731 lines
28 KiB
C
731 lines
28 KiB
C
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/*
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* Copyright 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// This file contains the PeerConnection interface as defined in
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// http://dev.w3.org/2011/webrtc/editor/webrtc.html#peer-to-peer-connections.
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// Applications must use this interface to implement peerconnection.
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// PeerConnectionFactory class provides factory methods to create
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// peerconnection, mediastream and media tracks objects.
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//
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// The Following steps are needed to setup a typical call using Jsep.
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// 1. Create a PeerConnectionFactoryInterface. Check constructors for more
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// information about input parameters.
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// 2. Create a PeerConnection object. Provide a configuration string which
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// points either to stun or turn server to generate ICE candidates and provide
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// an object that implements the PeerConnectionObserver interface.
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// 3. Create local MediaStream and MediaTracks using the PeerConnectionFactory
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// and add it to PeerConnection by calling AddStream.
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// 4. Create an offer and serialize it and send it to the remote peer.
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// 5. Once an ice candidate have been found PeerConnection will call the
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// observer function OnIceCandidate. The candidates must also be serialized and
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// sent to the remote peer.
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// 6. Once an answer is received from the remote peer, call
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// SetLocalSessionDescription with the offer and SetRemoteSessionDescription
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// with the remote answer.
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// 7. Once a remote candidate is received from the remote peer, provide it to
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// the peerconnection by calling AddIceCandidate.
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// The Receiver of a call can decide to accept or reject the call.
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// This decision will be taken by the application not peerconnection.
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// If application decides to accept the call
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// 1. Create PeerConnectionFactoryInterface if it doesn't exist.
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// 2. Create a new PeerConnection.
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// 3. Provide the remote offer to the new PeerConnection object by calling
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// SetRemoteSessionDescription.
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// 4. Generate an answer to the remote offer by calling CreateAnswer and send it
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// back to the remote peer.
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// 5. Provide the local answer to the new PeerConnection by calling
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// SetLocalSessionDescription with the answer.
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// 6. Provide the remote ice candidates by calling AddIceCandidate.
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// 7. Once a candidate have been found PeerConnection will call the observer
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// function OnIceCandidate. Send these candidates to the remote peer.
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#ifndef WEBRTC_API_PEERCONNECTIONINTERFACE_H_
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#define WEBRTC_API_PEERCONNECTIONINTERFACE_H_
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#include <memory>
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#include <string>
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#include <utility>
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#include <vector>
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#include "webrtc/api/datachannelinterface.h"
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#include "webrtc/api/dtmfsenderinterface.h"
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#include "webrtc/api/jsep.h"
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#include "webrtc/api/mediastreaminterface.h"
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#include "webrtc/api/rtpreceiverinterface.h"
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#include "webrtc/api/rtpsenderinterface.h"
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#include "webrtc/api/statstypes.h"
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#include "webrtc/api/umametrics.h"
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#include "webrtc/base/fileutils.h"
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#include "webrtc/base/network.h"
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#include "webrtc/base/rtccertificate.h"
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#include "webrtc/base/rtccertificategenerator.h"
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#include "webrtc/base/socketaddress.h"
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#include "webrtc/base/sslstreamadapter.h"
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#include "webrtc/media/base/mediachannel.h"
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#include "webrtc/p2p/base/portallocator.h"
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namespace rtc {
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class SSLIdentity;
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class Thread;
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}
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namespace cricket {
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class WebRtcVideoDecoderFactory;
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class WebRtcVideoEncoderFactory;
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}
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namespace webrtc {
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class AudioDeviceModule;
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class MediaConstraintsInterface;
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// MediaStream container interface.
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class StreamCollectionInterface : public rtc::RefCountInterface {
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public:
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// TODO(ronghuawu): Update the function names to c++ style, e.g. find -> Find.
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virtual size_t count() = 0;
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virtual MediaStreamInterface* at(size_t index) = 0;
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virtual MediaStreamInterface* find(const std::string& label) = 0;
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virtual MediaStreamTrackInterface* FindAudioTrack(
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const std::string& id) = 0;
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virtual MediaStreamTrackInterface* FindVideoTrack(
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const std::string& id) = 0;
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protected:
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// Dtor protected as objects shouldn't be deleted via this interface.
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~StreamCollectionInterface() {}
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};
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class StatsObserver : public rtc::RefCountInterface {
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public:
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virtual void OnComplete(const StatsReports& reports) = 0;
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protected:
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virtual ~StatsObserver() {}
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};
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class MetricsObserverInterface : public rtc::RefCountInterface {
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public:
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// |type| is the type of the enum counter to be incremented. |counter|
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// is the particular counter in that type. |counter_max| is the next sequence
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// number after the highest counter.
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virtual void IncrementEnumCounter(PeerConnectionEnumCounterType type,
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int counter,
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int counter_max) {}
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// This is used to handle sparse counters like SSL cipher suites.
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// TODO(guoweis): Remove the implementation once the dependency's interface
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// definition is updated.
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virtual void IncrementSparseEnumCounter(PeerConnectionEnumCounterType type,
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int counter) {
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IncrementEnumCounter(type, counter, 0 /* Ignored */);
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}
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virtual void AddHistogramSample(PeerConnectionMetricsName type,
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int value) = 0;
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protected:
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virtual ~MetricsObserverInterface() {}
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};
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typedef MetricsObserverInterface UMAObserver;
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class PeerConnectionInterface : public rtc::RefCountInterface {
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public:
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// See http://dev.w3.org/2011/webrtc/editor/webrtc.html#state-definitions .
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enum SignalingState {
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kStable,
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kHaveLocalOffer,
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kHaveLocalPrAnswer,
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kHaveRemoteOffer,
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kHaveRemotePrAnswer,
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kClosed,
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};
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// TODO(bemasc): Remove IceState when callers are changed to
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// IceConnection/GatheringState.
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enum IceState {
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kIceNew,
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kIceGathering,
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kIceWaiting,
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kIceChecking,
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kIceConnected,
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kIceCompleted,
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kIceFailed,
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kIceClosed,
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};
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enum IceGatheringState {
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kIceGatheringNew,
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kIceGatheringGathering,
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kIceGatheringComplete
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};
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enum IceConnectionState {
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kIceConnectionNew,
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kIceConnectionChecking,
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kIceConnectionConnected,
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kIceConnectionCompleted,
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kIceConnectionFailed,
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kIceConnectionDisconnected,
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kIceConnectionClosed,
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kIceConnectionMax,
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};
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struct IceServer {
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// TODO(jbauch): Remove uri when all code using it has switched to urls.
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std::string uri;
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std::vector<std::string> urls;
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std::string username;
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std::string password;
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};
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typedef std::vector<IceServer> IceServers;
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enum IceTransportsType {
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// TODO(pthatcher): Rename these kTransporTypeXXX, but update
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// Chromium at the same time.
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kNone,
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kRelay,
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kNoHost,
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kAll
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};
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// https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-08#section-4.1.1
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enum BundlePolicy {
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kBundlePolicyBalanced,
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kBundlePolicyMaxBundle,
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kBundlePolicyMaxCompat
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};
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// https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-09#section-4.1.1
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enum RtcpMuxPolicy {
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kRtcpMuxPolicyNegotiate,
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kRtcpMuxPolicyRequire,
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};
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enum TcpCandidatePolicy {
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kTcpCandidatePolicyEnabled,
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kTcpCandidatePolicyDisabled
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};
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enum CandidateNetworkPolicy {
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kCandidateNetworkPolicyAll,
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kCandidateNetworkPolicyLowCost
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};
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enum ContinualGatheringPolicy {
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GATHER_ONCE,
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GATHER_CONTINUALLY
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};
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// TODO(hbos): Change into class with private data and public getters.
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// TODO(nisse): In particular, accessing fields directly from an
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// application is brittle, since the organization mirrors the
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// organization of the implementation, which isn't stable. So we
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// need getters and setters at least for fields which applications
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// are interested in.
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struct RTCConfiguration {
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// This struct is subject to reorganization, both for naming
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// consistency, and to group settings to match where they are used
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// in the implementation. To do that, we need getter and setter
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// methods for all settings which are of interest to applications,
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// Chrome in particular.
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bool dscp() { return media_config.enable_dscp; }
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void set_dscp(bool enable) { media_config.enable_dscp = enable; }
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// TODO(nisse): The corresponding flag in MediaConfig and
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// elsewhere should be renamed enable_cpu_adaptation.
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bool cpu_adaptation() {
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return media_config.video.enable_cpu_overuse_detection;
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}
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void set_cpu_adaptation(bool enable) {
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media_config.video.enable_cpu_overuse_detection = enable;
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}
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bool suspend_below_min_bitrate() {
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return media_config.video.suspend_below_min_bitrate;
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}
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void set_suspend_below_min_bitrate(bool enable) {
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media_config.video.suspend_below_min_bitrate = enable;
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}
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// TODO(nisse): The negation in the corresponding MediaConfig
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// attribute is inconsistent, and it should be renamed at some
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// point.
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bool prerenderer_smoothing() {
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return !media_config.video.disable_prerenderer_smoothing;
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}
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void set_prerenderer_smoothing(bool enable) {
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media_config.video.disable_prerenderer_smoothing = !enable;
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}
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static const int kUndefined = -1;
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// Default maximum number of packets in the audio jitter buffer.
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static const int kAudioJitterBufferMaxPackets = 50;
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// TODO(pthatcher): Rename this ice_transport_type, but update
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// Chromium at the same time.
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IceTransportsType type = kAll;
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// TODO(pthatcher): Rename this ice_servers, but update Chromium
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// at the same time.
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IceServers servers;
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BundlePolicy bundle_policy = kBundlePolicyBalanced;
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RtcpMuxPolicy rtcp_mux_policy = kRtcpMuxPolicyNegotiate;
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TcpCandidatePolicy tcp_candidate_policy = kTcpCandidatePolicyEnabled;
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CandidateNetworkPolicy candidate_network_policy =
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kCandidateNetworkPolicyAll;
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int audio_jitter_buffer_max_packets = kAudioJitterBufferMaxPackets;
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bool audio_jitter_buffer_fast_accelerate = false;
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int ice_connection_receiving_timeout = kUndefined; // ms
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int ice_backup_candidate_pair_ping_interval = kUndefined; // ms
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ContinualGatheringPolicy continual_gathering_policy = GATHER_ONCE;
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std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
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bool prioritize_most_likely_ice_candidate_pairs = false;
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struct cricket::MediaConfig media_config;
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// Flags corresponding to values set by constraint flags.
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// rtc::Optional flags can be "missing", in which case the webrtc
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// default applies.
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bool disable_ipv6 = false;
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bool enable_rtp_data_channel = false;
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rtc::Optional<int> screencast_min_bitrate;
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rtc::Optional<bool> combined_audio_video_bwe;
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rtc::Optional<bool> enable_dtls_srtp;
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int ice_candidate_pool_size = 0;
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};
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struct RTCOfferAnswerOptions {
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static const int kUndefined = -1;
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static const int kMaxOfferToReceiveMedia = 1;
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// The default value for constraint offerToReceiveX:true.
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static const int kOfferToReceiveMediaTrue = 1;
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int offer_to_receive_video;
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int offer_to_receive_audio;
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bool voice_activity_detection;
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bool ice_restart;
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bool use_rtp_mux;
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RTCOfferAnswerOptions()
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: offer_to_receive_video(kUndefined),
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offer_to_receive_audio(kUndefined),
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voice_activity_detection(true),
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ice_restart(false),
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use_rtp_mux(true) {}
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RTCOfferAnswerOptions(int offer_to_receive_video,
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int offer_to_receive_audio,
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bool voice_activity_detection,
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bool ice_restart,
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bool use_rtp_mux)
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: offer_to_receive_video(offer_to_receive_video),
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offer_to_receive_audio(offer_to_receive_audio),
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voice_activity_detection(voice_activity_detection),
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ice_restart(ice_restart),
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use_rtp_mux(use_rtp_mux) {}
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};
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// Used by GetStats to decide which stats to include in the stats reports.
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// |kStatsOutputLevelStandard| includes the standard stats for Javascript API;
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// |kStatsOutputLevelDebug| includes both the standard stats and additional
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// stats for debugging purposes.
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enum StatsOutputLevel {
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kStatsOutputLevelStandard,
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kStatsOutputLevelDebug,
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};
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// Accessor methods to active local streams.
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virtual rtc::scoped_refptr<StreamCollectionInterface>
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local_streams() = 0;
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// Accessor methods to remote streams.
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virtual rtc::scoped_refptr<StreamCollectionInterface>
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remote_streams() = 0;
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// Add a new MediaStream to be sent on this PeerConnection.
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// Note that a SessionDescription negotiation is needed before the
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// remote peer can receive the stream.
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virtual bool AddStream(MediaStreamInterface* stream) = 0;
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// Remove a MediaStream from this PeerConnection.
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// Note that a SessionDescription negotiation is need before the
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||
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// remote peer is notified.
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virtual void RemoveStream(MediaStreamInterface* stream) = 0;
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||
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// TODO(deadbeef): Make the following two methods pure virtual once
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||
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// implemented by all subclasses of PeerConnectionInterface.
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||
|
|
// Add a new MediaStreamTrack to be sent on this PeerConnection.
|
||
|
|
// |streams| indicates which stream labels the track should be associated
|
||
|
|
// with.
|
||
|
|
virtual rtc::scoped_refptr<RtpSenderInterface> AddTrack(
|
||
|
|
MediaStreamTrackInterface* track,
|
||
|
|
std::vector<MediaStreamInterface*> streams) {
|
||
|
|
return nullptr;
|
||
|
|
}
|
||
|
|
|
||
|
|
// Remove an RtpSender from this PeerConnection.
|
||
|
|
// Returns true on success.
|
||
|
|
virtual bool RemoveTrack(RtpSenderInterface* sender) {
|
||
|
|
return false;
|
||
|
|
}
|
||
|
|
|
||
|
|
// Returns pointer to the created DtmfSender on success.
|
||
|
|
// Otherwise returns NULL.
|
||
|
|
virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
|
||
|
|
AudioTrackInterface* track) = 0;
|
||
|
|
|
||
|
|
// TODO(deadbeef): Make these pure virtual once all subclasses implement them.
|
||
|
|
// |kind| must be "audio" or "video".
|
||
|
|
// |stream_id| is used to populate the msid attribute; if empty, one will
|
||
|
|
// be generated automatically.
|
||
|
|
virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
|
||
|
|
const std::string& kind,
|
||
|
|
const std::string& stream_id) {
|
||
|
|
return rtc::scoped_refptr<RtpSenderInterface>();
|
||
|
|
}
|
||
|
|
|
||
|
|
virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
|
||
|
|
const {
|
||
|
|
return std::vector<rtc::scoped_refptr<RtpSenderInterface>>();
|
||
|
|
}
|
||
|
|
|
||
|
|
virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
|
||
|
|
const {
|
||
|
|
return std::vector<rtc::scoped_refptr<RtpReceiverInterface>>();
|
||
|
|
}
|
||
|
|
|
||
|
|
virtual bool GetStats(StatsObserver* observer,
|
||
|
|
MediaStreamTrackInterface* track,
|
||
|
|
StatsOutputLevel level) = 0;
|
||
|
|
|
||
|
|
virtual rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
|
||
|
|
const std::string& label,
|
||
|
|
const DataChannelInit* config) = 0;
|
||
|
|
|
||
|
|
virtual const SessionDescriptionInterface* local_description() const = 0;
|
||
|
|
virtual const SessionDescriptionInterface* remote_description() const = 0;
|
||
|
|
|
||
|
|
// Create a new offer.
|
||
|
|
// The CreateSessionDescriptionObserver callback will be called when done.
|
||
|
|
virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
|
||
|
|
const MediaConstraintsInterface* constraints) {}
|
||
|
|
|
||
|
|
// TODO(jiayl): remove the default impl and the old interface when chromium
|
||
|
|
// code is updated.
|
||
|
|
virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
|
||
|
|
const RTCOfferAnswerOptions& options) {}
|
||
|
|
|
||
|
|
// Create an answer to an offer.
|
||
|
|
// The CreateSessionDescriptionObserver callback will be called when done.
|
||
|
|
virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
|
||
|
|
const RTCOfferAnswerOptions& options) {}
|
||
|
|
// Deprecated - use version above.
|
||
|
|
// TODO(hta): Remove and remove default implementations when all callers
|
||
|
|
// are updated.
|
||
|
|
virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
|
||
|
|
const MediaConstraintsInterface* constraints) {}
|
||
|
|
|
||
|
|
// Sets the local session description.
|
||
|
|
// JsepInterface takes the ownership of |desc| even if it fails.
|
||
|
|
// The |observer| callback will be called when done.
|
||
|
|
virtual void SetLocalDescription(SetSessionDescriptionObserver* observer,
|
||
|
|
SessionDescriptionInterface* desc) = 0;
|
||
|
|
// Sets the remote session description.
|
||
|
|
// JsepInterface takes the ownership of |desc| even if it fails.
|
||
|
|
// The |observer| callback will be called when done.
|
||
|
|
virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
|
||
|
|
SessionDescriptionInterface* desc) = 0;
|
||
|
|
// Restarts or updates the ICE Agent process of gathering local candidates
|
||
|
|
// and pinging remote candidates.
|
||
|
|
// TODO(deadbeef): Remove once Chrome is moved over to SetConfiguration.
|
||
|
|
virtual bool UpdateIce(const IceServers& configuration,
|
||
|
|
const MediaConstraintsInterface* constraints) {
|
||
|
|
return false;
|
||
|
|
}
|
||
|
|
virtual bool UpdateIce(const IceServers& configuration) { return false; }
|
||
|
|
// Sets the PeerConnection's global configuration to |config|.
|
||
|
|
// Any changes to STUN/TURN servers or ICE candidate policy will affect the
|
||
|
|
// next gathering phase, and cause the next call to createOffer to generate
|
||
|
|
// new ICE credentials. Note that the BUNDLE and RTCP-multiplexing policies
|
||
|
|
// cannot be changed with this method.
|
||
|
|
// TODO(deadbeef): Make this pure virtual once all Chrome subclasses of
|
||
|
|
// PeerConnectionInterface implement it.
|
||
|
|
virtual bool SetConfiguration(
|
||
|
|
const PeerConnectionInterface::RTCConfiguration& config) {
|
||
|
|
return false;
|
||
|
|
}
|
||
|
|
// Provides a remote candidate to the ICE Agent.
|
||
|
|
// A copy of the |candidate| will be created and added to the remote
|
||
|
|
// description. So the caller of this method still has the ownership of the
|
||
|
|
// |candidate|.
|
||
|
|
// TODO(ronghuawu): Consider to change this so that the AddIceCandidate will
|
||
|
|
// take the ownership of the |candidate|.
|
||
|
|
virtual bool AddIceCandidate(const IceCandidateInterface* candidate) = 0;
|
||
|
|
|
||
|
|
// Removes a group of remote candidates from the ICE agent.
|
||
|
|
virtual bool RemoveIceCandidates(
|
||
|
|
const std::vector<cricket::Candidate>& candidates) {
|
||
|
|
return false;
|
||
|
|
}
|
||
|
|
|
||
|
|
virtual void RegisterUMAObserver(UMAObserver* observer) = 0;
|
||
|
|
|
||
|
|
// Returns the current SignalingState.
|
||
|
|
virtual SignalingState signaling_state() = 0;
|
||
|
|
|
||
|
|
// TODO(bemasc): Remove ice_state when callers are changed to
|
||
|
|
// IceConnection/GatheringState.
|
||
|
|
// Returns the current IceState.
|
||
|
|
virtual IceState ice_state() = 0;
|
||
|
|
virtual IceConnectionState ice_connection_state() = 0;
|
||
|
|
virtual IceGatheringState ice_gathering_state() = 0;
|
||
|
|
|
||
|
|
// Terminates all media and closes the transport.
|
||
|
|
virtual void Close() = 0;
|
||
|
|
|
||
|
|
protected:
|
||
|
|
// Dtor protected as objects shouldn't be deleted via this interface.
|
||
|
|
~PeerConnectionInterface() {}
|
||
|
|
};
|
||
|
|
|
||
|
|
// PeerConnection callback interface. Application should implement these
|
||
|
|
// methods.
|
||
|
|
class PeerConnectionObserver {
|
||
|
|
public:
|
||
|
|
enum StateType {
|
||
|
|
kSignalingState,
|
||
|
|
kIceState,
|
||
|
|
};
|
||
|
|
|
||
|
|
// Triggered when the SignalingState changed.
|
||
|
|
virtual void OnSignalingChange(
|
||
|
|
PeerConnectionInterface::SignalingState new_state) = 0;
|
||
|
|
|
||
|
|
// TODO(deadbeef): Once all subclasses override the scoped_refptr versions
|
||
|
|
// of the below three methods, make them pure virtual and remove the raw
|
||
|
|
// pointer version.
|
||
|
|
|
||
|
|
// Triggered when media is received on a new stream from remote peer.
|
||
|
|
virtual void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) {}
|
||
|
|
// Deprecated; please use the version that uses a scoped_refptr.
|
||
|
|
virtual void OnAddStream(MediaStreamInterface* stream) {}
|
||
|
|
|
||
|
|
// Triggered when a remote peer close a stream.
|
||
|
|
virtual void OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface> stream) {
|
||
|
|
}
|
||
|
|
// Deprecated; please use the version that uses a scoped_refptr.
|
||
|
|
virtual void OnRemoveStream(MediaStreamInterface* stream) {}
|
||
|
|
|
||
|
|
// Triggered when a remote peer opens a data channel.
|
||
|
|
virtual void OnDataChannel(
|
||
|
|
rtc::scoped_refptr<DataChannelInterface> data_channel){};
|
||
|
|
// Deprecated; please use the version that uses a scoped_refptr.
|
||
|
|
virtual void OnDataChannel(DataChannelInterface* data_channel) {}
|
||
|
|
|
||
|
|
// Triggered when renegotiation is needed. For example, an ICE restart
|
||
|
|
// has begun.
|
||
|
|
virtual void OnRenegotiationNeeded() = 0;
|
||
|
|
|
||
|
|
// Called any time the IceConnectionState changes.
|
||
|
|
virtual void OnIceConnectionChange(
|
||
|
|
PeerConnectionInterface::IceConnectionState new_state) = 0;
|
||
|
|
|
||
|
|
// Called any time the IceGatheringState changes.
|
||
|
|
virtual void OnIceGatheringChange(
|
||
|
|
PeerConnectionInterface::IceGatheringState new_state) = 0;
|
||
|
|
|
||
|
|
// A new ICE candidate has been gathered.
|
||
|
|
virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
|
||
|
|
|
||
|
|
// Ice candidates have been removed.
|
||
|
|
// TODO(honghaiz): Make this a pure virtual method when all its subclasses
|
||
|
|
// implement it.
|
||
|
|
virtual void OnIceCandidatesRemoved(
|
||
|
|
const std::vector<cricket::Candidate>& candidates) {}
|
||
|
|
|
||
|
|
// Called when the ICE connection receiving status changes.
|
||
|
|
virtual void OnIceConnectionReceivingChange(bool receiving) {}
|
||
|
|
|
||
|
|
protected:
|
||
|
|
// Dtor protected as objects shouldn't be deleted via this interface.
|
||
|
|
~PeerConnectionObserver() {}
|
||
|
|
};
|
||
|
|
|
||
|
|
// PeerConnectionFactoryInterface is the factory interface use for creating
|
||
|
|
// PeerConnection, MediaStream and media tracks.
|
||
|
|
// PeerConnectionFactoryInterface will create required libjingle threads,
|
||
|
|
// socket and network manager factory classes for networking.
|
||
|
|
// If an application decides to provide its own threads and network
|
||
|
|
// implementation of these classes it should use the alternate
|
||
|
|
// CreatePeerConnectionFactory method which accepts threads as input and use the
|
||
|
|
// CreatePeerConnection version that takes a PortAllocator as an
|
||
|
|
// argument.
|
||
|
|
class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
|
||
|
|
public:
|
||
|
|
class Options {
|
||
|
|
public:
|
||
|
|
Options()
|
||
|
|
: disable_encryption(false),
|
||
|
|
disable_sctp_data_channels(false),
|
||
|
|
disable_network_monitor(false),
|
||
|
|
network_ignore_mask(rtc::kDefaultNetworkIgnoreMask),
|
||
|
|
ssl_max_version(rtc::SSL_PROTOCOL_DTLS_12) {}
|
||
|
|
bool disable_encryption;
|
||
|
|
bool disable_sctp_data_channels;
|
||
|
|
bool disable_network_monitor;
|
||
|
|
|
||
|
|
// Sets the network types to ignore. For instance, calling this with
|
||
|
|
// ADAPTER_TYPE_ETHERNET | ADAPTER_TYPE_LOOPBACK will ignore Ethernet and
|
||
|
|
// loopback interfaces.
|
||
|
|
int network_ignore_mask;
|
||
|
|
|
||
|
|
// Sets the maximum supported protocol version. The highest version
|
||
|
|
// supported by both ends will be used for the connection, i.e. if one
|
||
|
|
// party supports DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used.
|
||
|
|
rtc::SSLProtocolVersion ssl_max_version;
|
||
|
|
};
|
||
|
|
|
||
|
|
virtual void SetOptions(const Options& options) = 0;
|
||
|
|
|
||
|
|
virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
|
||
|
|
const PeerConnectionInterface::RTCConfiguration& configuration,
|
||
|
|
const MediaConstraintsInterface* constraints,
|
||
|
|
std::unique_ptr<cricket::PortAllocator> allocator,
|
||
|
|
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
|
||
|
|
PeerConnectionObserver* observer) = 0;
|
||
|
|
|
||
|
|
virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
|
||
|
|
const PeerConnectionInterface::RTCConfiguration& configuration,
|
||
|
|
std::unique_ptr<cricket::PortAllocator> allocator,
|
||
|
|
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
|
||
|
|
PeerConnectionObserver* observer) = 0;
|
||
|
|
|
||
|
|
virtual rtc::scoped_refptr<MediaStreamInterface>
|
||
|
|
CreateLocalMediaStream(const std::string& label) = 0;
|
||
|
|
|
||
|
|
// Creates a AudioSourceInterface.
|
||
|
|
// |constraints| decides audio processing settings but can be NULL.
|
||
|
|
virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
|
||
|
|
const cricket::AudioOptions& options) = 0;
|
||
|
|
// Deprecated - use version above.
|
||
|
|
virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
|
||
|
|
const MediaConstraintsInterface* constraints) = 0;
|
||
|
|
|
||
|
|
// Creates a VideoTrackSourceInterface. The new source take ownership of
|
||
|
|
// |capturer|.
|
||
|
|
virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
|
||
|
|
cricket::VideoCapturer* capturer) = 0;
|
||
|
|
// A video source creator that allows selection of resolution and frame rate.
|
||
|
|
// |constraints| decides video resolution and frame rate but can
|
||
|
|
// be NULL.
|
||
|
|
// In the NULL case, use the version above.
|
||
|
|
virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
|
||
|
|
cricket::VideoCapturer* capturer,
|
||
|
|
const MediaConstraintsInterface* constraints) = 0;
|
||
|
|
|
||
|
|
// Creates a new local VideoTrack. The same |source| can be used in several
|
||
|
|
// tracks.
|
||
|
|
virtual rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
|
||
|
|
const std::string& label,
|
||
|
|
VideoTrackSourceInterface* source) = 0;
|
||
|
|
|
||
|
|
// Creates an new AudioTrack. At the moment |source| can be NULL.
|
||
|
|
virtual rtc::scoped_refptr<AudioTrackInterface>
|
||
|
|
CreateAudioTrack(const std::string& label,
|
||
|
|
AudioSourceInterface* source) = 0;
|
||
|
|
|
||
|
|
// Starts AEC dump using existing file. Takes ownership of |file| and passes
|
||
|
|
// it on to VoiceEngine (via other objects) immediately, which will take
|
||
|
|
// the ownerhip. If the operation fails, the file will be closed.
|
||
|
|
// A maximum file size in bytes can be specified. When the file size limit is
|
||
|
|
// reached, logging is stopped automatically. If max_size_bytes is set to a
|
||
|
|
// value <= 0, no limit will be used, and logging will continue until the
|
||
|
|
// StopAecDump function is called.
|
||
|
|
virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0;
|
||
|
|
|
||
|
|
// Stops logging the AEC dump.
|
||
|
|
virtual void StopAecDump() = 0;
|
||
|
|
|
||
|
|
// Starts RtcEventLog using existing file. Takes ownership of |file| and
|
||
|
|
// passes it on to VoiceEngine, which will take the ownership. If the
|
||
|
|
// operation fails the file will be closed. The logging will stop
|
||
|
|
// automatically after 10 minutes have passed, or when the StopRtcEventLog
|
||
|
|
// function is called. A maximum filesize in bytes can be set, the logging
|
||
|
|
// will be stopped before exceeding this limit. If max_size_bytes is set to a
|
||
|
|
// value <= 0, no limit will be used.
|
||
|
|
// This function as well as the StopRtcEventLog don't really belong on this
|
||
|
|
// interface, this is a temporary solution until we move the logging object
|
||
|
|
// from inside voice engine to webrtc::Call, which will happen when the VoE
|
||
|
|
// restructuring effort is further along.
|
||
|
|
// TODO(ivoc): Move this into being:
|
||
|
|
// PeerConnection => MediaController => webrtc::Call.
|
||
|
|
virtual bool StartRtcEventLog(rtc::PlatformFile file,
|
||
|
|
int64_t max_size_bytes) = 0;
|
||
|
|
// Deprecated, use the version above.
|
||
|
|
virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0;
|
||
|
|
|
||
|
|
// Stops logging the RtcEventLog.
|
||
|
|
virtual void StopRtcEventLog() = 0;
|
||
|
|
|
||
|
|
protected:
|
||
|
|
// Dtor and ctor protected as objects shouldn't be created or deleted via
|
||
|
|
// this interface.
|
||
|
|
PeerConnectionFactoryInterface() {}
|
||
|
|
~PeerConnectionFactoryInterface() {} // NOLINT
|
||
|
|
};
|
||
|
|
|
||
|
|
// Create a new instance of PeerConnectionFactoryInterface.
|
||
|
|
//
|
||
|
|
// This method relies on the thread it's called on as the "signaling thread"
|
||
|
|
// for the PeerConnectionFactory it creates.
|
||
|
|
//
|
||
|
|
// As such, if the current thread is not already running an rtc::Thread message
|
||
|
|
// loop, an application using this method must eventually either call
|
||
|
|
// rtc::Thread::Current()->Run(), or call
|
||
|
|
// rtc::Thread::Current()->ProcessMessages() within the application's own
|
||
|
|
// message loop.
|
||
|
|
rtc::scoped_refptr<PeerConnectionFactoryInterface>
|
||
|
|
CreatePeerConnectionFactory();
|
||
|
|
|
||
|
|
// Create a new instance of PeerConnectionFactoryInterface.
|
||
|
|
//
|
||
|
|
// |network_thread|, |worker_thread| and |signaling_thread| are
|
||
|
|
// the only mandatory parameters.
|
||
|
|
//
|
||
|
|
// If non-null, ownership of |default_adm|, |encoder_factory| and
|
||
|
|
// |decoder_factory| are transferred to the returned factory.
|
||
|
|
rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
|
||
|
|
rtc::Thread* network_thread,
|
||
|
|
rtc::Thread* worker_thread,
|
||
|
|
rtc::Thread* signaling_thread,
|
||
|
|
AudioDeviceModule* default_adm,
|
||
|
|
cricket::WebRtcVideoEncoderFactory* encoder_factory,
|
||
|
|
cricket::WebRtcVideoDecoderFactory* decoder_factory);
|
||
|
|
|
||
|
|
// Create a new instance of PeerConnectionFactoryInterface.
|
||
|
|
// Same thread is used as worker and network thread.
|
||
|
|
inline rtc::scoped_refptr<PeerConnectionFactoryInterface>
|
||
|
|
CreatePeerConnectionFactory(
|
||
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rtc::Thread* worker_and_network_thread,
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rtc::Thread* signaling_thread,
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AudioDeviceModule* default_adm,
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cricket::WebRtcVideoEncoderFactory* encoder_factory,
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cricket::WebRtcVideoDecoderFactory* decoder_factory) {
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return CreatePeerConnectionFactory(
|
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worker_and_network_thread, worker_and_network_thread, signaling_thread,
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default_adm, encoder_factory, decoder_factory);
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}
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} // namespace webrtc
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#endif // WEBRTC_API_PEERCONNECTIONINTERFACE_H_
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