b78306a7d3
RTCStatsCollector relies on PeerConnection and its WebRtcSession. If the PeerConnection is destroyed, reference counting keeps the RTCStatsCollector alive until the request has completed. But the request is using PeerConnection/WebRtcSession resources that are destroyed in ~PeerConnection(). To get around this problem, RTCStatsCollector::WaitForPendingRequest() is added, which is invoked at ~PeerConnection(). Integration test added, it caused a segmentation fault before this change / EXPECT failure. BUG=chromium:627816 Review-Url: https://codereview.webrtc.org/2583613003 Cr-Commit-Position: refs/heads/master@{#15674}
2420 lines
86 KiB
C++
2420 lines
86 KiB
C++
/*
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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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#include "webrtc/api/peerconnection.h"
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#include <algorithm>
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#include <cctype> // for isdigit
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#include <utility>
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#include <vector>
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#include "webrtc/api/audiotrack.h"
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#include "webrtc/api/dtmfsender.h"
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#include "webrtc/api/jsepicecandidate.h"
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#include "webrtc/api/jsepsessiondescription.h"
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#include "webrtc/api/mediaconstraintsinterface.h"
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#include "webrtc/api/mediastream.h"
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#include "webrtc/api/mediastreamobserver.h"
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#include "webrtc/api/mediastreamproxy.h"
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#include "webrtc/api/mediastreamtrackproxy.h"
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#include "webrtc/api/remoteaudiosource.h"
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#include "webrtc/api/rtpreceiver.h"
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#include "webrtc/api/rtpsender.h"
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#include "webrtc/api/streamcollection.h"
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#include "webrtc/api/videocapturertracksource.h"
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#include "webrtc/api/videotrack.h"
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#include "webrtc/base/arraysize.h"
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#include "webrtc/base/bind.h"
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#include "webrtc/base/logging.h"
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#include "webrtc/base/stringencode.h"
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#include "webrtc/base/stringutils.h"
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#include "webrtc/base/trace_event.h"
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#include "webrtc/call/call.h"
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#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
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#include "webrtc/media/sctp/sctpdataengine.h"
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#include "webrtc/pc/channelmanager.h"
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#include "webrtc/system_wrappers/include/clock.h"
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#include "webrtc/system_wrappers/include/field_trial.h"
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namespace {
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using webrtc::DataChannel;
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using webrtc::MediaConstraintsInterface;
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using webrtc::MediaStreamInterface;
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using webrtc::PeerConnectionInterface;
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using webrtc::RtpSenderInternal;
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using webrtc::RtpSenderInterface;
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using webrtc::RtpSenderProxy;
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using webrtc::RtpSenderProxyWithInternal;
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using webrtc::StreamCollection;
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static const char kDefaultStreamLabel[] = "default";
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static const char kDefaultAudioTrackLabel[] = "defaulta0";
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static const char kDefaultVideoTrackLabel[] = "defaultv0";
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// The min number of tokens must present in Turn host uri.
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// e.g. user@turn.example.org
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static const size_t kTurnHostTokensNum = 2;
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// Number of tokens must be preset when TURN uri has transport param.
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static const size_t kTurnTransportTokensNum = 2;
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// The default stun port.
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static const int kDefaultStunPort = 3478;
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static const int kDefaultStunTlsPort = 5349;
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static const char kTransport[] = "transport";
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// NOTE: Must be in the same order as the ServiceType enum.
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static const char* kValidIceServiceTypes[] = {"stun", "stuns", "turn", "turns"};
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// The length of RTCP CNAMEs.
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static const int kRtcpCnameLength = 16;
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// NOTE: A loop below assumes that the first value of this enum is 0 and all
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// other values are incremental.
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enum ServiceType {
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STUN = 0, // Indicates a STUN server.
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STUNS, // Indicates a STUN server used with a TLS session.
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TURN, // Indicates a TURN server
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TURNS, // Indicates a TURN server used with a TLS session.
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INVALID, // Unknown.
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};
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static_assert(INVALID == arraysize(kValidIceServiceTypes),
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"kValidIceServiceTypes must have as many strings as ServiceType "
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"has values.");
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enum {
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MSG_SET_SESSIONDESCRIPTION_SUCCESS = 0,
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MSG_SET_SESSIONDESCRIPTION_FAILED,
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MSG_CREATE_SESSIONDESCRIPTION_FAILED,
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MSG_GETSTATS,
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MSG_FREE_DATACHANNELS,
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};
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struct SetSessionDescriptionMsg : public rtc::MessageData {
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explicit SetSessionDescriptionMsg(
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webrtc::SetSessionDescriptionObserver* observer)
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: observer(observer) {
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}
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rtc::scoped_refptr<webrtc::SetSessionDescriptionObserver> observer;
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std::string error;
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};
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struct CreateSessionDescriptionMsg : public rtc::MessageData {
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explicit CreateSessionDescriptionMsg(
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webrtc::CreateSessionDescriptionObserver* observer)
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: observer(observer) {}
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rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserver> observer;
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std::string error;
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};
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struct GetStatsMsg : public rtc::MessageData {
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GetStatsMsg(webrtc::StatsObserver* observer,
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webrtc::MediaStreamTrackInterface* track)
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: observer(observer), track(track) {
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}
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rtc::scoped_refptr<webrtc::StatsObserver> observer;
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rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track;
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};
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// |in_str| should be of format
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// stunURI = scheme ":" stun-host [ ":" stun-port ]
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// scheme = "stun" / "stuns"
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// stun-host = IP-literal / IPv4address / reg-name
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// stun-port = *DIGIT
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//
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// draft-petithuguenin-behave-turn-uris-01
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// turnURI = scheme ":" turn-host [ ":" turn-port ]
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// turn-host = username@IP-literal / IPv4address / reg-name
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bool GetServiceTypeAndHostnameFromUri(const std::string& in_str,
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ServiceType* service_type,
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std::string* hostname) {
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const std::string::size_type colonpos = in_str.find(':');
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if (colonpos == std::string::npos) {
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LOG(LS_WARNING) << "Missing ':' in ICE URI: " << in_str;
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return false;
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}
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if ((colonpos + 1) == in_str.length()) {
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LOG(LS_WARNING) << "Empty hostname in ICE URI: " << in_str;
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return false;
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}
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*service_type = INVALID;
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for (size_t i = 0; i < arraysize(kValidIceServiceTypes); ++i) {
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if (in_str.compare(0, colonpos, kValidIceServiceTypes[i]) == 0) {
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*service_type = static_cast<ServiceType>(i);
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break;
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}
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}
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if (*service_type == INVALID) {
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return false;
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}
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*hostname = in_str.substr(colonpos + 1, std::string::npos);
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return true;
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}
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bool ParsePort(const std::string& in_str, int* port) {
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// Make sure port only contains digits. FromString doesn't check this.
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for (const char& c : in_str) {
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if (!std::isdigit(c)) {
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return false;
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}
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}
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return rtc::FromString(in_str, port);
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}
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// This method parses IPv6 and IPv4 literal strings, along with hostnames in
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// standard hostname:port format.
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// Consider following formats as correct.
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// |hostname:port|, |[IPV6 address]:port|, |IPv4 address|:port,
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// |hostname|, |[IPv6 address]|, |IPv4 address|.
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bool ParseHostnameAndPortFromString(const std::string& in_str,
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std::string* host,
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int* port) {
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RTC_DCHECK(host->empty());
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if (in_str.at(0) == '[') {
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std::string::size_type closebracket = in_str.rfind(']');
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if (closebracket != std::string::npos) {
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std::string::size_type colonpos = in_str.find(':', closebracket);
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if (std::string::npos != colonpos) {
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if (!ParsePort(in_str.substr(closebracket + 2, std::string::npos),
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port)) {
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return false;
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}
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}
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*host = in_str.substr(1, closebracket - 1);
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} else {
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return false;
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}
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} else {
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std::string::size_type colonpos = in_str.find(':');
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if (std::string::npos != colonpos) {
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if (!ParsePort(in_str.substr(colonpos + 1, std::string::npos), port)) {
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return false;
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}
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*host = in_str.substr(0, colonpos);
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} else {
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*host = in_str;
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}
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}
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return !host->empty();
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}
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// Adds a STUN or TURN server to the appropriate list,
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// by parsing |url| and using the username/password in |server|.
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bool ParseIceServerUrl(const PeerConnectionInterface::IceServer& server,
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const std::string& url,
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cricket::ServerAddresses* stun_servers,
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std::vector<cricket::RelayServerConfig>* turn_servers) {
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// draft-nandakumar-rtcweb-stun-uri-01
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// stunURI = scheme ":" stun-host [ ":" stun-port ]
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// scheme = "stun" / "stuns"
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// stun-host = IP-literal / IPv4address / reg-name
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// stun-port = *DIGIT
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// draft-petithuguenin-behave-turn-uris-01
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// turnURI = scheme ":" turn-host [ ":" turn-port ]
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// [ "?transport=" transport ]
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// scheme = "turn" / "turns"
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// transport = "udp" / "tcp" / transport-ext
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// transport-ext = 1*unreserved
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// turn-host = IP-literal / IPv4address / reg-name
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// turn-port = *DIGIT
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RTC_DCHECK(stun_servers != nullptr);
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RTC_DCHECK(turn_servers != nullptr);
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std::vector<std::string> tokens;
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cricket::ProtocolType turn_transport_type = cricket::PROTO_UDP;
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RTC_DCHECK(!url.empty());
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rtc::tokenize_with_empty_tokens(url, '?', &tokens);
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std::string uri_without_transport = tokens[0];
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// Let's look into transport= param, if it exists.
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if (tokens.size() == kTurnTransportTokensNum) { // ?transport= is present.
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std::string uri_transport_param = tokens[1];
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rtc::tokenize_with_empty_tokens(uri_transport_param, '=', &tokens);
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if (tokens[0] != kTransport) {
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LOG(LS_WARNING) << "Invalid transport parameter key.";
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return false;
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}
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if (tokens.size() < 2 ||
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!cricket::StringToProto(tokens[1].c_str(), &turn_transport_type) ||
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(turn_transport_type != cricket::PROTO_UDP &&
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turn_transport_type != cricket::PROTO_TCP)) {
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LOG(LS_WARNING) << "Transport param should always be udp or tcp.";
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return false;
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}
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}
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std::string hoststring;
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ServiceType service_type;
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if (!GetServiceTypeAndHostnameFromUri(uri_without_transport,
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&service_type,
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&hoststring)) {
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LOG(LS_WARNING) << "Invalid transport parameter in ICE URI: " << url;
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return false;
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}
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// GetServiceTypeAndHostnameFromUri should never give an empty hoststring
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RTC_DCHECK(!hoststring.empty());
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// Let's break hostname.
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tokens.clear();
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rtc::tokenize_with_empty_tokens(hoststring, '@', &tokens);
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std::string username(server.username);
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if (tokens.size() > kTurnHostTokensNum) {
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LOG(LS_WARNING) << "Invalid user@hostname format: " << hoststring;
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return false;
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}
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if (tokens.size() == kTurnHostTokensNum) {
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if (tokens[0].empty() || tokens[1].empty()) {
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LOG(LS_WARNING) << "Invalid user@hostname format: " << hoststring;
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return false;
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}
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username.assign(rtc::s_url_decode(tokens[0]));
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hoststring = tokens[1];
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} else {
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hoststring = tokens[0];
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}
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int port = kDefaultStunPort;
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if (service_type == TURNS) {
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port = kDefaultStunTlsPort;
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turn_transport_type = cricket::PROTO_TLS;
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}
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std::string address;
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if (!ParseHostnameAndPortFromString(hoststring, &address, &port)) {
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LOG(WARNING) << "Invalid hostname format: " << uri_without_transport;
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return false;
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}
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if (port <= 0 || port > 0xffff) {
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LOG(WARNING) << "Invalid port: " << port;
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return false;
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}
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switch (service_type) {
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case STUN:
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case STUNS:
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stun_servers->insert(rtc::SocketAddress(address, port));
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break;
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case TURN:
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case TURNS: {
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cricket::RelayServerConfig config = cricket::RelayServerConfig(
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address, port, username, server.password, turn_transport_type);
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if (server.tls_cert_policy ==
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PeerConnectionInterface::kTlsCertPolicyInsecureNoCheck) {
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config.tls_cert_policy =
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cricket::TlsCertPolicy::TLS_CERT_POLICY_INSECURE_NO_CHECK;
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}
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turn_servers->push_back(config);
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break;
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}
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case INVALID:
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default:
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LOG(WARNING) << "Configuration not supported: " << url;
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return false;
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}
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return true;
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}
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// Check if we can send |new_stream| on a PeerConnection.
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bool CanAddLocalMediaStream(webrtc::StreamCollectionInterface* current_streams,
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webrtc::MediaStreamInterface* new_stream) {
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if (!new_stream || !current_streams) {
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return false;
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}
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if (current_streams->find(new_stream->label()) != nullptr) {
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LOG(LS_ERROR) << "MediaStream with label " << new_stream->label()
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<< " is already added.";
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return false;
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}
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return true;
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}
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bool MediaContentDirectionHasSend(cricket::MediaContentDirection dir) {
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return dir == cricket::MD_SENDONLY || dir == cricket::MD_SENDRECV;
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}
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// If the direction is "recvonly" or "inactive", treat the description
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// as containing no streams.
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// See: https://code.google.com/p/webrtc/issues/detail?id=5054
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std::vector<cricket::StreamParams> GetActiveStreams(
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const cricket::MediaContentDescription* desc) {
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return MediaContentDirectionHasSend(desc->direction())
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? desc->streams()
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: std::vector<cricket::StreamParams>();
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}
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bool IsValidOfferToReceiveMedia(int value) {
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typedef PeerConnectionInterface::RTCOfferAnswerOptions Options;
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return (value >= Options::kUndefined) &&
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(value <= Options::kMaxOfferToReceiveMedia);
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}
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// Add the stream and RTP data channel info to |session_options|.
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void AddSendStreams(
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cricket::MediaSessionOptions* session_options,
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const std::vector<rtc::scoped_refptr<
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RtpSenderProxyWithInternal<RtpSenderInternal>>>& senders,
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const std::map<std::string, rtc::scoped_refptr<DataChannel>>&
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rtp_data_channels) {
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session_options->streams.clear();
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for (const auto& sender : senders) {
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session_options->AddSendStream(sender->media_type(), sender->id(),
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sender->internal()->stream_id());
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}
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// Check for data channels.
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for (const auto& kv : rtp_data_channels) {
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const DataChannel* channel = kv.second;
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if (channel->state() == DataChannel::kConnecting ||
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channel->state() == DataChannel::kOpen) {
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// |streamid| and |sync_label| are both set to the DataChannel label
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// here so they can be signaled the same way as MediaStreams and Tracks.
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// For MediaStreams, the sync_label is the MediaStream label and the
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// track label is the same as |streamid|.
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const std::string& streamid = channel->label();
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const std::string& sync_label = channel->label();
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session_options->AddSendStream(cricket::MEDIA_TYPE_DATA, streamid,
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sync_label);
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}
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}
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}
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uint32_t ConvertIceTransportTypeToCandidateFilter(
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PeerConnectionInterface::IceTransportsType type) {
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switch (type) {
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case PeerConnectionInterface::kNone:
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return cricket::CF_NONE;
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case PeerConnectionInterface::kRelay:
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return cricket::CF_RELAY;
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case PeerConnectionInterface::kNoHost:
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return (cricket::CF_ALL & ~cricket::CF_HOST);
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case PeerConnectionInterface::kAll:
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return cricket::CF_ALL;
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default:
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ASSERT(false);
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}
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return cricket::CF_NONE;
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}
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|
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// Helper method to set a voice/video channel on all applicable senders
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// and receivers when one is created/destroyed by WebRtcSession.
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//
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// Used by On(Voice|Video)Channel(Created|Destroyed)
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template <class SENDER,
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class RECEIVER,
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class CHANNEL,
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class SENDERS,
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class RECEIVERS>
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void SetChannelOnSendersAndReceivers(CHANNEL* channel,
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SENDERS& senders,
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RECEIVERS& receivers,
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cricket::MediaType media_type) {
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for (auto& sender : senders) {
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if (sender->media_type() == media_type) {
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static_cast<SENDER*>(sender->internal())->SetChannel(channel);
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}
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}
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for (auto& receiver : receivers) {
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if (receiver->media_type() == media_type) {
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if (!channel) {
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receiver->internal()->Stop();
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}
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static_cast<RECEIVER*>(receiver->internal())->SetChannel(channel);
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}
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}
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}
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|
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} // namespace
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|
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namespace webrtc {
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|
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static const char* const kRtcErrorNames[] = {
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"NONE",
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"UNSUPPORTED_PARAMETER",
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"INVALID_PARAMETER",
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"INVALID_RANGE",
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"SYNTAX_ERROR",
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"INVALID_STATE",
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"INVALID_MODIFICATION",
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"NETWORK_ERROR",
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"INTERNAL_ERROR",
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};
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std::ostream& operator<<(std::ostream& stream, RtcError error) {
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int index = static_cast<int>(error);
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RTC_CHECK(index < static_cast<int>(sizeof(kRtcErrorNames) /
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sizeof(kRtcErrorNames[0])));
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return stream << kRtcErrorNames[index];
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}
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|
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// Generate a RTCP CNAME when a PeerConnection is created.
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std::string GenerateRtcpCname() {
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std::string cname;
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if (!rtc::CreateRandomString(kRtcpCnameLength, &cname)) {
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LOG(LS_ERROR) << "Failed to generate CNAME.";
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RTC_DCHECK(false);
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}
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return cname;
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}
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|
|
bool ExtractMediaSessionOptions(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
|
|
bool is_offer,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
typedef PeerConnectionInterface::RTCOfferAnswerOptions RTCOfferAnswerOptions;
|
|
if (!IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_audio) ||
|
|
!IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_video)) {
|
|
return false;
|
|
}
|
|
|
|
// If constraints don't prevent us, we always accept video.
|
|
if (rtc_options.offer_to_receive_audio != RTCOfferAnswerOptions::kUndefined) {
|
|
session_options->recv_audio = (rtc_options.offer_to_receive_audio > 0);
|
|
} else {
|
|
session_options->recv_audio = true;
|
|
}
|
|
// For offers, we only offer video if we have it or it's forced by options.
|
|
// For answers, we will always accept video (if offered).
|
|
if (rtc_options.offer_to_receive_video != RTCOfferAnswerOptions::kUndefined) {
|
|
session_options->recv_video = (rtc_options.offer_to_receive_video > 0);
|
|
} else if (is_offer) {
|
|
session_options->recv_video = false;
|
|
} else {
|
|
session_options->recv_video = true;
|
|
}
|
|
|
|
session_options->vad_enabled = rtc_options.voice_activity_detection;
|
|
session_options->bundle_enabled = rtc_options.use_rtp_mux;
|
|
for (auto& kv : session_options->transport_options) {
|
|
kv.second.ice_restart = rtc_options.ice_restart;
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
bool ParseConstraintsForAnswer(const MediaConstraintsInterface* constraints,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
bool value = false;
|
|
size_t mandatory_constraints_satisfied = 0;
|
|
|
|
// kOfferToReceiveAudio defaults to true according to spec.
|
|
if (!FindConstraint(constraints,
|
|
MediaConstraintsInterface::kOfferToReceiveAudio, &value,
|
|
&mandatory_constraints_satisfied) ||
|
|
value) {
|
|
session_options->recv_audio = true;
|
|
}
|
|
|
|
// kOfferToReceiveVideo defaults to false according to spec. But
|
|
// if it is an answer and video is offered, we should still accept video
|
|
// per default.
|
|
value = false;
|
|
if (!FindConstraint(constraints,
|
|
MediaConstraintsInterface::kOfferToReceiveVideo, &value,
|
|
&mandatory_constraints_satisfied) ||
|
|
value) {
|
|
session_options->recv_video = true;
|
|
}
|
|
|
|
if (FindConstraint(constraints,
|
|
MediaConstraintsInterface::kVoiceActivityDetection, &value,
|
|
&mandatory_constraints_satisfied)) {
|
|
session_options->vad_enabled = value;
|
|
}
|
|
|
|
if (FindConstraint(constraints, MediaConstraintsInterface::kUseRtpMux, &value,
|
|
&mandatory_constraints_satisfied)) {
|
|
session_options->bundle_enabled = value;
|
|
} else {
|
|
// kUseRtpMux defaults to true according to spec.
|
|
session_options->bundle_enabled = true;
|
|
}
|
|
|
|
bool ice_restart = false;
|
|
if (FindConstraint(constraints, MediaConstraintsInterface::kIceRestart,
|
|
&value, &mandatory_constraints_satisfied)) {
|
|
// kIceRestart defaults to false according to spec.
|
|
ice_restart = true;
|
|
}
|
|
for (auto& kv : session_options->transport_options) {
|
|
kv.second.ice_restart = ice_restart;
|
|
}
|
|
|
|
if (!constraints) {
|
|
return true;
|
|
}
|
|
return mandatory_constraints_satisfied == constraints->GetMandatory().size();
|
|
}
|
|
|
|
bool ParseIceServers(const PeerConnectionInterface::IceServers& servers,
|
|
cricket::ServerAddresses* stun_servers,
|
|
std::vector<cricket::RelayServerConfig>* turn_servers) {
|
|
for (const webrtc::PeerConnectionInterface::IceServer& server : servers) {
|
|
if (!server.urls.empty()) {
|
|
for (const std::string& url : server.urls) {
|
|
if (url.empty()) {
|
|
LOG(LS_ERROR) << "Empty uri.";
|
|
return false;
|
|
}
|
|
if (!ParseIceServerUrl(server, url, stun_servers, turn_servers)) {
|
|
return false;
|
|
}
|
|
}
|
|
} else if (!server.uri.empty()) {
|
|
// Fallback to old .uri if new .urls isn't present.
|
|
if (!ParseIceServerUrl(server, server.uri, stun_servers, turn_servers)) {
|
|
return false;
|
|
}
|
|
} else {
|
|
LOG(LS_ERROR) << "Empty uri.";
|
|
return false;
|
|
}
|
|
}
|
|
// Candidates must have unique priorities, so that connectivity checks
|
|
// are performed in a well-defined order.
|
|
int priority = static_cast<int>(turn_servers->size() - 1);
|
|
for (cricket::RelayServerConfig& turn_server : *turn_servers) {
|
|
// First in the list gets highest priority.
|
|
turn_server.priority = priority--;
|
|
}
|
|
return true;
|
|
}
|
|
|
|
PeerConnection::PeerConnection(PeerConnectionFactory* factory)
|
|
: factory_(factory),
|
|
observer_(NULL),
|
|
uma_observer_(NULL),
|
|
signaling_state_(kStable),
|
|
ice_connection_state_(kIceConnectionNew),
|
|
ice_gathering_state_(kIceGatheringNew),
|
|
event_log_(RtcEventLog::Create(webrtc::Clock::GetRealTimeClock())),
|
|
rtcp_cname_(GenerateRtcpCname()),
|
|
local_streams_(StreamCollection::Create()),
|
|
remote_streams_(StreamCollection::Create()) {}
|
|
|
|
PeerConnection::~PeerConnection() {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
// Need to detach RTP senders/receivers from WebRtcSession,
|
|
// since it's about to be destroyed.
|
|
for (const auto& sender : senders_) {
|
|
sender->internal()->Stop();
|
|
}
|
|
for (const auto& receiver : receivers_) {
|
|
receiver->internal()->Stop();
|
|
}
|
|
// Destroy stats_ because it depends on session_.
|
|
stats_.reset(nullptr);
|
|
if (stats_collector_) {
|
|
stats_collector_->WaitForPendingRequest();
|
|
stats_collector_ = nullptr;
|
|
}
|
|
// Now destroy session_ before destroying other members,
|
|
// because its destruction fires signals (such as VoiceChannelDestroyed)
|
|
// which will trigger some final actions in PeerConnection...
|
|
session_.reset(nullptr);
|
|
// port_allocator_ lives on the network thread and should be destroyed there.
|
|
network_thread()->Invoke<void>(RTC_FROM_HERE,
|
|
[this] { port_allocator_.reset(nullptr); });
|
|
}
|
|
|
|
bool PeerConnection::Initialize(
|
|
const PeerConnectionInterface::RTCConfiguration& configuration,
|
|
std::unique_ptr<cricket::PortAllocator> allocator,
|
|
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
|
|
PeerConnectionObserver* observer) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::Initialize");
|
|
RTC_DCHECK(observer != nullptr);
|
|
if (!observer) {
|
|
return false;
|
|
}
|
|
observer_ = observer;
|
|
|
|
port_allocator_ = std::move(allocator);
|
|
|
|
// The port allocator lives on the network thread and should be initialized
|
|
// there.
|
|
if (!network_thread()->Invoke<bool>(
|
|
RTC_FROM_HERE, rtc::Bind(&PeerConnection::InitializePortAllocator_n,
|
|
this, configuration))) {
|
|
return false;
|
|
}
|
|
|
|
media_controller_.reset(factory_->CreateMediaController(
|
|
configuration.media_config, event_log_.get()));
|
|
|
|
session_.reset(new WebRtcSession(
|
|
media_controller_.get(), factory_->network_thread(),
|
|
factory_->worker_thread(), factory_->signaling_thread(),
|
|
port_allocator_.get(),
|
|
std::unique_ptr<cricket::TransportController>(
|
|
factory_->CreateTransportController(
|
|
port_allocator_.get(),
|
|
configuration.redetermine_role_on_ice_restart))));
|
|
|
|
stats_.reset(new StatsCollector(this));
|
|
stats_collector_ = RTCStatsCollector::Create(this);
|
|
|
|
// Initialize the WebRtcSession. It creates transport channels etc.
|
|
if (!session_->Initialize(factory_->options(), std::move(cert_generator),
|
|
configuration)) {
|
|
return false;
|
|
}
|
|
|
|
// Register PeerConnection as receiver of local ice candidates.
|
|
// All the callbacks will be posted to the application from PeerConnection.
|
|
session_->RegisterIceObserver(this);
|
|
session_->SignalState.connect(this, &PeerConnection::OnSessionStateChange);
|
|
session_->SignalVoiceChannelCreated.connect(
|
|
this, &PeerConnection::OnVoiceChannelCreated);
|
|
session_->SignalVoiceChannelDestroyed.connect(
|
|
this, &PeerConnection::OnVoiceChannelDestroyed);
|
|
session_->SignalVideoChannelCreated.connect(
|
|
this, &PeerConnection::OnVideoChannelCreated);
|
|
session_->SignalVideoChannelDestroyed.connect(
|
|
this, &PeerConnection::OnVideoChannelDestroyed);
|
|
session_->SignalDataChannelCreated.connect(
|
|
this, &PeerConnection::OnDataChannelCreated);
|
|
session_->SignalDataChannelDestroyed.connect(
|
|
this, &PeerConnection::OnDataChannelDestroyed);
|
|
session_->SignalDataChannelOpenMessage.connect(
|
|
this, &PeerConnection::OnDataChannelOpenMessage);
|
|
|
|
configuration_ = configuration;
|
|
return true;
|
|
}
|
|
|
|
rtc::scoped_refptr<StreamCollectionInterface>
|
|
PeerConnection::local_streams() {
|
|
return local_streams_;
|
|
}
|
|
|
|
rtc::scoped_refptr<StreamCollectionInterface>
|
|
PeerConnection::remote_streams() {
|
|
return remote_streams_;
|
|
}
|
|
|
|
bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::AddStream");
|
|
if (IsClosed()) {
|
|
return false;
|
|
}
|
|
if (!CanAddLocalMediaStream(local_streams_, local_stream)) {
|
|
return false;
|
|
}
|
|
|
|
local_streams_->AddStream(local_stream);
|
|
MediaStreamObserver* observer = new MediaStreamObserver(local_stream);
|
|
observer->SignalAudioTrackAdded.connect(this,
|
|
&PeerConnection::OnAudioTrackAdded);
|
|
observer->SignalAudioTrackRemoved.connect(
|
|
this, &PeerConnection::OnAudioTrackRemoved);
|
|
observer->SignalVideoTrackAdded.connect(this,
|
|
&PeerConnection::OnVideoTrackAdded);
|
|
observer->SignalVideoTrackRemoved.connect(
|
|
this, &PeerConnection::OnVideoTrackRemoved);
|
|
stream_observers_.push_back(std::unique_ptr<MediaStreamObserver>(observer));
|
|
|
|
for (const auto& track : local_stream->GetAudioTracks()) {
|
|
OnAudioTrackAdded(track.get(), local_stream);
|
|
}
|
|
for (const auto& track : local_stream->GetVideoTracks()) {
|
|
OnVideoTrackAdded(track.get(), local_stream);
|
|
}
|
|
|
|
stats_->AddStream(local_stream);
|
|
observer_->OnRenegotiationNeeded();
|
|
return true;
|
|
}
|
|
|
|
void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::RemoveStream");
|
|
for (const auto& track : local_stream->GetAudioTracks()) {
|
|
OnAudioTrackRemoved(track.get(), local_stream);
|
|
}
|
|
for (const auto& track : local_stream->GetVideoTracks()) {
|
|
OnVideoTrackRemoved(track.get(), local_stream);
|
|
}
|
|
|
|
local_streams_->RemoveStream(local_stream);
|
|
stream_observers_.erase(
|
|
std::remove_if(
|
|
stream_observers_.begin(), stream_observers_.end(),
|
|
[local_stream](const std::unique_ptr<MediaStreamObserver>& observer) {
|
|
return observer->stream()->label().compare(local_stream->label()) ==
|
|
0;
|
|
}),
|
|
stream_observers_.end());
|
|
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
observer_->OnRenegotiationNeeded();
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpSenderInterface> PeerConnection::AddTrack(
|
|
MediaStreamTrackInterface* track,
|
|
std::vector<MediaStreamInterface*> streams) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::AddTrack");
|
|
if (IsClosed()) {
|
|
return nullptr;
|
|
}
|
|
if (streams.size() >= 2) {
|
|
LOG(LS_ERROR)
|
|
<< "Adding a track with two streams is not currently supported.";
|
|
return nullptr;
|
|
}
|
|
// TODO(deadbeef): Support adding a track to two different senders.
|
|
if (FindSenderForTrack(track) != senders_.end()) {
|
|
LOG(LS_ERROR) << "Sender for track " << track->id() << " already exists.";
|
|
return nullptr;
|
|
}
|
|
|
|
// TODO(deadbeef): Support adding a track to multiple streams.
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender;
|
|
if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
|
|
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(),
|
|
new AudioRtpSender(static_cast<AudioTrackInterface*>(track),
|
|
session_->voice_channel(), stats_.get()));
|
|
if (!streams.empty()) {
|
|
new_sender->internal()->set_stream_id(streams[0]->label());
|
|
}
|
|
const TrackInfo* track_info = FindTrackInfo(
|
|
local_audio_tracks_, new_sender->internal()->stream_id(), track->id());
|
|
if (track_info) {
|
|
new_sender->internal()->SetSsrc(track_info->ssrc);
|
|
}
|
|
} else if (track->kind() == MediaStreamTrackInterface::kVideoKind) {
|
|
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(),
|
|
new VideoRtpSender(static_cast<VideoTrackInterface*>(track),
|
|
session_->video_channel()));
|
|
if (!streams.empty()) {
|
|
new_sender->internal()->set_stream_id(streams[0]->label());
|
|
}
|
|
const TrackInfo* track_info = FindTrackInfo(
|
|
local_video_tracks_, new_sender->internal()->stream_id(), track->id());
|
|
if (track_info) {
|
|
new_sender->internal()->SetSsrc(track_info->ssrc);
|
|
}
|
|
} else {
|
|
LOG(LS_ERROR) << "CreateSender called with invalid kind: " << track->kind();
|
|
return rtc::scoped_refptr<RtpSenderInterface>();
|
|
}
|
|
|
|
senders_.push_back(new_sender);
|
|
observer_->OnRenegotiationNeeded();
|
|
return new_sender;
|
|
}
|
|
|
|
bool PeerConnection::RemoveTrack(RtpSenderInterface* sender) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::RemoveTrack");
|
|
if (IsClosed()) {
|
|
return false;
|
|
}
|
|
|
|
auto it = std::find(senders_.begin(), senders_.end(), sender);
|
|
if (it == senders_.end()) {
|
|
LOG(LS_ERROR) << "Couldn't find sender " << sender->id() << " to remove.";
|
|
return false;
|
|
}
|
|
(*it)->internal()->Stop();
|
|
senders_.erase(it);
|
|
|
|
observer_->OnRenegotiationNeeded();
|
|
return true;
|
|
}
|
|
|
|
rtc::scoped_refptr<DtmfSenderInterface> PeerConnection::CreateDtmfSender(
|
|
AudioTrackInterface* track) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::CreateDtmfSender");
|
|
if (IsClosed()) {
|
|
return nullptr;
|
|
}
|
|
if (!track) {
|
|
LOG(LS_ERROR) << "CreateDtmfSender - track is NULL.";
|
|
return NULL;
|
|
}
|
|
if (!local_streams_->FindAudioTrack(track->id())) {
|
|
LOG(LS_ERROR) << "CreateDtmfSender is called with a non local audio track.";
|
|
return NULL;
|
|
}
|
|
|
|
rtc::scoped_refptr<DtmfSenderInterface> sender(
|
|
DtmfSender::Create(track, signaling_thread(), session_.get()));
|
|
if (!sender.get()) {
|
|
LOG(LS_ERROR) << "CreateDtmfSender failed on DtmfSender::Create.";
|
|
return NULL;
|
|
}
|
|
return DtmfSenderProxy::Create(signaling_thread(), sender.get());
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
|
|
const std::string& kind,
|
|
const std::string& stream_id) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
|
|
if (IsClosed()) {
|
|
return nullptr;
|
|
}
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender;
|
|
if (kind == MediaStreamTrackInterface::kAudioKind) {
|
|
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(),
|
|
new AudioRtpSender(session_->voice_channel(), stats_.get()));
|
|
} else if (kind == MediaStreamTrackInterface::kVideoKind) {
|
|
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(), new VideoRtpSender(session_->video_channel()));
|
|
} else {
|
|
LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
|
|
return new_sender;
|
|
}
|
|
if (!stream_id.empty()) {
|
|
new_sender->internal()->set_stream_id(stream_id);
|
|
}
|
|
senders_.push_back(new_sender);
|
|
return new_sender;
|
|
}
|
|
|
|
std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
|
|
const {
|
|
std::vector<rtc::scoped_refptr<RtpSenderInterface>> ret;
|
|
for (const auto& sender : senders_) {
|
|
ret.push_back(sender.get());
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
std::vector<rtc::scoped_refptr<RtpReceiverInterface>>
|
|
PeerConnection::GetReceivers() const {
|
|
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> ret;
|
|
for (const auto& receiver : receivers_) {
|
|
ret.push_back(receiver.get());
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
bool PeerConnection::GetStats(StatsObserver* observer,
|
|
MediaStreamTrackInterface* track,
|
|
StatsOutputLevel level) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
if (!VERIFY(observer != NULL)) {
|
|
LOG(LS_ERROR) << "GetStats - observer is NULL.";
|
|
return false;
|
|
}
|
|
|
|
stats_->UpdateStats(level);
|
|
// The StatsCollector is used to tell if a track is valid because it may
|
|
// remember tracks that the PeerConnection previously removed.
|
|
if (track && !stats_->IsValidTrack(track->id())) {
|
|
LOG(LS_WARNING) << "GetStats is called with an invalid track: "
|
|
<< track->id();
|
|
return false;
|
|
}
|
|
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_GETSTATS,
|
|
new GetStatsMsg(observer, track));
|
|
return true;
|
|
}
|
|
|
|
void PeerConnection::GetStats(RTCStatsCollectorCallback* callback) {
|
|
RTC_DCHECK(stats_collector_);
|
|
stats_collector_->GetStatsReport(callback);
|
|
}
|
|
|
|
PeerConnectionInterface::SignalingState PeerConnection::signaling_state() {
|
|
return signaling_state_;
|
|
}
|
|
|
|
PeerConnectionInterface::IceConnectionState
|
|
PeerConnection::ice_connection_state() {
|
|
return ice_connection_state_;
|
|
}
|
|
|
|
PeerConnectionInterface::IceGatheringState
|
|
PeerConnection::ice_gathering_state() {
|
|
return ice_gathering_state_;
|
|
}
|
|
|
|
rtc::scoped_refptr<DataChannelInterface>
|
|
PeerConnection::CreateDataChannel(
|
|
const std::string& label,
|
|
const DataChannelInit* config) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::CreateDataChannel");
|
|
#ifdef HAVE_QUIC
|
|
if (session_->data_channel_type() == cricket::DCT_QUIC) {
|
|
// TODO(zhihuang): Handle case when config is NULL.
|
|
if (!config) {
|
|
LOG(LS_ERROR) << "Missing config for QUIC data channel.";
|
|
return nullptr;
|
|
}
|
|
// TODO(zhihuang): Allow unreliable or ordered QUIC data channels.
|
|
if (!config->reliable || config->ordered) {
|
|
LOG(LS_ERROR) << "QUIC data channel does not implement unreliable or "
|
|
"ordered delivery.";
|
|
return nullptr;
|
|
}
|
|
return session_->quic_data_transport()->CreateDataChannel(label, config);
|
|
}
|
|
#endif // HAVE_QUIC
|
|
|
|
bool first_datachannel = !HasDataChannels();
|
|
|
|
std::unique_ptr<InternalDataChannelInit> internal_config;
|
|
if (config) {
|
|
internal_config.reset(new InternalDataChannelInit(*config));
|
|
}
|
|
rtc::scoped_refptr<DataChannelInterface> channel(
|
|
InternalCreateDataChannel(label, internal_config.get()));
|
|
if (!channel.get()) {
|
|
return nullptr;
|
|
}
|
|
|
|
// Trigger the onRenegotiationNeeded event for every new RTP DataChannel, or
|
|
// the first SCTP DataChannel.
|
|
if (session_->data_channel_type() == cricket::DCT_RTP || first_datachannel) {
|
|
observer_->OnRenegotiationNeeded();
|
|
}
|
|
|
|
return DataChannelProxy::Create(signaling_thread(), channel.get());
|
|
}
|
|
|
|
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
|
|
const MediaConstraintsInterface* constraints) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
|
|
if (!VERIFY(observer != nullptr)) {
|
|
LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
|
|
return;
|
|
}
|
|
RTCOfferAnswerOptions options;
|
|
|
|
bool value;
|
|
size_t mandatory_constraints = 0;
|
|
|
|
if (FindConstraint(constraints,
|
|
MediaConstraintsInterface::kOfferToReceiveAudio,
|
|
&value,
|
|
&mandatory_constraints)) {
|
|
options.offer_to_receive_audio =
|
|
value ? RTCOfferAnswerOptions::kOfferToReceiveMediaTrue : 0;
|
|
}
|
|
|
|
if (FindConstraint(constraints,
|
|
MediaConstraintsInterface::kOfferToReceiveVideo,
|
|
&value,
|
|
&mandatory_constraints)) {
|
|
options.offer_to_receive_video =
|
|
value ? RTCOfferAnswerOptions::kOfferToReceiveMediaTrue : 0;
|
|
}
|
|
|
|
if (FindConstraint(constraints,
|
|
MediaConstraintsInterface::kVoiceActivityDetection,
|
|
&value,
|
|
&mandatory_constraints)) {
|
|
options.voice_activity_detection = value;
|
|
}
|
|
|
|
if (FindConstraint(constraints,
|
|
MediaConstraintsInterface::kIceRestart,
|
|
&value,
|
|
&mandatory_constraints)) {
|
|
options.ice_restart = value;
|
|
}
|
|
|
|
if (FindConstraint(constraints,
|
|
MediaConstraintsInterface::kUseRtpMux,
|
|
&value,
|
|
&mandatory_constraints)) {
|
|
options.use_rtp_mux = value;
|
|
}
|
|
|
|
CreateOffer(observer, options);
|
|
}
|
|
|
|
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
|
|
const RTCOfferAnswerOptions& options) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
|
|
if (!VERIFY(observer != nullptr)) {
|
|
LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
|
|
return;
|
|
}
|
|
|
|
cricket::MediaSessionOptions session_options;
|
|
if (!GetOptionsForOffer(options, &session_options)) {
|
|
std::string error = "CreateOffer called with invalid options.";
|
|
LOG(LS_ERROR) << error;
|
|
PostCreateSessionDescriptionFailure(observer, error);
|
|
return;
|
|
}
|
|
|
|
session_->CreateOffer(observer, options, session_options);
|
|
}
|
|
|
|
void PeerConnection::CreateAnswer(
|
|
CreateSessionDescriptionObserver* observer,
|
|
const MediaConstraintsInterface* constraints) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::CreateAnswer");
|
|
if (!VERIFY(observer != nullptr)) {
|
|
LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
|
|
return;
|
|
}
|
|
|
|
cricket::MediaSessionOptions session_options;
|
|
if (!GetOptionsForAnswer(constraints, &session_options)) {
|
|
std::string error = "CreateAnswer called with invalid constraints.";
|
|
LOG(LS_ERROR) << error;
|
|
PostCreateSessionDescriptionFailure(observer, error);
|
|
return;
|
|
}
|
|
|
|
session_->CreateAnswer(observer, session_options);
|
|
}
|
|
|
|
void PeerConnection::CreateAnswer(CreateSessionDescriptionObserver* observer,
|
|
const RTCOfferAnswerOptions& options) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::CreateAnswer");
|
|
if (!VERIFY(observer != nullptr)) {
|
|
LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
|
|
return;
|
|
}
|
|
|
|
cricket::MediaSessionOptions session_options;
|
|
if (!GetOptionsForAnswer(options, &session_options)) {
|
|
std::string error = "CreateAnswer called with invalid options.";
|
|
LOG(LS_ERROR) << error;
|
|
PostCreateSessionDescriptionFailure(observer, error);
|
|
return;
|
|
}
|
|
|
|
session_->CreateAnswer(observer, session_options);
|
|
}
|
|
|
|
void PeerConnection::SetLocalDescription(
|
|
SetSessionDescriptionObserver* observer,
|
|
SessionDescriptionInterface* desc) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::SetLocalDescription");
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
if (!VERIFY(observer != nullptr)) {
|
|
LOG(LS_ERROR) << "SetLocalDescription - observer is NULL.";
|
|
return;
|
|
}
|
|
if (!desc) {
|
|
PostSetSessionDescriptionFailure(observer, "SessionDescription is NULL.");
|
|
return;
|
|
}
|
|
// Update stats here so that we have the most recent stats for tracks and
|
|
// streams that might be removed by updating the session description.
|
|
stats_->UpdateStats(kStatsOutputLevelStandard);
|
|
std::string error;
|
|
if (!session_->SetLocalDescription(desc, &error)) {
|
|
PostSetSessionDescriptionFailure(observer, error);
|
|
return;
|
|
}
|
|
|
|
// If setting the description decided our SSL role, allocate any necessary
|
|
// SCTP sids.
|
|
rtc::SSLRole role;
|
|
if (session_->data_channel_type() == cricket::DCT_SCTP &&
|
|
session_->GetSslRole(session_->data_channel(), &role)) {
|
|
AllocateSctpSids(role);
|
|
}
|
|
|
|
// Update state and SSRC of local MediaStreams and DataChannels based on the
|
|
// local session description.
|
|
const cricket::ContentInfo* audio_content =
|
|
GetFirstAudioContent(desc->description());
|
|
if (audio_content) {
|
|
if (audio_content->rejected) {
|
|
RemoveTracks(cricket::MEDIA_TYPE_AUDIO);
|
|
} else {
|
|
const cricket::AudioContentDescription* audio_desc =
|
|
static_cast<const cricket::AudioContentDescription*>(
|
|
audio_content->description);
|
|
UpdateLocalTracks(audio_desc->streams(), audio_desc->type());
|
|
}
|
|
}
|
|
|
|
const cricket::ContentInfo* video_content =
|
|
GetFirstVideoContent(desc->description());
|
|
if (video_content) {
|
|
if (video_content->rejected) {
|
|
RemoveTracks(cricket::MEDIA_TYPE_VIDEO);
|
|
} else {
|
|
const cricket::VideoContentDescription* video_desc =
|
|
static_cast<const cricket::VideoContentDescription*>(
|
|
video_content->description);
|
|
UpdateLocalTracks(video_desc->streams(), video_desc->type());
|
|
}
|
|
}
|
|
|
|
const cricket::ContentInfo* data_content =
|
|
GetFirstDataContent(desc->description());
|
|
if (data_content) {
|
|
const cricket::DataContentDescription* data_desc =
|
|
static_cast<const cricket::DataContentDescription*>(
|
|
data_content->description);
|
|
if (rtc::starts_with(data_desc->protocol().data(),
|
|
cricket::kMediaProtocolRtpPrefix)) {
|
|
UpdateLocalRtpDataChannels(data_desc->streams());
|
|
}
|
|
}
|
|
|
|
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
|
|
signaling_thread()->Post(RTC_FROM_HERE, this,
|
|
MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
|
|
|
|
// MaybeStartGathering needs to be called after posting
|
|
// MSG_SET_SESSIONDESCRIPTION_SUCCESS, so that we don't signal any candidates
|
|
// before signaling that SetLocalDescription completed.
|
|
session_->MaybeStartGathering();
|
|
}
|
|
|
|
void PeerConnection::SetRemoteDescription(
|
|
SetSessionDescriptionObserver* observer,
|
|
SessionDescriptionInterface* desc) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::SetRemoteDescription");
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
if (!VERIFY(observer != nullptr)) {
|
|
LOG(LS_ERROR) << "SetRemoteDescription - observer is NULL.";
|
|
return;
|
|
}
|
|
if (!desc) {
|
|
PostSetSessionDescriptionFailure(observer, "SessionDescription is NULL.");
|
|
return;
|
|
}
|
|
// Update stats here so that we have the most recent stats for tracks and
|
|
// streams that might be removed by updating the session description.
|
|
stats_->UpdateStats(kStatsOutputLevelStandard);
|
|
std::string error;
|
|
if (!session_->SetRemoteDescription(desc, &error)) {
|
|
PostSetSessionDescriptionFailure(observer, error);
|
|
return;
|
|
}
|
|
|
|
// If setting the description decided our SSL role, allocate any necessary
|
|
// SCTP sids.
|
|
rtc::SSLRole role;
|
|
if (session_->data_channel_type() == cricket::DCT_SCTP &&
|
|
session_->GetSslRole(session_->data_channel(), &role)) {
|
|
AllocateSctpSids(role);
|
|
}
|
|
|
|
const cricket::SessionDescription* remote_desc = desc->description();
|
|
const cricket::ContentInfo* audio_content = GetFirstAudioContent(remote_desc);
|
|
const cricket::ContentInfo* video_content = GetFirstVideoContent(remote_desc);
|
|
const cricket::AudioContentDescription* audio_desc =
|
|
GetFirstAudioContentDescription(remote_desc);
|
|
const cricket::VideoContentDescription* video_desc =
|
|
GetFirstVideoContentDescription(remote_desc);
|
|
const cricket::DataContentDescription* data_desc =
|
|
GetFirstDataContentDescription(remote_desc);
|
|
|
|
// Check if the descriptions include streams, just in case the peer supports
|
|
// MSID, but doesn't indicate so with "a=msid-semantic".
|
|
if (remote_desc->msid_supported() ||
|
|
(audio_desc && !audio_desc->streams().empty()) ||
|
|
(video_desc && !video_desc->streams().empty())) {
|
|
remote_peer_supports_msid_ = true;
|
|
}
|
|
|
|
// We wait to signal new streams until we finish processing the description,
|
|
// since only at that point will new streams have all their tracks.
|
|
rtc::scoped_refptr<StreamCollection> new_streams(StreamCollection::Create());
|
|
|
|
// Find all audio rtp streams and create corresponding remote AudioTracks
|
|
// and MediaStreams.
|
|
if (audio_content) {
|
|
if (audio_content->rejected) {
|
|
RemoveTracks(cricket::MEDIA_TYPE_AUDIO);
|
|
} else {
|
|
bool default_audio_track_needed =
|
|
!remote_peer_supports_msid_ &&
|
|
MediaContentDirectionHasSend(audio_desc->direction());
|
|
UpdateRemoteStreamsList(GetActiveStreams(audio_desc),
|
|
default_audio_track_needed, audio_desc->type(),
|
|
new_streams);
|
|
}
|
|
}
|
|
|
|
// Find all video rtp streams and create corresponding remote VideoTracks
|
|
// and MediaStreams.
|
|
if (video_content) {
|
|
if (video_content->rejected) {
|
|
RemoveTracks(cricket::MEDIA_TYPE_VIDEO);
|
|
} else {
|
|
bool default_video_track_needed =
|
|
!remote_peer_supports_msid_ &&
|
|
MediaContentDirectionHasSend(video_desc->direction());
|
|
UpdateRemoteStreamsList(GetActiveStreams(video_desc),
|
|
default_video_track_needed, video_desc->type(),
|
|
new_streams);
|
|
}
|
|
}
|
|
|
|
// Update the DataChannels with the information from the remote peer.
|
|
if (data_desc) {
|
|
if (rtc::starts_with(data_desc->protocol().data(),
|
|
cricket::kMediaProtocolRtpPrefix)) {
|
|
UpdateRemoteRtpDataChannels(GetActiveStreams(data_desc));
|
|
}
|
|
}
|
|
|
|
// Iterate new_streams and notify the observer about new MediaStreams.
|
|
for (size_t i = 0; i < new_streams->count(); ++i) {
|
|
MediaStreamInterface* new_stream = new_streams->at(i);
|
|
stats_->AddStream(new_stream);
|
|
// Call both the raw pointer and scoped_refptr versions of the method
|
|
// for compatibility.
|
|
observer_->OnAddStream(new_stream);
|
|
observer_->OnAddStream(
|
|
rtc::scoped_refptr<MediaStreamInterface>(new_stream));
|
|
}
|
|
|
|
UpdateEndedRemoteMediaStreams();
|
|
|
|
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
|
|
signaling_thread()->Post(RTC_FROM_HERE, this,
|
|
MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
|
|
}
|
|
|
|
PeerConnectionInterface::RTCConfiguration PeerConnection::GetConfiguration() {
|
|
return configuration_;
|
|
}
|
|
|
|
bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
|
|
|
|
if (session_->local_description() &&
|
|
configuration.ice_candidate_pool_size !=
|
|
configuration_.ice_candidate_pool_size) {
|
|
LOG(LS_ERROR) << "Can't change candidate pool size after calling "
|
|
"SetLocalDescription.";
|
|
return false;
|
|
}
|
|
// TODO(deadbeef): Return false and log an error if there are any unsupported
|
|
// modifications.
|
|
if (port_allocator_) {
|
|
if (!network_thread()->Invoke<bool>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&PeerConnection::ReconfigurePortAllocator_n, this,
|
|
configuration))) {
|
|
LOG(LS_ERROR) << "Failed to apply configuration to PortAllocator.";
|
|
return false;
|
|
}
|
|
}
|
|
|
|
// TODO(deadbeef): Shouldn't have to hop to the network thread twice...
|
|
session_->SetIceConfig(session_->ParseIceConfig(configuration));
|
|
|
|
// As described in JSEP, calling setConfiguration with new ICE servers or
|
|
// candidate policy must set a "needs-ice-restart" bit so that the next offer
|
|
// triggers an ICE restart which will pick up the changes.
|
|
if (configuration.servers != configuration_.servers ||
|
|
configuration.type != configuration_.type) {
|
|
session_->SetNeedsIceRestartFlag();
|
|
}
|
|
configuration_ = configuration;
|
|
return true;
|
|
}
|
|
|
|
bool PeerConnection::AddIceCandidate(
|
|
const IceCandidateInterface* ice_candidate) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::AddIceCandidate");
|
|
if (IsClosed()) {
|
|
return false;
|
|
}
|
|
return session_->ProcessIceMessage(ice_candidate);
|
|
}
|
|
|
|
bool PeerConnection::RemoveIceCandidates(
|
|
const std::vector<cricket::Candidate>& candidates) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::RemoveIceCandidates");
|
|
return session_->RemoveRemoteIceCandidates(candidates);
|
|
}
|
|
|
|
void PeerConnection::RegisterUMAObserver(UMAObserver* observer) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::RegisterUmaObserver");
|
|
uma_observer_ = observer;
|
|
|
|
if (session_) {
|
|
session_->set_metrics_observer(uma_observer_);
|
|
}
|
|
|
|
// Send information about IPv4/IPv6 status.
|
|
if (uma_observer_ && port_allocator_) {
|
|
port_allocator_->SetMetricsObserver(uma_observer_);
|
|
if (port_allocator_->flags() & cricket::PORTALLOCATOR_ENABLE_IPV6) {
|
|
uma_observer_->IncrementEnumCounter(
|
|
kEnumCounterAddressFamily, kPeerConnection_IPv6,
|
|
kPeerConnectionAddressFamilyCounter_Max);
|
|
} else {
|
|
uma_observer_->IncrementEnumCounter(
|
|
kEnumCounterAddressFamily, kPeerConnection_IPv4,
|
|
kPeerConnectionAddressFamilyCounter_Max);
|
|
}
|
|
}
|
|
}
|
|
|
|
bool PeerConnection::StartRtcEventLog(rtc::PlatformFile file,
|
|
int64_t max_size_bytes) {
|
|
return factory_->worker_thread()->Invoke<bool>(
|
|
RTC_FROM_HERE, rtc::Bind(&PeerConnection::StartRtcEventLog_w, this, file,
|
|
max_size_bytes));
|
|
}
|
|
|
|
void PeerConnection::StopRtcEventLog() {
|
|
factory_->worker_thread()->Invoke<void>(
|
|
RTC_FROM_HERE, rtc::Bind(&PeerConnection::StopRtcEventLog_w, this));
|
|
}
|
|
|
|
const SessionDescriptionInterface* PeerConnection::local_description() const {
|
|
return session_->local_description();
|
|
}
|
|
|
|
const SessionDescriptionInterface* PeerConnection::remote_description() const {
|
|
return session_->remote_description();
|
|
}
|
|
|
|
void PeerConnection::Close() {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::Close");
|
|
// Update stats here so that we have the most recent stats for tracks and
|
|
// streams before the channels are closed.
|
|
stats_->UpdateStats(kStatsOutputLevelStandard);
|
|
|
|
session_->Close();
|
|
}
|
|
|
|
void PeerConnection::OnSessionStateChange(WebRtcSession* /*session*/,
|
|
WebRtcSession::State state) {
|
|
switch (state) {
|
|
case WebRtcSession::STATE_INIT:
|
|
ChangeSignalingState(PeerConnectionInterface::kStable);
|
|
break;
|
|
case WebRtcSession::STATE_SENTOFFER:
|
|
ChangeSignalingState(PeerConnectionInterface::kHaveLocalOffer);
|
|
break;
|
|
case WebRtcSession::STATE_SENTPRANSWER:
|
|
ChangeSignalingState(PeerConnectionInterface::kHaveLocalPrAnswer);
|
|
break;
|
|
case WebRtcSession::STATE_RECEIVEDOFFER:
|
|
ChangeSignalingState(PeerConnectionInterface::kHaveRemoteOffer);
|
|
break;
|
|
case WebRtcSession::STATE_RECEIVEDPRANSWER:
|
|
ChangeSignalingState(PeerConnectionInterface::kHaveRemotePrAnswer);
|
|
break;
|
|
case WebRtcSession::STATE_INPROGRESS:
|
|
ChangeSignalingState(PeerConnectionInterface::kStable);
|
|
break;
|
|
case WebRtcSession::STATE_CLOSED:
|
|
ChangeSignalingState(PeerConnectionInterface::kClosed);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnMessage(rtc::Message* msg) {
|
|
switch (msg->message_id) {
|
|
case MSG_SET_SESSIONDESCRIPTION_SUCCESS: {
|
|
SetSessionDescriptionMsg* param =
|
|
static_cast<SetSessionDescriptionMsg*>(msg->pdata);
|
|
param->observer->OnSuccess();
|
|
delete param;
|
|
break;
|
|
}
|
|
case MSG_SET_SESSIONDESCRIPTION_FAILED: {
|
|
SetSessionDescriptionMsg* param =
|
|
static_cast<SetSessionDescriptionMsg*>(msg->pdata);
|
|
param->observer->OnFailure(param->error);
|
|
delete param;
|
|
break;
|
|
}
|
|
case MSG_CREATE_SESSIONDESCRIPTION_FAILED: {
|
|
CreateSessionDescriptionMsg* param =
|
|
static_cast<CreateSessionDescriptionMsg*>(msg->pdata);
|
|
param->observer->OnFailure(param->error);
|
|
delete param;
|
|
break;
|
|
}
|
|
case MSG_GETSTATS: {
|
|
GetStatsMsg* param = static_cast<GetStatsMsg*>(msg->pdata);
|
|
StatsReports reports;
|
|
stats_->GetStats(param->track, &reports);
|
|
param->observer->OnComplete(reports);
|
|
delete param;
|
|
break;
|
|
}
|
|
case MSG_FREE_DATACHANNELS: {
|
|
sctp_data_channels_to_free_.clear();
|
|
break;
|
|
}
|
|
default:
|
|
RTC_DCHECK(false && "Not implemented");
|
|
break;
|
|
}
|
|
}
|
|
|
|
void PeerConnection::CreateAudioReceiver(MediaStreamInterface* stream,
|
|
const std::string& track_id,
|
|
uint32_t ssrc) {
|
|
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
|
|
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
|
|
signaling_thread(), new AudioRtpReceiver(stream, track_id, ssrc,
|
|
session_->voice_channel()));
|
|
|
|
receivers_.push_back(receiver);
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams;
|
|
streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
|
|
observer_->OnAddTrack(receiver, streams);
|
|
}
|
|
|
|
void PeerConnection::CreateVideoReceiver(MediaStreamInterface* stream,
|
|
const std::string& track_id,
|
|
uint32_t ssrc) {
|
|
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
|
|
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
|
|
signaling_thread(),
|
|
new VideoRtpReceiver(stream, track_id, factory_->worker_thread(),
|
|
ssrc, session_->video_channel()));
|
|
receivers_.push_back(receiver);
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams;
|
|
streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
|
|
observer_->OnAddTrack(receiver, streams);
|
|
}
|
|
|
|
// TODO(deadbeef): Keep RtpReceivers around even if track goes away in remote
|
|
// description.
|
|
void PeerConnection::DestroyReceiver(const std::string& track_id) {
|
|
auto it = FindReceiverForTrack(track_id);
|
|
if (it == receivers_.end()) {
|
|
LOG(LS_WARNING) << "RtpReceiver for track with id " << track_id
|
|
<< " doesn't exist.";
|
|
} else {
|
|
(*it)->internal()->Stop();
|
|
receivers_.erase(it);
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnIceConnectionChange(
|
|
PeerConnectionInterface::IceConnectionState new_state) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
// After transitioning to "closed", ignore any additional states from
|
|
// WebRtcSession (such as "disconnected").
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
ice_connection_state_ = new_state;
|
|
observer_->OnIceConnectionChange(ice_connection_state_);
|
|
}
|
|
|
|
void PeerConnection::OnIceGatheringChange(
|
|
PeerConnectionInterface::IceGatheringState new_state) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
ice_gathering_state_ = new_state;
|
|
observer_->OnIceGatheringChange(ice_gathering_state_);
|
|
}
|
|
|
|
void PeerConnection::OnIceCandidate(const IceCandidateInterface* candidate) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
observer_->OnIceCandidate(candidate);
|
|
}
|
|
|
|
void PeerConnection::OnIceCandidatesRemoved(
|
|
const std::vector<cricket::Candidate>& candidates) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
observer_->OnIceCandidatesRemoved(candidates);
|
|
}
|
|
|
|
void PeerConnection::OnIceConnectionReceivingChange(bool receiving) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
observer_->OnIceConnectionReceivingChange(receiving);
|
|
}
|
|
|
|
void PeerConnection::ChangeSignalingState(
|
|
PeerConnectionInterface::SignalingState signaling_state) {
|
|
signaling_state_ = signaling_state;
|
|
if (signaling_state == kClosed) {
|
|
ice_connection_state_ = kIceConnectionClosed;
|
|
observer_->OnIceConnectionChange(ice_connection_state_);
|
|
if (ice_gathering_state_ != kIceGatheringComplete) {
|
|
ice_gathering_state_ = kIceGatheringComplete;
|
|
observer_->OnIceGatheringChange(ice_gathering_state_);
|
|
}
|
|
}
|
|
observer_->OnSignalingChange(signaling_state_);
|
|
}
|
|
|
|
void PeerConnection::OnAudioTrackAdded(AudioTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
auto sender = FindSenderForTrack(track);
|
|
if (sender != senders_.end()) {
|
|
// We already have a sender for this track, so just change the stream_id
|
|
// so that it's correct in the next call to CreateOffer.
|
|
(*sender)->internal()->set_stream_id(stream->label());
|
|
return;
|
|
}
|
|
|
|
// Normal case; we've never seen this track before.
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender =
|
|
RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(),
|
|
new AudioRtpSender(track, stream->label(), session_->voice_channel(),
|
|
stats_.get()));
|
|
senders_.push_back(new_sender);
|
|
// If the sender has already been configured in SDP, we call SetSsrc,
|
|
// which will connect the sender to the underlying transport. This can
|
|
// occur if a local session description that contains the ID of the sender
|
|
// is set before AddStream is called. It can also occur if the local
|
|
// session description is not changed and RemoveStream is called, and
|
|
// later AddStream is called again with the same stream.
|
|
const TrackInfo* track_info =
|
|
FindTrackInfo(local_audio_tracks_, stream->label(), track->id());
|
|
if (track_info) {
|
|
new_sender->internal()->SetSsrc(track_info->ssrc);
|
|
}
|
|
}
|
|
|
|
// TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
|
|
// indefinitely, when we have unified plan SDP.
|
|
void PeerConnection::OnAudioTrackRemoved(AudioTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
auto sender = FindSenderForTrack(track);
|
|
if (sender == senders_.end()) {
|
|
LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
|
|
<< " doesn't exist.";
|
|
return;
|
|
}
|
|
(*sender)->internal()->Stop();
|
|
senders_.erase(sender);
|
|
}
|
|
|
|
void PeerConnection::OnVideoTrackAdded(VideoTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
auto sender = FindSenderForTrack(track);
|
|
if (sender != senders_.end()) {
|
|
// We already have a sender for this track, so just change the stream_id
|
|
// so that it's correct in the next call to CreateOffer.
|
|
(*sender)->internal()->set_stream_id(stream->label());
|
|
return;
|
|
}
|
|
|
|
// Normal case; we've never seen this track before.
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender =
|
|
RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(), new VideoRtpSender(track, stream->label(),
|
|
session_->video_channel()));
|
|
senders_.push_back(new_sender);
|
|
const TrackInfo* track_info =
|
|
FindTrackInfo(local_video_tracks_, stream->label(), track->id());
|
|
if (track_info) {
|
|
new_sender->internal()->SetSsrc(track_info->ssrc);
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnVideoTrackRemoved(VideoTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
auto sender = FindSenderForTrack(track);
|
|
if (sender == senders_.end()) {
|
|
LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
|
|
<< " doesn't exist.";
|
|
return;
|
|
}
|
|
(*sender)->internal()->Stop();
|
|
senders_.erase(sender);
|
|
}
|
|
|
|
void PeerConnection::PostSetSessionDescriptionFailure(
|
|
SetSessionDescriptionObserver* observer,
|
|
const std::string& error) {
|
|
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
|
|
msg->error = error;
|
|
signaling_thread()->Post(RTC_FROM_HERE, this,
|
|
MSG_SET_SESSIONDESCRIPTION_FAILED, msg);
|
|
}
|
|
|
|
void PeerConnection::PostCreateSessionDescriptionFailure(
|
|
CreateSessionDescriptionObserver* observer,
|
|
const std::string& error) {
|
|
CreateSessionDescriptionMsg* msg = new CreateSessionDescriptionMsg(observer);
|
|
msg->error = error;
|
|
signaling_thread()->Post(RTC_FROM_HERE, this,
|
|
MSG_CREATE_SESSIONDESCRIPTION_FAILED, msg);
|
|
}
|
|
|
|
bool PeerConnection::GetOptionsForOffer(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
// TODO(deadbeef): Once we have transceivers, enumerate them here instead of
|
|
// ContentInfos.
|
|
if (session_->local_description()) {
|
|
for (const cricket::ContentInfo& content :
|
|
session_->local_description()->description()->contents()) {
|
|
session_options->transport_options[content.name] =
|
|
cricket::TransportOptions();
|
|
}
|
|
}
|
|
session_options->enable_ice_renomination =
|
|
configuration_.enable_ice_renomination;
|
|
|
|
if (!ExtractMediaSessionOptions(rtc_options, true, session_options)) {
|
|
return false;
|
|
}
|
|
|
|
AddSendStreams(session_options, senders_, rtp_data_channels_);
|
|
// Offer to receive audio/video if the constraint is not set and there are
|
|
// send streams, or we're currently receiving.
|
|
if (rtc_options.offer_to_receive_audio == RTCOfferAnswerOptions::kUndefined) {
|
|
session_options->recv_audio =
|
|
session_options->HasSendMediaStream(cricket::MEDIA_TYPE_AUDIO) ||
|
|
!remote_audio_tracks_.empty();
|
|
}
|
|
if (rtc_options.offer_to_receive_video == RTCOfferAnswerOptions::kUndefined) {
|
|
session_options->recv_video =
|
|
session_options->HasSendMediaStream(cricket::MEDIA_TYPE_VIDEO) ||
|
|
!remote_video_tracks_.empty();
|
|
}
|
|
|
|
// Intentionally unset the data channel type for RTP data channel with the
|
|
// second condition. Otherwise the RTP data channels would be successfully
|
|
// negotiated by default and the unit tests in WebRtcDataBrowserTest will fail
|
|
// when building with chromium. We want to leave RTP data channels broken, so
|
|
// people won't try to use them.
|
|
if (HasDataChannels() && session_->data_channel_type() != cricket::DCT_RTP) {
|
|
session_options->data_channel_type = session_->data_channel_type();
|
|
}
|
|
|
|
session_options->bundle_enabled =
|
|
session_options->bundle_enabled &&
|
|
(session_options->has_audio() || session_options->has_video() ||
|
|
session_options->has_data());
|
|
|
|
session_options->rtcp_cname = rtcp_cname_;
|
|
session_options->crypto_options = factory_->options().crypto_options;
|
|
return true;
|
|
}
|
|
|
|
void PeerConnection::InitializeOptionsForAnswer(
|
|
cricket::MediaSessionOptions* session_options) {
|
|
session_options->recv_audio = false;
|
|
session_options->recv_video = false;
|
|
session_options->enable_ice_renomination =
|
|
configuration_.enable_ice_renomination;
|
|
}
|
|
|
|
void PeerConnection::FinishOptionsForAnswer(
|
|
cricket::MediaSessionOptions* session_options) {
|
|
// TODO(deadbeef): Once we have transceivers, enumerate them here instead of
|
|
// ContentInfos.
|
|
if (session_->remote_description()) {
|
|
// Initialize the transport_options map.
|
|
for (const cricket::ContentInfo& content :
|
|
session_->remote_description()->description()->contents()) {
|
|
session_options->transport_options[content.name] =
|
|
cricket::TransportOptions();
|
|
}
|
|
}
|
|
AddSendStreams(session_options, senders_, rtp_data_channels_);
|
|
// RTP data channel is handled in MediaSessionOptions::AddStream. SCTP streams
|
|
// are not signaled in the SDP so does not go through that path and must be
|
|
// handled here.
|
|
// Intentionally unset the data channel type for RTP data channel. Otherwise
|
|
// the RTP data channels would be successfully negotiated by default and the
|
|
// unit tests in WebRtcDataBrowserTest will fail when building with chromium.
|
|
// We want to leave RTP data channels broken, so people won't try to use them.
|
|
if (session_->data_channel_type() != cricket::DCT_RTP) {
|
|
session_options->data_channel_type = session_->data_channel_type();
|
|
}
|
|
session_options->bundle_enabled =
|
|
session_options->bundle_enabled &&
|
|
(session_options->has_audio() || session_options->has_video() ||
|
|
session_options->has_data());
|
|
|
|
session_options->crypto_options = factory_->options().crypto_options;
|
|
}
|
|
|
|
bool PeerConnection::GetOptionsForAnswer(
|
|
const MediaConstraintsInterface* constraints,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
InitializeOptionsForAnswer(session_options);
|
|
if (!ParseConstraintsForAnswer(constraints, session_options)) {
|
|
return false;
|
|
}
|
|
session_options->rtcp_cname = rtcp_cname_;
|
|
|
|
FinishOptionsForAnswer(session_options);
|
|
return true;
|
|
}
|
|
|
|
bool PeerConnection::GetOptionsForAnswer(
|
|
const RTCOfferAnswerOptions& options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
InitializeOptionsForAnswer(session_options);
|
|
if (!ExtractMediaSessionOptions(options, false, session_options)) {
|
|
return false;
|
|
}
|
|
session_options->rtcp_cname = rtcp_cname_;
|
|
|
|
FinishOptionsForAnswer(session_options);
|
|
return true;
|
|
}
|
|
|
|
void PeerConnection::RemoveTracks(cricket::MediaType media_type) {
|
|
UpdateLocalTracks(std::vector<cricket::StreamParams>(), media_type);
|
|
UpdateRemoteStreamsList(std::vector<cricket::StreamParams>(), false,
|
|
media_type, nullptr);
|
|
}
|
|
|
|
void PeerConnection::UpdateRemoteStreamsList(
|
|
const cricket::StreamParamsVec& streams,
|
|
bool default_track_needed,
|
|
cricket::MediaType media_type,
|
|
StreamCollection* new_streams) {
|
|
TrackInfos* current_tracks = GetRemoteTracks(media_type);
|
|
|
|
// Find removed tracks. I.e., tracks where the track id or ssrc don't match
|
|
// the new StreamParam.
|
|
auto track_it = current_tracks->begin();
|
|
while (track_it != current_tracks->end()) {
|
|
const TrackInfo& info = *track_it;
|
|
const cricket::StreamParams* params =
|
|
cricket::GetStreamBySsrc(streams, info.ssrc);
|
|
bool track_exists = params && params->id == info.track_id;
|
|
// If this is a default track, and we still need it, don't remove it.
|
|
if ((info.stream_label == kDefaultStreamLabel && default_track_needed) ||
|
|
track_exists) {
|
|
++track_it;
|
|
} else {
|
|
OnRemoteTrackRemoved(info.stream_label, info.track_id, media_type);
|
|
track_it = current_tracks->erase(track_it);
|
|
}
|
|
}
|
|
|
|
// Find new and active tracks.
|
|
for (const cricket::StreamParams& params : streams) {
|
|
// The sync_label is the MediaStream label and the |stream.id| is the
|
|
// track id.
|
|
const std::string& stream_label = params.sync_label;
|
|
const std::string& track_id = params.id;
|
|
uint32_t ssrc = params.first_ssrc();
|
|
|
|
rtc::scoped_refptr<MediaStreamInterface> stream =
|
|
remote_streams_->find(stream_label);
|
|
if (!stream) {
|
|
// This is a new MediaStream. Create a new remote MediaStream.
|
|
stream = MediaStreamProxy::Create(rtc::Thread::Current(),
|
|
MediaStream::Create(stream_label));
|
|
remote_streams_->AddStream(stream);
|
|
new_streams->AddStream(stream);
|
|
}
|
|
|
|
const TrackInfo* track_info =
|
|
FindTrackInfo(*current_tracks, stream_label, track_id);
|
|
if (!track_info) {
|
|
current_tracks->push_back(TrackInfo(stream_label, track_id, ssrc));
|
|
OnRemoteTrackSeen(stream_label, track_id, ssrc, media_type);
|
|
}
|
|
}
|
|
|
|
// Add default track if necessary.
|
|
if (default_track_needed) {
|
|
rtc::scoped_refptr<MediaStreamInterface> default_stream =
|
|
remote_streams_->find(kDefaultStreamLabel);
|
|
if (!default_stream) {
|
|
// Create the new default MediaStream.
|
|
default_stream = MediaStreamProxy::Create(
|
|
rtc::Thread::Current(), MediaStream::Create(kDefaultStreamLabel));
|
|
remote_streams_->AddStream(default_stream);
|
|
new_streams->AddStream(default_stream);
|
|
}
|
|
std::string default_track_id = (media_type == cricket::MEDIA_TYPE_AUDIO)
|
|
? kDefaultAudioTrackLabel
|
|
: kDefaultVideoTrackLabel;
|
|
const TrackInfo* default_track_info =
|
|
FindTrackInfo(*current_tracks, kDefaultStreamLabel, default_track_id);
|
|
if (!default_track_info) {
|
|
current_tracks->push_back(
|
|
TrackInfo(kDefaultStreamLabel, default_track_id, 0));
|
|
OnRemoteTrackSeen(kDefaultStreamLabel, default_track_id, 0, media_type);
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnRemoteTrackSeen(const std::string& stream_label,
|
|
const std::string& track_id,
|
|
uint32_t ssrc,
|
|
cricket::MediaType media_type) {
|
|
MediaStreamInterface* stream = remote_streams_->find(stream_label);
|
|
|
|
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
|
|
CreateAudioReceiver(stream, track_id, ssrc);
|
|
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
|
|
CreateVideoReceiver(stream, track_id, ssrc);
|
|
} else {
|
|
RTC_DCHECK(false && "Invalid media type");
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnRemoteTrackRemoved(const std::string& stream_label,
|
|
const std::string& track_id,
|
|
cricket::MediaType media_type) {
|
|
MediaStreamInterface* stream = remote_streams_->find(stream_label);
|
|
|
|
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
|
|
// When the MediaEngine audio channel is destroyed, the RemoteAudioSource
|
|
// will be notified which will end the AudioRtpReceiver::track().
|
|
DestroyReceiver(track_id);
|
|
rtc::scoped_refptr<AudioTrackInterface> audio_track =
|
|
stream->FindAudioTrack(track_id);
|
|
if (audio_track) {
|
|
stream->RemoveTrack(audio_track);
|
|
}
|
|
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
|
|
// Stopping or destroying a VideoRtpReceiver will end the
|
|
// VideoRtpReceiver::track().
|
|
DestroyReceiver(track_id);
|
|
rtc::scoped_refptr<VideoTrackInterface> video_track =
|
|
stream->FindVideoTrack(track_id);
|
|
if (video_track) {
|
|
// There's no guarantee the track is still available, e.g. the track may
|
|
// have been removed from the stream by an application.
|
|
stream->RemoveTrack(video_track);
|
|
}
|
|
} else {
|
|
ASSERT(false && "Invalid media type");
|
|
}
|
|
}
|
|
|
|
void PeerConnection::UpdateEndedRemoteMediaStreams() {
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_to_remove;
|
|
for (size_t i = 0; i < remote_streams_->count(); ++i) {
|
|
MediaStreamInterface* stream = remote_streams_->at(i);
|
|
if (stream->GetAudioTracks().empty() && stream->GetVideoTracks().empty()) {
|
|
streams_to_remove.push_back(stream);
|
|
}
|
|
}
|
|
|
|
for (auto& stream : streams_to_remove) {
|
|
remote_streams_->RemoveStream(stream);
|
|
// Call both the raw pointer and scoped_refptr versions of the method
|
|
// for compatibility.
|
|
observer_->OnRemoveStream(stream.get());
|
|
observer_->OnRemoveStream(std::move(stream));
|
|
}
|
|
}
|
|
|
|
void PeerConnection::UpdateLocalTracks(
|
|
const std::vector<cricket::StreamParams>& streams,
|
|
cricket::MediaType media_type) {
|
|
TrackInfos* current_tracks = GetLocalTracks(media_type);
|
|
|
|
// Find removed tracks. I.e., tracks where the track id, stream label or ssrc
|
|
// don't match the new StreamParam.
|
|
TrackInfos::iterator track_it = current_tracks->begin();
|
|
while (track_it != current_tracks->end()) {
|
|
const TrackInfo& info = *track_it;
|
|
const cricket::StreamParams* params =
|
|
cricket::GetStreamBySsrc(streams, info.ssrc);
|
|
if (!params || params->id != info.track_id ||
|
|
params->sync_label != info.stream_label) {
|
|
OnLocalTrackRemoved(info.stream_label, info.track_id, info.ssrc,
|
|
media_type);
|
|
track_it = current_tracks->erase(track_it);
|
|
} else {
|
|
++track_it;
|
|
}
|
|
}
|
|
|
|
// Find new and active tracks.
|
|
for (const cricket::StreamParams& params : streams) {
|
|
// The sync_label is the MediaStream label and the |stream.id| is the
|
|
// track id.
|
|
const std::string& stream_label = params.sync_label;
|
|
const std::string& track_id = params.id;
|
|
uint32_t ssrc = params.first_ssrc();
|
|
const TrackInfo* track_info =
|
|
FindTrackInfo(*current_tracks, stream_label, track_id);
|
|
if (!track_info) {
|
|
current_tracks->push_back(TrackInfo(stream_label, track_id, ssrc));
|
|
OnLocalTrackSeen(stream_label, track_id, params.first_ssrc(), media_type);
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnLocalTrackSeen(const std::string& stream_label,
|
|
const std::string& track_id,
|
|
uint32_t ssrc,
|
|
cricket::MediaType media_type) {
|
|
RtpSenderInternal* sender = FindSenderById(track_id);
|
|
if (!sender) {
|
|
LOG(LS_WARNING) << "An unknown RtpSender with id " << track_id
|
|
<< " has been configured in the local description.";
|
|
return;
|
|
}
|
|
|
|
if (sender->media_type() != media_type) {
|
|
LOG(LS_WARNING) << "An RtpSender has been configured in the local"
|
|
<< " description with an unexpected media type.";
|
|
return;
|
|
}
|
|
|
|
sender->set_stream_id(stream_label);
|
|
sender->SetSsrc(ssrc);
|
|
}
|
|
|
|
void PeerConnection::OnLocalTrackRemoved(const std::string& stream_label,
|
|
const std::string& track_id,
|
|
uint32_t ssrc,
|
|
cricket::MediaType media_type) {
|
|
RtpSenderInternal* sender = FindSenderById(track_id);
|
|
if (!sender) {
|
|
// This is the normal case. I.e., RemoveStream has been called and the
|
|
// SessionDescriptions has been renegotiated.
|
|
return;
|
|
}
|
|
|
|
// A sender has been removed from the SessionDescription but it's still
|
|
// associated with the PeerConnection. This only occurs if the SDP doesn't
|
|
// match with the calls to CreateSender, AddStream and RemoveStream.
|
|
if (sender->media_type() != media_type) {
|
|
LOG(LS_WARNING) << "An RtpSender has been configured in the local"
|
|
<< " description with an unexpected media type.";
|
|
return;
|
|
}
|
|
|
|
sender->SetSsrc(0);
|
|
}
|
|
|
|
void PeerConnection::UpdateLocalRtpDataChannels(
|
|
const cricket::StreamParamsVec& streams) {
|
|
std::vector<std::string> existing_channels;
|
|
|
|
// Find new and active data channels.
|
|
for (const cricket::StreamParams& params : streams) {
|
|
// |it->sync_label| is actually the data channel label. The reason is that
|
|
// we use the same naming of data channels as we do for
|
|
// MediaStreams and Tracks.
|
|
// For MediaStreams, the sync_label is the MediaStream label and the
|
|
// track label is the same as |streamid|.
|
|
const std::string& channel_label = params.sync_label;
|
|
auto data_channel_it = rtp_data_channels_.find(channel_label);
|
|
if (!VERIFY(data_channel_it != rtp_data_channels_.end())) {
|
|
continue;
|
|
}
|
|
// Set the SSRC the data channel should use for sending.
|
|
data_channel_it->second->SetSendSsrc(params.first_ssrc());
|
|
existing_channels.push_back(data_channel_it->first);
|
|
}
|
|
|
|
UpdateClosingRtpDataChannels(existing_channels, true);
|
|
}
|
|
|
|
void PeerConnection::UpdateRemoteRtpDataChannels(
|
|
const cricket::StreamParamsVec& streams) {
|
|
std::vector<std::string> existing_channels;
|
|
|
|
// Find new and active data channels.
|
|
for (const cricket::StreamParams& params : streams) {
|
|
// The data channel label is either the mslabel or the SSRC if the mslabel
|
|
// does not exist. Ex a=ssrc:444330170 mslabel:test1.
|
|
std::string label = params.sync_label.empty()
|
|
? rtc::ToString(params.first_ssrc())
|
|
: params.sync_label;
|
|
auto data_channel_it = rtp_data_channels_.find(label);
|
|
if (data_channel_it == rtp_data_channels_.end()) {
|
|
// This is a new data channel.
|
|
CreateRemoteRtpDataChannel(label, params.first_ssrc());
|
|
} else {
|
|
data_channel_it->second->SetReceiveSsrc(params.first_ssrc());
|
|
}
|
|
existing_channels.push_back(label);
|
|
}
|
|
|
|
UpdateClosingRtpDataChannels(existing_channels, false);
|
|
}
|
|
|
|
void PeerConnection::UpdateClosingRtpDataChannels(
|
|
const std::vector<std::string>& active_channels,
|
|
bool is_local_update) {
|
|
auto it = rtp_data_channels_.begin();
|
|
while (it != rtp_data_channels_.end()) {
|
|
DataChannel* data_channel = it->second;
|
|
if (std::find(active_channels.begin(), active_channels.end(),
|
|
data_channel->label()) != active_channels.end()) {
|
|
++it;
|
|
continue;
|
|
}
|
|
|
|
if (is_local_update) {
|
|
data_channel->SetSendSsrc(0);
|
|
} else {
|
|
data_channel->RemotePeerRequestClose();
|
|
}
|
|
|
|
if (data_channel->state() == DataChannel::kClosed) {
|
|
rtp_data_channels_.erase(it);
|
|
it = rtp_data_channels_.begin();
|
|
} else {
|
|
++it;
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::CreateRemoteRtpDataChannel(const std::string& label,
|
|
uint32_t remote_ssrc) {
|
|
rtc::scoped_refptr<DataChannel> channel(
|
|
InternalCreateDataChannel(label, nullptr));
|
|
if (!channel.get()) {
|
|
LOG(LS_WARNING) << "Remote peer requested a DataChannel but"
|
|
<< "CreateDataChannel failed.";
|
|
return;
|
|
}
|
|
channel->SetReceiveSsrc(remote_ssrc);
|
|
rtc::scoped_refptr<DataChannelInterface> proxy_channel =
|
|
DataChannelProxy::Create(signaling_thread(), channel);
|
|
// Call both the raw pointer and scoped_refptr versions of the method
|
|
// for compatibility.
|
|
observer_->OnDataChannel(proxy_channel.get());
|
|
observer_->OnDataChannel(std::move(proxy_channel));
|
|
}
|
|
|
|
rtc::scoped_refptr<DataChannel> PeerConnection::InternalCreateDataChannel(
|
|
const std::string& label,
|
|
const InternalDataChannelInit* config) {
|
|
if (IsClosed()) {
|
|
return nullptr;
|
|
}
|
|
if (session_->data_channel_type() == cricket::DCT_NONE) {
|
|
LOG(LS_ERROR)
|
|
<< "InternalCreateDataChannel: Data is not supported in this call.";
|
|
return nullptr;
|
|
}
|
|
InternalDataChannelInit new_config =
|
|
config ? (*config) : InternalDataChannelInit();
|
|
if (session_->data_channel_type() == cricket::DCT_SCTP) {
|
|
if (new_config.id < 0) {
|
|
rtc::SSLRole role;
|
|
if ((session_->GetSslRole(session_->data_channel(), &role)) &&
|
|
!sid_allocator_.AllocateSid(role, &new_config.id)) {
|
|
LOG(LS_ERROR) << "No id can be allocated for the SCTP data channel.";
|
|
return nullptr;
|
|
}
|
|
} else if (!sid_allocator_.ReserveSid(new_config.id)) {
|
|
LOG(LS_ERROR) << "Failed to create a SCTP data channel "
|
|
<< "because the id is already in use or out of range.";
|
|
return nullptr;
|
|
}
|
|
}
|
|
|
|
rtc::scoped_refptr<DataChannel> channel(DataChannel::Create(
|
|
session_.get(), session_->data_channel_type(), label, new_config));
|
|
if (!channel) {
|
|
sid_allocator_.ReleaseSid(new_config.id);
|
|
return nullptr;
|
|
}
|
|
|
|
if (channel->data_channel_type() == cricket::DCT_RTP) {
|
|
if (rtp_data_channels_.find(channel->label()) != rtp_data_channels_.end()) {
|
|
LOG(LS_ERROR) << "DataChannel with label " << channel->label()
|
|
<< " already exists.";
|
|
return nullptr;
|
|
}
|
|
rtp_data_channels_[channel->label()] = channel;
|
|
} else {
|
|
RTC_DCHECK(channel->data_channel_type() == cricket::DCT_SCTP);
|
|
sctp_data_channels_.push_back(channel);
|
|
channel->SignalClosed.connect(this,
|
|
&PeerConnection::OnSctpDataChannelClosed);
|
|
}
|
|
|
|
SignalDataChannelCreated(channel.get());
|
|
return channel;
|
|
}
|
|
|
|
bool PeerConnection::HasDataChannels() const {
|
|
#ifdef HAVE_QUIC
|
|
return !rtp_data_channels_.empty() || !sctp_data_channels_.empty() ||
|
|
(session_->quic_data_transport() &&
|
|
session_->quic_data_transport()->HasDataChannels());
|
|
#else
|
|
return !rtp_data_channels_.empty() || !sctp_data_channels_.empty();
|
|
#endif // HAVE_QUIC
|
|
}
|
|
|
|
void PeerConnection::AllocateSctpSids(rtc::SSLRole role) {
|
|
for (const auto& channel : sctp_data_channels_) {
|
|
if (channel->id() < 0) {
|
|
int sid;
|
|
if (!sid_allocator_.AllocateSid(role, &sid)) {
|
|
LOG(LS_ERROR) << "Failed to allocate SCTP sid.";
|
|
continue;
|
|
}
|
|
channel->SetSctpSid(sid);
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnSctpDataChannelClosed(DataChannel* channel) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
for (auto it = sctp_data_channels_.begin(); it != sctp_data_channels_.end();
|
|
++it) {
|
|
if (it->get() == channel) {
|
|
if (channel->id() >= 0) {
|
|
sid_allocator_.ReleaseSid(channel->id());
|
|
}
|
|
// Since this method is triggered by a signal from the DataChannel,
|
|
// we can't free it directly here; we need to free it asynchronously.
|
|
sctp_data_channels_to_free_.push_back(*it);
|
|
sctp_data_channels_.erase(it);
|
|
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_FREE_DATACHANNELS,
|
|
nullptr);
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnVoiceChannelCreated() {
|
|
SetChannelOnSendersAndReceivers<AudioRtpSender, AudioRtpReceiver>(
|
|
session_->voice_channel(), senders_, receivers_,
|
|
cricket::MEDIA_TYPE_AUDIO);
|
|
}
|
|
|
|
void PeerConnection::OnVoiceChannelDestroyed() {
|
|
SetChannelOnSendersAndReceivers<AudioRtpSender, AudioRtpReceiver,
|
|
cricket::VoiceChannel>(
|
|
nullptr, senders_, receivers_, cricket::MEDIA_TYPE_AUDIO);
|
|
}
|
|
|
|
void PeerConnection::OnVideoChannelCreated() {
|
|
SetChannelOnSendersAndReceivers<VideoRtpSender, VideoRtpReceiver>(
|
|
session_->video_channel(), senders_, receivers_,
|
|
cricket::MEDIA_TYPE_VIDEO);
|
|
}
|
|
|
|
void PeerConnection::OnVideoChannelDestroyed() {
|
|
SetChannelOnSendersAndReceivers<VideoRtpSender, VideoRtpReceiver,
|
|
cricket::VideoChannel>(
|
|
nullptr, senders_, receivers_, cricket::MEDIA_TYPE_VIDEO);
|
|
}
|
|
|
|
void PeerConnection::OnDataChannelCreated() {
|
|
for (const auto& channel : sctp_data_channels_) {
|
|
channel->OnTransportChannelCreated();
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnDataChannelDestroyed() {
|
|
// Use a temporary copy of the RTP/SCTP DataChannel list because the
|
|
// DataChannel may callback to us and try to modify the list.
|
|
std::map<std::string, rtc::scoped_refptr<DataChannel>> temp_rtp_dcs;
|
|
temp_rtp_dcs.swap(rtp_data_channels_);
|
|
for (const auto& kv : temp_rtp_dcs) {
|
|
kv.second->OnTransportChannelDestroyed();
|
|
}
|
|
|
|
std::vector<rtc::scoped_refptr<DataChannel>> temp_sctp_dcs;
|
|
temp_sctp_dcs.swap(sctp_data_channels_);
|
|
for (const auto& channel : temp_sctp_dcs) {
|
|
channel->OnTransportChannelDestroyed();
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnDataChannelOpenMessage(
|
|
const std::string& label,
|
|
const InternalDataChannelInit& config) {
|
|
rtc::scoped_refptr<DataChannel> channel(
|
|
InternalCreateDataChannel(label, &config));
|
|
if (!channel.get()) {
|
|
LOG(LS_ERROR) << "Failed to create DataChannel from the OPEN message.";
|
|
return;
|
|
}
|
|
|
|
rtc::scoped_refptr<DataChannelInterface> proxy_channel =
|
|
DataChannelProxy::Create(signaling_thread(), channel);
|
|
// Call both the raw pointer and scoped_refptr versions of the method
|
|
// for compatibility.
|
|
observer_->OnDataChannel(proxy_channel.get());
|
|
observer_->OnDataChannel(std::move(proxy_channel));
|
|
}
|
|
|
|
RtpSenderInternal* PeerConnection::FindSenderById(const std::string& id) {
|
|
auto it = std::find_if(
|
|
senders_.begin(), senders_.end(),
|
|
[id](const rtc::scoped_refptr<
|
|
RtpSenderProxyWithInternal<RtpSenderInternal>>& sender) {
|
|
return sender->id() == id;
|
|
});
|
|
return it != senders_.end() ? (*it)->internal() : nullptr;
|
|
}
|
|
|
|
std::vector<
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>::iterator
|
|
PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) {
|
|
return std::find_if(
|
|
senders_.begin(), senders_.end(),
|
|
[track](const rtc::scoped_refptr<
|
|
RtpSenderProxyWithInternal<RtpSenderInternal>>& sender) {
|
|
return sender->track() == track;
|
|
});
|
|
}
|
|
|
|
std::vector<rtc::scoped_refptr<
|
|
RtpReceiverProxyWithInternal<RtpReceiverInternal>>>::iterator
|
|
PeerConnection::FindReceiverForTrack(const std::string& track_id) {
|
|
return std::find_if(
|
|
receivers_.begin(), receivers_.end(),
|
|
[track_id](const rtc::scoped_refptr<
|
|
RtpReceiverProxyWithInternal<RtpReceiverInternal>>& receiver) {
|
|
return receiver->id() == track_id;
|
|
});
|
|
}
|
|
|
|
PeerConnection::TrackInfos* PeerConnection::GetRemoteTracks(
|
|
cricket::MediaType media_type) {
|
|
RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
|
|
media_type == cricket::MEDIA_TYPE_VIDEO);
|
|
return (media_type == cricket::MEDIA_TYPE_AUDIO) ? &remote_audio_tracks_
|
|
: &remote_video_tracks_;
|
|
}
|
|
|
|
PeerConnection::TrackInfos* PeerConnection::GetLocalTracks(
|
|
cricket::MediaType media_type) {
|
|
RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
|
|
media_type == cricket::MEDIA_TYPE_VIDEO);
|
|
return (media_type == cricket::MEDIA_TYPE_AUDIO) ? &local_audio_tracks_
|
|
: &local_video_tracks_;
|
|
}
|
|
|
|
const PeerConnection::TrackInfo* PeerConnection::FindTrackInfo(
|
|
const PeerConnection::TrackInfos& infos,
|
|
const std::string& stream_label,
|
|
const std::string track_id) const {
|
|
for (const TrackInfo& track_info : infos) {
|
|
if (track_info.stream_label == stream_label &&
|
|
track_info.track_id == track_id) {
|
|
return &track_info;
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
DataChannel* PeerConnection::FindDataChannelBySid(int sid) const {
|
|
for (const auto& channel : sctp_data_channels_) {
|
|
if (channel->id() == sid) {
|
|
return channel;
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
bool PeerConnection::InitializePortAllocator_n(
|
|
const RTCConfiguration& configuration) {
|
|
cricket::ServerAddresses stun_servers;
|
|
std::vector<cricket::RelayServerConfig> turn_servers;
|
|
if (!ParseIceServers(configuration.servers, &stun_servers, &turn_servers)) {
|
|
return false;
|
|
}
|
|
|
|
port_allocator_->Initialize();
|
|
|
|
// To handle both internal and externally created port allocator, we will
|
|
// enable BUNDLE here.
|
|
int portallocator_flags = port_allocator_->flags();
|
|
portallocator_flags |= cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
|
|
cricket::PORTALLOCATOR_ENABLE_IPV6;
|
|
// If the disable-IPv6 flag was specified, we'll not override it
|
|
// by experiment.
|
|
if (configuration.disable_ipv6) {
|
|
portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
|
|
} else if (webrtc::field_trial::FindFullName("WebRTC-IPv6Default") ==
|
|
"Disabled") {
|
|
portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
|
|
}
|
|
|
|
if (configuration.tcp_candidate_policy == kTcpCandidatePolicyDisabled) {
|
|
portallocator_flags |= cricket::PORTALLOCATOR_DISABLE_TCP;
|
|
LOG(LS_INFO) << "TCP candidates are disabled.";
|
|
}
|
|
|
|
if (configuration.candidate_network_policy ==
|
|
kCandidateNetworkPolicyLowCost) {
|
|
portallocator_flags |= cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS;
|
|
LOG(LS_INFO) << "Do not gather candidates on high-cost networks";
|
|
}
|
|
|
|
port_allocator_->set_flags(portallocator_flags);
|
|
// No step delay is used while allocating ports.
|
|
port_allocator_->set_step_delay(cricket::kMinimumStepDelay);
|
|
port_allocator_->set_candidate_filter(
|
|
ConvertIceTransportTypeToCandidateFilter(configuration.type));
|
|
|
|
// Call this last since it may create pooled allocator sessions using the
|
|
// properties set above.
|
|
port_allocator_->SetConfiguration(stun_servers, turn_servers,
|
|
configuration.ice_candidate_pool_size,
|
|
configuration.prune_turn_ports);
|
|
return true;
|
|
}
|
|
|
|
bool PeerConnection::ReconfigurePortAllocator_n(
|
|
const RTCConfiguration& configuration) {
|
|
cricket::ServerAddresses stun_servers;
|
|
std::vector<cricket::RelayServerConfig> turn_servers;
|
|
if (!ParseIceServers(configuration.servers, &stun_servers, &turn_servers)) {
|
|
return false;
|
|
}
|
|
port_allocator_->set_candidate_filter(
|
|
ConvertIceTransportTypeToCandidateFilter(configuration.type));
|
|
// Call this last since it may create pooled allocator sessions using the
|
|
// candidate filter set above.
|
|
return port_allocator_->SetConfiguration(
|
|
stun_servers, turn_servers, configuration.ice_candidate_pool_size,
|
|
configuration.prune_turn_ports);
|
|
}
|
|
|
|
bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file,
|
|
int64_t max_size_bytes) {
|
|
return event_log_->StartLogging(file, max_size_bytes);
|
|
}
|
|
|
|
void PeerConnection::StopRtcEventLog_w() {
|
|
event_log_->StopLogging();
|
|
}
|
|
} // namespace webrtc
|