/answer exchange.
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +0200907 virtual const SessionDescriptionInterface* current_local_description() const;
908 virtual const SessionDescriptionInterface* current_remote_description() const;
deadbeefb10f32f2017-02-08 01:38:21 -0800909
deadbeeffe4a8a42016-12-20 17:56:17 -0800910 // A "pending" description is one that's part of an incomplete offer/answer
911 // exchange (thus, either an offer or a pranswer). Once the offer/answer
912 // exchange is finished, the "pending" description will become "current".
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +0200913 virtual const SessionDescriptionInterface* pending_local_description() const;
914 virtual const SessionDescriptionInterface* pending_remote_description() const;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000915
916 // Create a new offer.
917 // The CreateSessionDescriptionObserver callback will be called when done.
918 virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
Niels Möllerf06f9232018-08-07 12:32:18 +0200919 const RTCOfferAnswerOptions& options) = 0;
jiayl@webrtc.orgb18bf5e2014-08-04 18:34:16 +0000920
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000921 // Create an answer to an offer.
922 // The CreateSessionDescriptionObserver callback will be called when done.
923 virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
Niels Möllerf06f9232018-08-07 12:32:18 +0200924 const RTCOfferAnswerOptions& options) = 0;
htaa2a49d92016-03-04 02:51:39 -0800925
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000926 // Sets the local session description.
deadbeef1dcb1642017-03-29 21:08:16 -0700927 // The PeerConnection takes the ownership of |desc| even if it fails.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000928 // The |observer| callback will be called when done.
deadbeef1dcb1642017-03-29 21:08:16 -0700929 // TODO(deadbeef): Change |desc| to be a unique_ptr, to make it clear
930 // that this method always takes ownership of it.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000931 virtual void SetLocalDescription(SetSessionDescriptionObserver* observer,
932 SessionDescriptionInterface* desc) = 0;
933 // Sets the remote session description.
deadbeef1dcb1642017-03-29 21:08:16 -0700934 // The PeerConnection takes the ownership of |desc| even if it fails.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000935 // The |observer| callback will be called when done.
Henrik Boström31638672017-11-23 17:48:32 +0100936 // TODO(hbos): Remove when Chrome implements the new signature.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000937 virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
Henrik Boström07109652017-11-27 09:52:02 +0100938 SessionDescriptionInterface* desc) {}
Henrik Boström31638672017-11-23 17:48:32 +0100939 // TODO(hbos): Make pure virtual when Chrome has updated its signature.
940 virtual void SetRemoteDescription(
941 std::unique_ptr<SessionDescriptionInterface> desc,
942 rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {}
deadbeefb10f32f2017-02-08 01:38:21 -0800943
deadbeef46c73892016-11-16 19:42:04 -0800944 // TODO(deadbeef): Make this pure virtual once all Chrome subclasses of
945 // PeerConnectionInterface implement it.
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +0200946 virtual PeerConnectionInterface::RTCConfiguration GetConfiguration();
deadbeef293e9262017-01-11 12:28:30 -0800947
deadbeefa67696b2015-09-29 11:56:26 -0700948 // Sets the PeerConnection's global configuration to |config|.
deadbeef293e9262017-01-11 12:28:30 -0800949 //
950 // The members of |config| that may be changed are |type|, |servers|,
951 // |ice_candidate_pool_size| and |prune_turn_ports| (though the candidate
952 // pool size can't be changed after the first call to SetLocalDescription).
953 // Note that this means the BUNDLE and RTCP-multiplexing policies cannot be
954 // changed with this method.
955 //
deadbeefa67696b2015-09-29 11:56:26 -0700956 // Any changes to STUN/TURN servers or ICE candidate policy will affect the
957 // next gathering phase, and cause the next call to createOffer to generate
deadbeef293e9262017-01-11 12:28:30 -0800958 // new ICE credentials, as described in JSEP. This also occurs when
959 // |prune_turn_ports| changes, for the same reasoning.
960 //
961 // If an error occurs, returns false and populates |error| if non-null:
962 // - INVALID_MODIFICATION if |config| contains a modified parameter other
963 // than one of the parameters listed above.
964 // - INVALID_RANGE if |ice_candidate_pool_size| is out of range.
965 // - SYNTAX_ERROR if parsing an ICE server URL failed.
966 // - INVALID_PARAMETER if a TURN server is missing |username| or |password|.
967 // - INTERNAL_ERROR if an unexpected error occurred.
968 //
deadbeefa67696b2015-09-29 11:56:26 -0700969 // TODO(deadbeef): Make this pure virtual once all Chrome subclasses of
970 // PeerConnectionInterface implement it.
971 virtual bool SetConfiguration(
deadbeef293e9262017-01-11 12:28:30 -0800972 const PeerConnectionInterface::RTCConfiguration& config,
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +0200973 RTCError* error);
974
deadbeef293e9262017-01-11 12:28:30 -0800975 // Version without error output param for backwards compatibility.
976 // TODO(deadbeef): Remove once chromium is updated.
977 virtual bool SetConfiguration(
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +0200978 const PeerConnectionInterface::RTCConfiguration& config);
deadbeefb10f32f2017-02-08 01:38:21 -0800979
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000980 // Provides a remote candidate to the ICE Agent.
981 // A copy of the |candidate| will be created and added to the remote
982 // description. So the caller of this method still has the ownership of the
983 // |candidate|.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000984 virtual bool AddIceCandidate(const IceCandidateInterface* candidate) = 0;
985
deadbeefb10f32f2017-02-08 01:38:21 -0800986 // Removes a group of remote candidates from the ICE agent. Needed mainly for
987 // continual gathering, to avoid an ever-growing list of candidates as
988 // networks come and go.
Honghai Zhang7fb69db2016-03-14 11:59:18 -0700989 virtual bool RemoveIceCandidates(
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +0200990 const std::vector<cricket::Candidate>& candidates);
Honghai Zhang7fb69db2016-03-14 11:59:18 -0700991
zstein4b979802017-06-02 14:37:37 -0700992 // 0 <= min <= current <= max should hold for set parameters.
993 struct BitrateParameters {
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +0200994 BitrateParameters();
995 ~BitrateParameters();
996
Danil Chapovalov0bc58cf2018-06-21 13:32:56 +0200997 absl::optional<int> min_bitrate_bps;
998 absl::optional<int> current_bitrate_bps;
999 absl::optional<int> max_bitrate_bps;
zstein4b979802017-06-02 14:37:37 -07001000 };
1001
1002 // SetBitrate limits the bandwidth allocated for all RTP streams sent by
1003 // this PeerConnection. Other limitations might affect these limits and
1004 // are respected (for example "b=AS" in SDP).
1005 //
1006 // Setting |current_bitrate_bps| will reset the current bitrate estimate
1007 // to the provided value.
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +02001008 virtual RTCError SetBitrate(const BitrateSettings& bitrate);
Niels Möller0c4f7be2018-05-07 14:01:37 +02001009
1010 // TODO(nisse): Deprecated - use version above. These two default
1011 // implementations require subclasses to implement one or the other
1012 // of the methods.
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +02001013 virtual RTCError SetBitrate(const BitrateParameters& bitrate_parameters);
zstein4b979802017-06-02 14:37:37 -07001014
Alex Narest78609d52017-10-20 10:37:47 +02001015 // Sets current strategy. If not set default WebRTC allocator will be used.
1016 // May be changed during an active session. The strategy
1017 // ownership is passed with std::unique_ptr
1018 // TODO(alexnarest): Make this pure virtual when tests will be updated
1019 virtual void SetBitrateAllocationStrategy(
1020 std::unique_ptr<rtc::BitrateAllocationStrategy>
1021 bitrate_allocation_strategy) {}
1022
henrika5f6bf242017-11-01 11:06:56 +01001023 // Enable/disable playout of received audio streams. Enabled by default. Note
1024 // that even if playout is enabled, streams will only be played out if the
1025 // appropriate SDP is also applied. Setting |playout| to false will stop
1026 // playout of the underlying audio device but starts a task which will poll
1027 // for audio data every 10ms to ensure that audio processing happens and the
1028 // audio statistics are updated.
1029 // TODO(henrika): deprecate and remove this.
1030 virtual void SetAudioPlayout(bool playout) {}
1031
1032 // Enable/disable recording of transmitted audio streams. Enabled by default.
1033 // Note that even if recording is enabled, streams will only be recorded if
1034 // the appropriate SDP is also applied.
1035 // TODO(henrika): deprecate and remove this.
1036 virtual void SetAudioRecording(bool recording) {}
1037
Harald Alvestrandad88c882018-11-28 16:47:46 +01001038 // Looks up the DtlsTransport associated with a MID value.
1039 // In the Javascript API, DtlsTransport is a property of a sender, but
1040 // because the PeerConnection owns the DtlsTransport in this implementation,
1041 // it is better to look them up on the PeerConnection.
Harald Alvestrand41390472018-12-03 18:45:19 +01001042 // TODO(hta): Remove default implementation after updating Chrome.
Harald Alvestrandad88c882018-11-28 16:47:46 +01001043 virtual rtc::scoped_refptr<DtlsTransportInterface> LookupDtlsTransportByMid(
1044 const std::string& mid);
Harald Alvestrandad88c882018-11-28 16:47:46 +01001045
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001046 // Returns the current SignalingState.
1047 virtual SignalingState signaling_state() = 0;
Taylor Brandstettercb423c42017-10-22 11:52:32 -07001048
Jonas Olsson12046902018-12-06 11:25:14 +01001049 // Returns an aggregate state of all ICE *and* DTLS transports.
1050 // This is left in place to avoid breaking native clients who expect our old,
1051 // nonstandard behavior.
1052 // TODO(jonasolsson): deprecate and remove this.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001053 virtual IceConnectionState ice_connection_state() = 0;
Taylor Brandstettercb423c42017-10-22 11:52:32 -07001054
Jonas Olsson12046902018-12-06 11:25:14 +01001055 // Returns an aggregated state of all ICE transports.
1056 virtual IceConnectionState standardized_ice_connection_state();
1057
1058 // Returns an aggregated state of all ICE and DTLS transports.
Jonas Olsson635474e2018-10-18 15:58:17 +02001059 virtual PeerConnectionState peer_connection_state();
1060
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001061 virtual IceGatheringState ice_gathering_state() = 0;
1062
ivoc14d5dbe2016-07-04 07:06:55 -07001063 // Starts RtcEventLog using existing file. Takes ownership of |file| and
1064 // passes it on to Call, which will take the ownership. If the
1065 // operation fails the file will be closed. The logging will stop
1066 // automatically after 10 minutes have passed, or when the StopRtcEventLog
1067 // function is called.
Elad Alon99c3fe52017-10-13 16:29:40 +02001068 // TODO(eladalon): Deprecate and remove this.
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +02001069 virtual bool StartRtcEventLog(rtc::PlatformFile file, int64_t max_size_bytes);
ivoc14d5dbe2016-07-04 07:06:55 -07001070
Elad Alon99c3fe52017-10-13 16:29:40 +02001071 // Start RtcEventLog using an existing output-sink. Takes ownership of
1072 // |output| and passes it on to Call, which will take the ownership. If the
Bjorn Tereliusde939432017-11-20 17:38:14 +01001073 // operation fails the output will be closed and deallocated. The event log
1074 // will send serialized events to the output object every |output_period_ms|.
1075 virtual bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +02001076 int64_t output_period_ms);
Elad Alon99c3fe52017-10-13 16:29:40 +02001077
ivoc14d5dbe2016-07-04 07:06:55 -07001078 // Stops logging the RtcEventLog.
1079 // TODO(ivoc): Make this pure virtual when Chrome is updated.
1080 virtual void StopRtcEventLog() {}
1081
deadbeefb10f32f2017-02-08 01:38:21 -08001082 // Terminates all media, closes the transports, and in general releases any
1083 // resources used by the PeerConnection. This is an irreversible operation.
deadbeefd07061c2017-04-20 13:19:00 -07001084 //
1085 // Note that after this method completes, the PeerConnection will no longer
1086 // use the PeerConnectionObserver interface passed in on construction, and
1087 // thus the observer object can be safely destroyed.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001088 virtual void Close() = 0;
1089
1090 protected:
1091 // Dtor protected as objects shouldn't be deleted via this interface.
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +02001092 ~PeerConnectionInterface() override = default;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001093};
1094
deadbeefb10f32f2017-02-08 01:38:21 -08001095// PeerConnection callback interface, used for RTCPeerConnection events.
1096// Application should implement these methods.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001097class PeerConnectionObserver {
1098 public:
Sami Kalliomäki02879f92018-01-11 10:02:19 +01001099 virtual ~PeerConnectionObserver() = default;
1100
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001101 // Triggered when the SignalingState changed.
1102 virtual void OnSignalingChange(
perkjdfb769d2016-02-09 03:09:43 -08001103 PeerConnectionInterface::SignalingState new_state) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001104
1105 // Triggered when media is received on a new stream from remote peer.
Steve Anton772eb212018-01-16 10:11:06 -08001106 virtual void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001107
Steve Anton3172c032018-05-03 15:30:18 -07001108 // Triggered when a remote peer closes a stream.
Steve Anton772eb212018-01-16 10:11:06 -08001109 virtual void OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface> stream) {
1110 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001111
Taylor Brandstetter98cde262016-05-31 13:02:21 -07001112 // Triggered when a remote peer opens a data channel.
1113 virtual void OnDataChannel(
nisse7f067662017-03-08 06:59:45 -08001114 rtc::scoped_refptr<DataChannelInterface> data_channel) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001115
Taylor Brandstetter98cde262016-05-31 13:02:21 -07001116 // Triggered when renegotiation is needed. For example, an ICE restart
1117 // has begun.
fischman@webrtc.orgd7568a02014-01-13 22:04:12 +00001118 virtual void OnRenegotiationNeeded() = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001119
Jonas Olsson12046902018-12-06 11:25:14 +01001120 // Called any time the legacy IceConnectionState changes.
deadbeefb10f32f2017-02-08 01:38:21 -08001121 //
1122 // Note that our ICE states lag behind the standard slightly. The most
1123 // notable differences include the fact that "failed" occurs after 15
1124 // seconds, not 30, and this actually represents a combination ICE + DTLS
1125 // state, so it may be "failed" if DTLS fails while ICE succeeds.
Jonas Olsson12046902018-12-06 11:25:14 +01001126 //
1127 // TODO(jonasolsson): deprecate and remove this.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001128 virtual void OnIceConnectionChange(
perkjdfb769d2016-02-09 03:09:43 -08001129 PeerConnectionInterface::IceConnectionState new_state) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001130
Jonas Olsson12046902018-12-06 11:25:14 +01001131 // Called any time the standards-compliant IceConnectionState changes.
1132 virtual void OnStandardizedIceConnectionChange(
1133 PeerConnectionInterface::IceConnectionState new_state) {}
1134
Jonas Olsson635474e2018-10-18 15:58:17 +02001135 // Called any time the PeerConnectionState changes.
1136 virtual void OnConnectionChange(
1137 PeerConnectionInterface::PeerConnectionState new_state) {}
1138
Taylor Brandstetter98cde262016-05-31 13:02:21 -07001139 // Called any time the IceGatheringState changes.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001140 virtual void OnIceGatheringChange(
perkjdfb769d2016-02-09 03:09:43 -08001141 PeerConnectionInterface::IceGatheringState new_state) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001142
Taylor Brandstetter98cde262016-05-31 13:02:21 -07001143 // A new ICE candidate has been gathered.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001144 virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
1145
Honghai Zhang7fb69db2016-03-14 11:59:18 -07001146 // Ice candidates have been removed.
1147 // TODO(honghaiz): Make this a pure virtual method when all its subclasses
1148 // implement it.
1149 virtual void OnIceCandidatesRemoved(
1150 const std::vector<cricket::Candidate>& candidates) {}
1151
Peter Thatcher54360512015-07-08 11:08:35 -07001152 // Called when the ICE connection receiving status changes.
1153 virtual void OnIceConnectionReceivingChange(bool receiving) {}
1154
Steve Antonab6ea6b2018-02-26 14:23:09 -08001155 // This is called when a receiver and its track are created.
Henrik Boström933d8b02017-10-10 10:05:16 -07001156 // TODO(zhihuang): Make this pure virtual when all subclasses implement it.
Steve Anton8b815cd2018-02-16 16:14:42 -08001157 // Note: This is called with both Plan B and Unified Plan semantics. Unified
1158 // Plan users should prefer OnTrack, OnAddTrack is only called as backwards
1159 // compatibility (and is called in the exact same situations as OnTrack).
zhihuang81c3a032016-11-17 12:06:24 -08001160 virtual void OnAddTrack(
1161 rtc::scoped_refptr<RtpReceiverInterface> receiver,
zhihuangc63b8942016-12-02 15:41:10 -08001162 const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {}
zhihuang81c3a032016-11-17 12:06:24 -08001163
Steve Anton8b815cd2018-02-16 16:14:42 -08001164 // This is called when signaling indicates a transceiver will be receiving
1165 // media from the remote endpoint. This is fired during a call to
1166 // SetRemoteDescription. The receiving track can be accessed by:
1167 // |transceiver->receiver()->track()| and its associated streams by
1168 // |transceiver->receiver()->streams()|.
1169 // Note: This will only be called if Unified Plan semantics are specified.
1170 // This behavior is specified in section 2.2.8.2.5 of the "Set the
1171 // RTCSessionDescription" algorithm:
1172 // https://w3c.github.io/webrtc-pc/#set-description
1173 virtual void OnTrack(
1174 rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {}
1175
Steve Anton3172c032018-05-03 15:30:18 -07001176 // Called when signaling indicates that media will no longer be received on a
1177 // track.
1178 // With Plan B semantics, the given receiver will have been removed from the
1179 // PeerConnection and the track muted.
1180 // With Unified Plan semantics, the receiver will remain but the transceiver
1181 // will have changed direction to either sendonly or inactive.
Henrik Boström933d8b02017-10-10 10:05:16 -07001182 // https://w3c.github.io/webrtc-pc/#process-remote-track-removal
Henrik Boström933d8b02017-10-10 10:05:16 -07001183 // TODO(hbos,deadbeef): Make pure virtual when all subclasses implement it.
1184 virtual void OnRemoveTrack(
1185 rtc::scoped_refptr<RtpReceiverInterface> receiver) {}
Harald Alvestrandc0e97252018-07-26 10:39:55 +02001186
1187 // Called when an interesting usage is detected by WebRTC.
1188 // An appropriate action is to add information about the context of the
1189 // PeerConnection and write the event to some kind of "interesting events"
1190 // log function.
1191 // The heuristics for defining what constitutes "interesting" are
1192 // implementation-defined.
1193 virtual void OnInterestingUsage(int usage_pattern) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001194};
1195
Benjamin Wright6f7e6d62018-05-02 13:46:31 -07001196// PeerConnectionDependencies holds all of PeerConnections dependencies.
1197// A dependency is distinct from a configuration as it defines significant
1198// executable code that can be provided by a user of the API.
1199//
1200// All new dependencies should be added as a unique_ptr to allow the
1201// PeerConnection object to be the definitive owner of the dependencies
1202// lifetime making injection safer.
1203struct PeerConnectionDependencies final {
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +02001204 explicit PeerConnectionDependencies(PeerConnectionObserver* observer_in);
Benjamin Wright6f7e6d62018-05-02 13:46:31 -07001205 // This object is not copyable or assignable.
1206 PeerConnectionDependencies(const PeerConnectionDependencies&) = delete;
1207 PeerConnectionDependencies& operator=(const PeerConnectionDependencies&) =
1208 delete;
1209 // This object is only moveable.
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +02001210 PeerConnectionDependencies(PeerConnectionDependencies&&);
Benjamin Wright6f7e6d62018-05-02 13:46:31 -07001211 PeerConnectionDependencies& operator=(PeerConnectionDependencies&&) = default;
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +02001212 ~PeerConnectionDependencies();
Benjamin Wright6f7e6d62018-05-02 13:46:31 -07001213 // Mandatory dependencies
1214 PeerConnectionObserver* observer = nullptr;
1215 // Optional dependencies
1216 std::unique_ptr<cricket::PortAllocator> allocator;
Zach Steine20867f2018-08-02 13:20:15 -07001217 std::unique_ptr<webrtc::AsyncResolverFactory> async_resolver_factory;
Benjamin Wright6f7e6d62018-05-02 13:46:31 -07001218 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator;
Benjamin Wrightd6f86e82018-05-08 13:12:25 -07001219 std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier;
Benjamin Wright6f7e6d62018-05-02 13:46:31 -07001220};
1221
Benjamin Wright5234a492018-05-29 15:04:32 -07001222// PeerConnectionFactoryDependencies holds all of the PeerConnectionFactory
1223// dependencies. All new dependencies should be added here instead of
1224// overloading the function. This simplifies dependency injection and makes it
1225// clear which are mandatory and optional. If possible please allow the peer
1226// connection factory to take ownership of the dependency by adding a unique_ptr
1227// to this structure.
1228struct PeerConnectionFactoryDependencies final {
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +02001229 PeerConnectionFactoryDependencies();
Benjamin Wright5234a492018-05-29 15:04:32 -07001230 // This object is not copyable or assignable.
1231 PeerConnectionFactoryDependencies(const PeerConnectionFactoryDependencies&) =
1232 delete;
1233 PeerConnectionFactoryDependencies& operator=(
1234 const PeerConnectionFactoryDependencies&) = delete;
1235 // This object is only moveable.
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +02001236 PeerConnectionFactoryDependencies(PeerConnectionFactoryDependencies&&);
Benjamin Wright5234a492018-05-29 15:04:32 -07001237 PeerConnectionFactoryDependencies& operator=(
1238 PeerConnectionFactoryDependencies&&) = default;
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +02001239 ~PeerConnectionFactoryDependencies();
Benjamin Wright5234a492018-05-29 15:04:32 -07001240
1241 // Optional dependencies
1242 rtc::Thread* network_thread = nullptr;
1243 rtc::Thread* worker_thread = nullptr;
1244 rtc::Thread* signaling_thread = nullptr;
1245 std::unique_ptr<cricket::MediaEngineInterface> media_engine;
1246 std::unique_ptr<CallFactoryInterface> call_factory;
1247 std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory;
1248 std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory;
1249 std::unique_ptr<NetworkControllerFactoryInterface> network_controller_factory;
Piotr (Peter) Slatalae0c2e972018-10-08 09:43:21 -07001250 std::unique_ptr<MediaTransportFactory> media_transport_factory;
Benjamin Wright5234a492018-05-29 15:04:32 -07001251};
1252
deadbeefb10f32f2017-02-08 01:38:21 -08001253// PeerConnectionFactoryInterface is the factory interface used for creating
1254// PeerConnection, MediaStream and MediaStreamTrack objects.
1255//
1256// The simplest method for obtaiing one, CreatePeerConnectionFactory will
1257// create the required libjingle threads, socket and network manager factory
1258// classes for networking if none are provided, though it requires that the
1259// application runs a message loop on the thread that called the method (see
1260// explanation below)
1261//
1262// If an application decides to provide its own threads and/or implementation
1263// of networking classes, it should use the alternate
1264// CreatePeerConnectionFactory method which accepts threads as input, and use
1265// the CreatePeerConnection version that takes a PortAllocator as an argument.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001266class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001267 public:
wu@webrtc.org97077a32013-10-25 21:18:33 +00001268 class Options {
1269 public:
Benjamin Wrighta54daf12018-10-11 15:33:17 -07001270 Options() {}
deadbeefb10f32f2017-02-08 01:38:21 -08001271
1272 // If set to true, created PeerConnections won't enforce any SRTP
1273 // requirement, allowing unsecured media. Should only be used for
1274 // testing/debugging.
1275 bool disable_encryption = false;
1276
1277 // Deprecated. The only effect of setting this to true is that
1278 // CreateDataChannel will fail, which is not that useful.
1279 bool disable_sctp_data_channels = false;
1280
1281 // If set to true, any platform-supported network monitoring capability
1282 // won't be used, and instead networks will only be updated via polling.
1283 //
1284 // This only has an effect if a PeerConnection is created with the default
1285 // PortAllocator implementation.
1286 bool disable_network_monitor = false;
phoglund@webrtc.org006521d2015-02-12 09:23:59 +00001287
1288 // Sets the network types to ignore. For instance, calling this with
1289 // ADAPTER_TYPE_ETHERNET | ADAPTER_TYPE_LOOPBACK will ignore Ethernet and
1290 // loopback interfaces.
deadbeefb10f32f2017-02-08 01:38:21 -08001291 int network_ignore_mask = rtc::kDefaultNetworkIgnoreMask;
Joachim Bauch04e5b492015-05-29 09:40:39 +02001292
1293 // Sets the maximum supported protocol version. The highest version
1294 // supported by both ends will be used for the connection, i.e. if one
1295 // party supports DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used.
deadbeefb10f32f2017-02-08 01:38:21 -08001296 rtc::SSLProtocolVersion ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
jbauchcb560652016-08-04 05:20:32 -07001297
1298 // Sets crypto related options, e.g. enabled cipher suites.
Benjamin Wrighta54daf12018-10-11 15:33:17 -07001299 CryptoOptions crypto_options = CryptoOptions::NoGcm();
wu@webrtc.org97077a32013-10-25 21:18:33 +00001300 };
1301
deadbeef7914b8c2017-04-21 03:23:33 -07001302 // Set the options to be used for subsequently created PeerConnections.
wu@webrtc.org97077a32013-10-25 21:18:33 +00001303 virtual void SetOptions(const Options& options) = 0;
buildbot@webrtc.org41451d42014-05-03 05:39:45 +00001304
Benjamin Wright6f7e6d62018-05-02 13:46:31 -07001305 // The preferred way to create a new peer connection. Simply provide the
1306 // configuration and a PeerConnectionDependencies structure.
1307 // TODO(benwright): Make pure virtual once downstream mock PC factory classes
1308 // are updated.
1309 virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
1310 const PeerConnectionInterface::RTCConfiguration& configuration,
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +02001311 PeerConnectionDependencies dependencies);
Benjamin Wright6f7e6d62018-05-02 13:46:31 -07001312
1313 // Deprecated; |allocator| and |cert_generator| may be null, in which case
1314 // default implementations will be used.
deadbeefd07061c2017-04-20 13:19:00 -07001315 //
1316 // |observer| must not be null.
1317 //
1318 // Note that this method does not take ownership of |observer|; it's the
1319 // responsibility of the caller to delete it. It can be safely deleted after
1320 // Close has been called on the returned PeerConnection, which ensures no
1321 // more observer callbacks will be invoked.
deadbeef41b07982015-12-01 15:01:24 -08001322 virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
1323 const PeerConnectionInterface::RTCConfiguration& configuration,
kwibergd1fe2812016-04-27 06:47:29 -07001324 std::unique_ptr<cricket::PortAllocator> allocator,
Henrik Boströmd03c23b2016-06-01 11:44:18 +02001325 std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +02001326 PeerConnectionObserver* observer);
1327
Florent Castelli72b751a2018-06-28 14:09:33 +02001328 // Returns the capabilities of an RTP sender of type |kind|.
1329 // If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
1330 // TODO(orphis): Make pure virtual when all subclasses implement it.
1331 virtual RtpCapabilities GetRtpSenderCapabilities(
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +02001332 cricket::MediaType kind) const;
Florent Castelli72b751a2018-06-28 14:09:33 +02001333
1334 // Returns the capabilities of an RTP receiver of type |kind|.
1335 // If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
1336 // TODO(orphis): Make pure virtual when all subclasses implement it.
1337 virtual RtpCapabilities GetRtpReceiverCapabilities(
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +02001338 cricket::MediaType kind) const;
Florent Castelli72b751a2018-06-28 14:09:33 +02001339
Seth Hampson845e8782018-03-02 11:34:10 -08001340 virtual rtc::scoped_refptr<MediaStreamInterface> CreateLocalMediaStream(
1341 const std::string& stream_id) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001342
deadbeefe814a0d2017-02-25 18:15:09 -08001343 // Creates an AudioSourceInterface.
deadbeefb10f32f2017-02-08 01:38:21 -08001344 // |options| decides audio processing settings.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +00001345 virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
htaa2a49d92016-03-04 02:51:39 -08001346 const cricket::AudioOptions& options) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001347
deadbeef39e14da2017-02-13 09:49:58 -08001348 // Creates a VideoTrackSourceInterface from |capturer|.
1349 // TODO(deadbeef): We should aim to remove cricket::VideoCapturer from the
1350 // API. It's mainly used as a wrapper around webrtc's provided
1351 // platform-specific capturers, but these should be refactored to use
1352 // VideoTrackSourceInterface directly.
deadbeef112b2e92017-02-10 20:13:37 -08001353 // TODO(deadbeef): Make pure virtual once downstream mock PC factory classes
1354 // are updated.
perkja3ede6c2016-03-08 01:27:48 +01001355 virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +02001356 std::unique_ptr<cricket::VideoCapturer> capturer);
deadbeef112b2e92017-02-10 20:13:37 -08001357
htaa2a49d92016-03-04 02:51:39 -08001358 // A video source creator that allows selection of resolution and frame rate.
deadbeef8d60a942017-02-27 14:47:33 -08001359 // |constraints| decides video resolution and frame rate but can be null.
1360 // In the null case, use the version above.
deadbeef112b2e92017-02-10 20:13:37 -08001361 //
1362 // |constraints| is only used for the invocation of this method, and can
1363 // safely be destroyed afterwards.
1364 virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
1365 std::unique_ptr<cricket::VideoCapturer> capturer,
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +02001366 const MediaConstraintsInterface* constraints);
deadbeef112b2e92017-02-10 20:13:37 -08001367
1368 // Deprecated; please use the versions that take unique_ptrs above.
1369 // TODO(deadbeef): Remove these once safe to do so.
1370 virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +02001371 cricket::VideoCapturer* capturer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001372 // Creates a new local VideoTrack. The same |source| can be used in several
1373 // tracks.
perkja3ede6c2016-03-08 01:27:48 +01001374 virtual rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
1375 const std::string& label,
1376 VideoTrackSourceInterface* source) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001377
deadbeef8d60a942017-02-27 14:47:33 -08001378 // Creates an new AudioTrack. At the moment |source| can be null.
Yves Gerey665174f2018-06-19 15:03:05 +02001379 virtual rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
1380 const std::string& label,
1381 AudioSourceInterface* source) = 0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001382
wu@webrtc.orga9890802013-12-13 00:21:03 +00001383 // Starts AEC dump using existing file. Takes ownership of |file| and passes
1384 // it on to VoiceEngine (via other objects) immediately, which will take
wu@webrtc.orga8910d22014-01-23 22:12:45 +00001385 // the ownerhip. If the operation fails, the file will be closed.
ivocd66b44d2016-01-15 03:06:36 -08001386 // A maximum file size in bytes can be specified. When the file size limit is
1387 // reached, logging is stopped automatically. If max_size_bytes is set to a
1388 // value <= 0, no limit will be used, and logging will continue until the
1389 // StopAecDump function is called.
1390 virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0;
wu@webrtc.orga9890802013-12-13 00:21:03 +00001391
ivoc797ef122015-10-22 03:25:41 -07001392 // Stops logging the AEC dump.
1393 virtual void StopAecDump() = 0;
1394
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001395 protected:
1396 // Dtor and ctor protected as objects shouldn't be created or deleted via
1397 // this interface.
1398 PeerConnectionFactoryInterface() {}
Mirko Bonadei79eb4dd2018-07-19 10:39:30 +02001399 ~PeerConnectionFactoryInterface() override = default;
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001400};
1401
zhihuang38ede132017-06-15 12:52:32 -07001402// This is a lower-level version of the CreatePeerConnectionFactory functions
1403// above. It's implemented in the "peerconnection" build target, whereas the
1404// above methods are only implemented in the broader "libjingle_peerconnection"
1405// build target, which pulls in the implementations of every module webrtc may
1406// use.
1407//
1408// If an application knows it will only require certain modules, it can reduce
1409// webrtc's impact on its binary size by depending only on the "peerconnection"
1410// target and the modules the application requires, using
1411// CreateModularPeerConnectionFactory instead of one of the
1412// CreatePeerConnectionFactory methods above. For example, if an application
1413// only uses WebRTC for audio, it can pass in null pointers for the
1414// video-specific interfaces, and omit the corresponding modules from its
1415// build.
1416//
1417// If |network_thread| or |worker_thread| are null, the PeerConnectionFactory
1418// will create the necessary thread internally. If |signaling_thread| is null,
1419// the PeerConnectionFactory will use the thread on which this method is called
1420// as the signaling thread, wrapping it in an rtc::Thread object if needed.
1421//
1422// If non-null, a reference is added to |default_adm|, and ownership of
1423// |video_encoder_factory| and |video_decoder_factory| is transferred to the
1424// returned factory.
1425//
peaha9cc40b2017-06-29 08:32:09 -07001426// If |audio_mixer| is null, an internal audio mixer will be created and used.
1427//
zhihuang38ede132017-06-15 12:52:32 -07001428// TODO(deadbeef): Use rtc::scoped_refptr<> and std::unique_ptr<> to make this
1429// ownership transfer and ref counting more obvious.
1430//
1431// TODO(deadbeef): Encapsulate these modules in a struct, so that when a new
1432// module is inevitably exposed, we can just add a field to the struct instead
1433// of adding a whole new CreateModularPeerConnectionFactory overload.
1434rtc::scoped_refptr<PeerConnectionFactoryInterface>
1435CreateModularPeerConnectionFactory(
1436 rtc::Thread* network_thread,
1437 rtc::Thread* worker_thread,
1438 rtc::Thread* signaling_thread,
zhihuang38ede132017-06-15 12:52:32 -07001439 std::unique_ptr<cricket::MediaEngineInterface> media_engine,
1440 std::unique_ptr<CallFactoryInterface> call_factory,
1441 std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory);
1442
Ying Wang0dd1b0a2018-02-20 12:50:27 +01001443rtc::scoped_refptr<PeerConnectionFactoryInterface>
1444CreateModularPeerConnectionFactory(
1445 rtc::Thread* network_thread,
1446 rtc::Thread* worker_thread,
1447 rtc::Thread* signaling_thread,
1448 std::unique_ptr<cricket::MediaEngineInterface> media_engine,
1449 std::unique_ptr<CallFactoryInterface> call_factory,
1450 std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory,
Sebastian Janssondfce03a2018-05-18 18:05:10 +02001451 std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory,
1452 std::unique_ptr<NetworkControllerFactoryInterface>
1453 network_controller_factory = nullptr);
Ying Wang0dd1b0a2018-02-20 12:50:27 +01001454
Benjamin Wright5234a492018-05-29 15:04:32 -07001455rtc::scoped_refptr<PeerConnectionFactoryInterface>
1456CreateModularPeerConnectionFactory(
1457 PeerConnectionFactoryDependencies dependencies);
1458
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001459} // namespace webrtc
1460
Mirko Bonadei92ea95e2017-09-15 06:47:31 +02001461#endif // API_PEERCONNECTIONINTERFACE_H_