| Index: webrtc/api/peerconnection.cc
|
| diff --git a/webrtc/api/peerconnection.cc b/webrtc/api/peerconnection.cc
|
| deleted file mode 100644
|
| index 78e6790dd7aa0ecbffc0dd161061b338b09bed5f..0000000000000000000000000000000000000000
|
| --- a/webrtc/api/peerconnection.cc
|
| +++ /dev/null
|
| @@ -1,2573 +0,0 @@
|
| -/*
|
| - * Copyright 2012 The WebRTC project authors. All Rights Reserved.
|
| - *
|
| - * Use of this source code is governed by a BSD-style license
|
| - * that can be found in the LICENSE file in the root of the source
|
| - * tree. An additional intellectual property rights grant can be found
|
| - * in the file PATENTS. All contributing project authors may
|
| - * be found in the AUTHORS file in the root of the source tree.
|
| - */
|
| -
|
| -#include "webrtc/api/peerconnection.h"
|
| -
|
| -#include <algorithm>
|
| -#include <cctype> // for isdigit
|
| -#include <utility>
|
| -#include <vector>
|
| -
|
| -#include "webrtc/api/audiotrack.h"
|
| -#include "webrtc/api/dtmfsender.h"
|
| -#include "webrtc/api/jsepicecandidate.h"
|
| -#include "webrtc/api/jsepsessiondescription.h"
|
| -#include "webrtc/api/mediaconstraintsinterface.h"
|
| -#include "webrtc/api/mediastream.h"
|
| -#include "webrtc/api/mediastreamobserver.h"
|
| -#include "webrtc/api/mediastreamproxy.h"
|
| -#include "webrtc/api/mediastreamtrackproxy.h"
|
| -#include "webrtc/api/remoteaudiosource.h"
|
| -#include "webrtc/api/rtpreceiver.h"
|
| -#include "webrtc/api/rtpsender.h"
|
| -#include "webrtc/api/streamcollection.h"
|
| -#include "webrtc/api/videocapturertracksource.h"
|
| -#include "webrtc/api/videotrack.h"
|
| -#include "webrtc/base/arraysize.h"
|
| -#include "webrtc/base/bind.h"
|
| -#include "webrtc/base/checks.h"
|
| -#include "webrtc/base/logging.h"
|
| -#include "webrtc/base/stringencode.h"
|
| -#include "webrtc/base/stringutils.h"
|
| -#include "webrtc/base/trace_event.h"
|
| -#include "webrtc/call/call.h"
|
| -#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
|
| -#include "webrtc/media/sctp/sctptransport.h"
|
| -#include "webrtc/pc/channelmanager.h"
|
| -#include "webrtc/system_wrappers/include/field_trial.h"
|
| -
|
| -namespace {
|
| -
|
| -using webrtc::DataChannel;
|
| -using webrtc::MediaConstraintsInterface;
|
| -using webrtc::MediaStreamInterface;
|
| -using webrtc::PeerConnectionInterface;
|
| -using webrtc::RTCError;
|
| -using webrtc::RTCErrorType;
|
| -using webrtc::RtpSenderInternal;
|
| -using webrtc::RtpSenderInterface;
|
| -using webrtc::RtpSenderProxy;
|
| -using webrtc::RtpSenderProxyWithInternal;
|
| -using webrtc::StreamCollection;
|
| -
|
| -static const char kDefaultStreamLabel[] = "default";
|
| -static const char kDefaultAudioTrackLabel[] = "defaulta0";
|
| -static const char kDefaultVideoTrackLabel[] = "defaultv0";
|
| -
|
| -// The min number of tokens must present in Turn host uri.
|
| -// e.g. user@turn.example.org
|
| -static const size_t kTurnHostTokensNum = 2;
|
| -// Number of tokens must be preset when TURN uri has transport param.
|
| -static const size_t kTurnTransportTokensNum = 2;
|
| -// The default stun port.
|
| -static const int kDefaultStunPort = 3478;
|
| -static const int kDefaultStunTlsPort = 5349;
|
| -static const char kTransport[] = "transport";
|
| -
|
| -// NOTE: Must be in the same order as the ServiceType enum.
|
| -static const char* kValidIceServiceTypes[] = {"stun", "stuns", "turn", "turns"};
|
| -
|
| -// The length of RTCP CNAMEs.
|
| -static const int kRtcpCnameLength = 16;
|
| -
|
| -// NOTE: A loop below assumes that the first value of this enum is 0 and all
|
| -// other values are incremental.
|
| -enum ServiceType {
|
| - STUN = 0, // Indicates a STUN server.
|
| - STUNS, // Indicates a STUN server used with a TLS session.
|
| - TURN, // Indicates a TURN server
|
| - TURNS, // Indicates a TURN server used with a TLS session.
|
| - INVALID, // Unknown.
|
| -};
|
| -static_assert(INVALID == arraysize(kValidIceServiceTypes),
|
| - "kValidIceServiceTypes must have as many strings as ServiceType "
|
| - "has values.");
|
| -
|
| -enum {
|
| - MSG_SET_SESSIONDESCRIPTION_SUCCESS = 0,
|
| - MSG_SET_SESSIONDESCRIPTION_FAILED,
|
| - MSG_CREATE_SESSIONDESCRIPTION_FAILED,
|
| - MSG_GETSTATS,
|
| - MSG_FREE_DATACHANNELS,
|
| -};
|
| -
|
| -struct SetSessionDescriptionMsg : public rtc::MessageData {
|
| - explicit SetSessionDescriptionMsg(
|
| - webrtc::SetSessionDescriptionObserver* observer)
|
| - : observer(observer) {
|
| - }
|
| -
|
| - rtc::scoped_refptr<webrtc::SetSessionDescriptionObserver> observer;
|
| - std::string error;
|
| -};
|
| -
|
| -struct CreateSessionDescriptionMsg : public rtc::MessageData {
|
| - explicit CreateSessionDescriptionMsg(
|
| - webrtc::CreateSessionDescriptionObserver* observer)
|
| - : observer(observer) {}
|
| -
|
| - rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserver> observer;
|
| - std::string error;
|
| -};
|
| -
|
| -struct GetStatsMsg : public rtc::MessageData {
|
| - GetStatsMsg(webrtc::StatsObserver* observer,
|
| - webrtc::MediaStreamTrackInterface* track)
|
| - : observer(observer), track(track) {
|
| - }
|
| - rtc::scoped_refptr<webrtc::StatsObserver> observer;
|
| - rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track;
|
| -};
|
| -
|
| -// |in_str| should be of format
|
| -// stunURI = scheme ":" stun-host [ ":" stun-port ]
|
| -// scheme = "stun" / "stuns"
|
| -// stun-host = IP-literal / IPv4address / reg-name
|
| -// stun-port = *DIGIT
|
| -//
|
| -// draft-petithuguenin-behave-turn-uris-01
|
| -// turnURI = scheme ":" turn-host [ ":" turn-port ]
|
| -// turn-host = username@IP-literal / IPv4address / reg-name
|
| -bool GetServiceTypeAndHostnameFromUri(const std::string& in_str,
|
| - ServiceType* service_type,
|
| - std::string* hostname) {
|
| - const std::string::size_type colonpos = in_str.find(':');
|
| - if (colonpos == std::string::npos) {
|
| - LOG(LS_WARNING) << "Missing ':' in ICE URI: " << in_str;
|
| - return false;
|
| - }
|
| - if ((colonpos + 1) == in_str.length()) {
|
| - LOG(LS_WARNING) << "Empty hostname in ICE URI: " << in_str;
|
| - return false;
|
| - }
|
| - *service_type = INVALID;
|
| - for (size_t i = 0; i < arraysize(kValidIceServiceTypes); ++i) {
|
| - if (in_str.compare(0, colonpos, kValidIceServiceTypes[i]) == 0) {
|
| - *service_type = static_cast<ServiceType>(i);
|
| - break;
|
| - }
|
| - }
|
| - if (*service_type == INVALID) {
|
| - return false;
|
| - }
|
| - *hostname = in_str.substr(colonpos + 1, std::string::npos);
|
| - return true;
|
| -}
|
| -
|
| -bool ParsePort(const std::string& in_str, int* port) {
|
| - // Make sure port only contains digits. FromString doesn't check this.
|
| - for (const char& c : in_str) {
|
| - if (!std::isdigit(c)) {
|
| - return false;
|
| - }
|
| - }
|
| - return rtc::FromString(in_str, port);
|
| -}
|
| -
|
| -// This method parses IPv6 and IPv4 literal strings, along with hostnames in
|
| -// standard hostname:port format.
|
| -// Consider following formats as correct.
|
| -// |hostname:port|, |[IPV6 address]:port|, |IPv4 address|:port,
|
| -// |hostname|, |[IPv6 address]|, |IPv4 address|.
|
| -bool ParseHostnameAndPortFromString(const std::string& in_str,
|
| - std::string* host,
|
| - int* port) {
|
| - RTC_DCHECK(host->empty());
|
| - if (in_str.at(0) == '[') {
|
| - std::string::size_type closebracket = in_str.rfind(']');
|
| - if (closebracket != std::string::npos) {
|
| - std::string::size_type colonpos = in_str.find(':', closebracket);
|
| - if (std::string::npos != colonpos) {
|
| - if (!ParsePort(in_str.substr(closebracket + 2, std::string::npos),
|
| - port)) {
|
| - return false;
|
| - }
|
| - }
|
| - *host = in_str.substr(1, closebracket - 1);
|
| - } else {
|
| - return false;
|
| - }
|
| - } else {
|
| - std::string::size_type colonpos = in_str.find(':');
|
| - if (std::string::npos != colonpos) {
|
| - if (!ParsePort(in_str.substr(colonpos + 1, std::string::npos), port)) {
|
| - return false;
|
| - }
|
| - *host = in_str.substr(0, colonpos);
|
| - } else {
|
| - *host = in_str;
|
| - }
|
| - }
|
| - return !host->empty();
|
| -}
|
| -
|
| -// Adds a STUN or TURN server to the appropriate list,
|
| -// by parsing |url| and using the username/password in |server|.
|
| -RTCErrorType ParseIceServerUrl(
|
| - const PeerConnectionInterface::IceServer& server,
|
| - const std::string& url,
|
| - cricket::ServerAddresses* stun_servers,
|
| - std::vector<cricket::RelayServerConfig>* turn_servers) {
|
| - // draft-nandakumar-rtcweb-stun-uri-01
|
| - // stunURI = scheme ":" stun-host [ ":" stun-port ]
|
| - // scheme = "stun" / "stuns"
|
| - // stun-host = IP-literal / IPv4address / reg-name
|
| - // stun-port = *DIGIT
|
| -
|
| - // draft-petithuguenin-behave-turn-uris-01
|
| - // turnURI = scheme ":" turn-host [ ":" turn-port ]
|
| - // [ "?transport=" transport ]
|
| - // scheme = "turn" / "turns"
|
| - // transport = "udp" / "tcp" / transport-ext
|
| - // transport-ext = 1*unreserved
|
| - // turn-host = IP-literal / IPv4address / reg-name
|
| - // turn-port = *DIGIT
|
| - RTC_DCHECK(stun_servers != nullptr);
|
| - RTC_DCHECK(turn_servers != nullptr);
|
| - std::vector<std::string> tokens;
|
| - cricket::ProtocolType turn_transport_type = cricket::PROTO_UDP;
|
| - RTC_DCHECK(!url.empty());
|
| - rtc::tokenize_with_empty_tokens(url, '?', &tokens);
|
| - std::string uri_without_transport = tokens[0];
|
| - // Let's look into transport= param, if it exists.
|
| - if (tokens.size() == kTurnTransportTokensNum) { // ?transport= is present.
|
| - std::string uri_transport_param = tokens[1];
|
| - rtc::tokenize_with_empty_tokens(uri_transport_param, '=', &tokens);
|
| - if (tokens[0] != kTransport) {
|
| - LOG(LS_WARNING) << "Invalid transport parameter key.";
|
| - return RTCErrorType::SYNTAX_ERROR;
|
| - }
|
| - if (tokens.size() < 2 ||
|
| - !cricket::StringToProto(tokens[1].c_str(), &turn_transport_type) ||
|
| - (turn_transport_type != cricket::PROTO_UDP &&
|
| - turn_transport_type != cricket::PROTO_TCP)) {
|
| - LOG(LS_WARNING) << "Transport param should always be udp or tcp.";
|
| - return RTCErrorType::SYNTAX_ERROR;
|
| - }
|
| - }
|
| -
|
| - std::string hoststring;
|
| - ServiceType service_type;
|
| - if (!GetServiceTypeAndHostnameFromUri(uri_without_transport,
|
| - &service_type,
|
| - &hoststring)) {
|
| - LOG(LS_WARNING) << "Invalid transport parameter in ICE URI: " << url;
|
| - return RTCErrorType::SYNTAX_ERROR;
|
| - }
|
| -
|
| - // GetServiceTypeAndHostnameFromUri should never give an empty hoststring
|
| - RTC_DCHECK(!hoststring.empty());
|
| -
|
| - // Let's break hostname.
|
| - tokens.clear();
|
| - rtc::tokenize_with_empty_tokens(hoststring, '@', &tokens);
|
| -
|
| - std::string username(server.username);
|
| - if (tokens.size() > kTurnHostTokensNum) {
|
| - LOG(LS_WARNING) << "Invalid user@hostname format: " << hoststring;
|
| - return RTCErrorType::SYNTAX_ERROR;
|
| - }
|
| - if (tokens.size() == kTurnHostTokensNum) {
|
| - if (tokens[0].empty() || tokens[1].empty()) {
|
| - LOG(LS_WARNING) << "Invalid user@hostname format: " << hoststring;
|
| - return RTCErrorType::SYNTAX_ERROR;
|
| - }
|
| - username.assign(rtc::s_url_decode(tokens[0]));
|
| - hoststring = tokens[1];
|
| - } else {
|
| - hoststring = tokens[0];
|
| - }
|
| -
|
| - int port = kDefaultStunPort;
|
| - if (service_type == TURNS) {
|
| - port = kDefaultStunTlsPort;
|
| - turn_transport_type = cricket::PROTO_TLS;
|
| - }
|
| -
|
| - std::string address;
|
| - if (!ParseHostnameAndPortFromString(hoststring, &address, &port)) {
|
| - LOG(WARNING) << "Invalid hostname format: " << uri_without_transport;
|
| - return RTCErrorType::SYNTAX_ERROR;
|
| - }
|
| -
|
| - if (port <= 0 || port > 0xffff) {
|
| - LOG(WARNING) << "Invalid port: " << port;
|
| - return RTCErrorType::SYNTAX_ERROR;
|
| - }
|
| -
|
| - switch (service_type) {
|
| - case STUN:
|
| - case STUNS:
|
| - stun_servers->insert(rtc::SocketAddress(address, port));
|
| - break;
|
| - case TURN:
|
| - case TURNS: {
|
| - if (username.empty() || server.password.empty()) {
|
| - // The WebRTC spec requires throwing an InvalidAccessError when username
|
| - // or credential are ommitted; this is the native equivalent.
|
| - return RTCErrorType::INVALID_PARAMETER;
|
| - }
|
| - cricket::RelayServerConfig config = cricket::RelayServerConfig(
|
| - address, port, username, server.password, turn_transport_type);
|
| - if (server.tls_cert_policy ==
|
| - PeerConnectionInterface::kTlsCertPolicyInsecureNoCheck) {
|
| - config.tls_cert_policy =
|
| - cricket::TlsCertPolicy::TLS_CERT_POLICY_INSECURE_NO_CHECK;
|
| - }
|
| - turn_servers->push_back(config);
|
| - break;
|
| - }
|
| - default:
|
| - // We shouldn't get to this point with an invalid service_type, we should
|
| - // have returned an error already.
|
| - RTC_NOTREACHED() << "Unexpected service type";
|
| - return RTCErrorType::INTERNAL_ERROR;
|
| - }
|
| - return RTCErrorType::NONE;
|
| -}
|
| -
|
| -// Check if we can send |new_stream| on a PeerConnection.
|
| -bool CanAddLocalMediaStream(webrtc::StreamCollectionInterface* current_streams,
|
| - webrtc::MediaStreamInterface* new_stream) {
|
| - if (!new_stream || !current_streams) {
|
| - return false;
|
| - }
|
| - if (current_streams->find(new_stream->label()) != nullptr) {
|
| - LOG(LS_ERROR) << "MediaStream with label " << new_stream->label()
|
| - << " is already added.";
|
| - return false;
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -bool MediaContentDirectionHasSend(cricket::MediaContentDirection dir) {
|
| - return dir == cricket::MD_SENDONLY || dir == cricket::MD_SENDRECV;
|
| -}
|
| -
|
| -// If the direction is "recvonly" or "inactive", treat the description
|
| -// as containing no streams.
|
| -// See: https://code.google.com/p/webrtc/issues/detail?id=5054
|
| -std::vector<cricket::StreamParams> GetActiveStreams(
|
| - const cricket::MediaContentDescription* desc) {
|
| - return MediaContentDirectionHasSend(desc->direction())
|
| - ? desc->streams()
|
| - : std::vector<cricket::StreamParams>();
|
| -}
|
| -
|
| -bool IsValidOfferToReceiveMedia(int value) {
|
| - typedef PeerConnectionInterface::RTCOfferAnswerOptions Options;
|
| - return (value >= Options::kUndefined) &&
|
| - (value <= Options::kMaxOfferToReceiveMedia);
|
| -}
|
| -
|
| -// Add the stream and RTP data channel info to |session_options|.
|
| -void AddSendStreams(
|
| - cricket::MediaSessionOptions* session_options,
|
| - const std::vector<rtc::scoped_refptr<
|
| - RtpSenderProxyWithInternal<RtpSenderInternal>>>& senders,
|
| - const std::map<std::string, rtc::scoped_refptr<DataChannel>>&
|
| - rtp_data_channels) {
|
| - session_options->streams.clear();
|
| - for (const auto& sender : senders) {
|
| - session_options->AddSendStream(sender->media_type(), sender->id(),
|
| - sender->internal()->stream_id());
|
| - }
|
| -
|
| - // Check for data channels.
|
| - for (const auto& kv : rtp_data_channels) {
|
| - const DataChannel* channel = kv.second;
|
| - if (channel->state() == DataChannel::kConnecting ||
|
| - channel->state() == DataChannel::kOpen) {
|
| - // |streamid| and |sync_label| are both set to the DataChannel label
|
| - // here so they can be signaled the same way as MediaStreams and Tracks.
|
| - // For MediaStreams, the sync_label is the MediaStream label and the
|
| - // track label is the same as |streamid|.
|
| - const std::string& streamid = channel->label();
|
| - const std::string& sync_label = channel->label();
|
| - session_options->AddSendStream(cricket::MEDIA_TYPE_DATA, streamid,
|
| - sync_label);
|
| - }
|
| - }
|
| -}
|
| -
|
| -uint32_t ConvertIceTransportTypeToCandidateFilter(
|
| - PeerConnectionInterface::IceTransportsType type) {
|
| - switch (type) {
|
| - case PeerConnectionInterface::kNone:
|
| - return cricket::CF_NONE;
|
| - case PeerConnectionInterface::kRelay:
|
| - return cricket::CF_RELAY;
|
| - case PeerConnectionInterface::kNoHost:
|
| - return (cricket::CF_ALL & ~cricket::CF_HOST);
|
| - case PeerConnectionInterface::kAll:
|
| - return cricket::CF_ALL;
|
| - default:
|
| - RTC_NOTREACHED();
|
| - }
|
| - return cricket::CF_NONE;
|
| -}
|
| -
|
| -// Helper method to set a voice/video channel on all applicable senders
|
| -// and receivers when one is created/destroyed by WebRtcSession.
|
| -//
|
| -// Used by On(Voice|Video)Channel(Created|Destroyed)
|
| -template <class SENDER,
|
| - class RECEIVER,
|
| - class CHANNEL,
|
| - class SENDERS,
|
| - class RECEIVERS>
|
| -void SetChannelOnSendersAndReceivers(CHANNEL* channel,
|
| - SENDERS& senders,
|
| - RECEIVERS& receivers,
|
| - cricket::MediaType media_type) {
|
| - for (auto& sender : senders) {
|
| - if (sender->media_type() == media_type) {
|
| - static_cast<SENDER*>(sender->internal())->SetChannel(channel);
|
| - }
|
| - }
|
| - for (auto& receiver : receivers) {
|
| - if (receiver->media_type() == media_type) {
|
| - if (!channel) {
|
| - receiver->internal()->Stop();
|
| - }
|
| - static_cast<RECEIVER*>(receiver->internal())->SetChannel(channel);
|
| - }
|
| - }
|
| -}
|
| -
|
| -// Helper to set an error and return from a method.
|
| -bool SafeSetError(webrtc::RTCErrorType type, webrtc::RTCError* error) {
|
| - if (error) {
|
| - error->set_type(type);
|
| - }
|
| - return type == webrtc::RTCErrorType::NONE;
|
| -}
|
| -
|
| -} // namespace
|
| -
|
| -namespace webrtc {
|
| -
|
| -static const char* const kRTCErrorTypeNames[] = {
|
| - "NONE",
|
| - "UNSUPPORTED_PARAMETER",
|
| - "INVALID_PARAMETER",
|
| - "INVALID_RANGE",
|
| - "SYNTAX_ERROR",
|
| - "INVALID_STATE",
|
| - "INVALID_MODIFICATION",
|
| - "NETWORK_ERROR",
|
| - "INTERNAL_ERROR",
|
| -};
|
| -static_assert(static_cast<int>(RTCErrorType::INTERNAL_ERROR) ==
|
| - (arraysize(kRTCErrorTypeNames) - 1),
|
| - "kRTCErrorTypeNames must have as many strings as RTCErrorType "
|
| - "has values.");
|
| -
|
| -std::ostream& operator<<(std::ostream& stream, RTCErrorType error) {
|
| - int index = static_cast<int>(error);
|
| - return stream << kRTCErrorTypeNames[index];
|
| -}
|
| -
|
| -bool PeerConnectionInterface::RTCConfiguration::operator==(
|
| - const PeerConnectionInterface::RTCConfiguration& o) const {
|
| - // This static_assert prevents us from accidentally breaking operator==.
|
| - struct stuff_being_tested_for_equality {
|
| - IceTransportsType type;
|
| - IceServers servers;
|
| - BundlePolicy bundle_policy;
|
| - RtcpMuxPolicy rtcp_mux_policy;
|
| - TcpCandidatePolicy tcp_candidate_policy;
|
| - CandidateNetworkPolicy candidate_network_policy;
|
| - int audio_jitter_buffer_max_packets;
|
| - bool audio_jitter_buffer_fast_accelerate;
|
| - int ice_connection_receiving_timeout;
|
| - int ice_backup_candidate_pair_ping_interval;
|
| - ContinualGatheringPolicy continual_gathering_policy;
|
| - std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
|
| - bool prioritize_most_likely_ice_candidate_pairs;
|
| - struct cricket::MediaConfig media_config;
|
| - bool disable_ipv6;
|
| - bool enable_rtp_data_channel;
|
| - bool enable_quic;
|
| - rtc::Optional<int> screencast_min_bitrate;
|
| - rtc::Optional<bool> combined_audio_video_bwe;
|
| - rtc::Optional<bool> enable_dtls_srtp;
|
| - int ice_candidate_pool_size;
|
| - bool prune_turn_ports;
|
| - bool presume_writable_when_fully_relayed;
|
| - bool enable_ice_renomination;
|
| - bool redetermine_role_on_ice_restart;
|
| - };
|
| - static_assert(sizeof(stuff_being_tested_for_equality) == sizeof(*this),
|
| - "Did you add something to RTCConfiguration and forget to "
|
| - "update operator==?");
|
| - return type == o.type && servers == o.servers &&
|
| - bundle_policy == o.bundle_policy &&
|
| - rtcp_mux_policy == o.rtcp_mux_policy &&
|
| - tcp_candidate_policy == o.tcp_candidate_policy &&
|
| - candidate_network_policy == o.candidate_network_policy &&
|
| - audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
|
| - audio_jitter_buffer_fast_accelerate ==
|
| - o.audio_jitter_buffer_fast_accelerate &&
|
| - ice_connection_receiving_timeout ==
|
| - o.ice_connection_receiving_timeout &&
|
| - ice_backup_candidate_pair_ping_interval ==
|
| - o.ice_backup_candidate_pair_ping_interval &&
|
| - continual_gathering_policy == o.continual_gathering_policy &&
|
| - certificates == o.certificates &&
|
| - prioritize_most_likely_ice_candidate_pairs ==
|
| - o.prioritize_most_likely_ice_candidate_pairs &&
|
| - media_config == o.media_config && disable_ipv6 == o.disable_ipv6 &&
|
| - enable_rtp_data_channel == o.enable_rtp_data_channel &&
|
| - enable_quic == o.enable_quic &&
|
| - screencast_min_bitrate == o.screencast_min_bitrate &&
|
| - combined_audio_video_bwe == o.combined_audio_video_bwe &&
|
| - enable_dtls_srtp == o.enable_dtls_srtp &&
|
| - ice_candidate_pool_size == o.ice_candidate_pool_size &&
|
| - prune_turn_ports == o.prune_turn_ports &&
|
| - presume_writable_when_fully_relayed ==
|
| - o.presume_writable_when_fully_relayed &&
|
| - enable_ice_renomination == o.enable_ice_renomination &&
|
| - redetermine_role_on_ice_restart == o.redetermine_role_on_ice_restart;
|
| -}
|
| -
|
| -bool PeerConnectionInterface::RTCConfiguration::operator!=(
|
| - const PeerConnectionInterface::RTCConfiguration& o) const {
|
| - return !(*this == o);
|
| -}
|
| -
|
| -// Generate a RTCP CNAME when a PeerConnection is created.
|
| -std::string GenerateRtcpCname() {
|
| - std::string cname;
|
| - if (!rtc::CreateRandomString(kRtcpCnameLength, &cname)) {
|
| - LOG(LS_ERROR) << "Failed to generate CNAME.";
|
| - RTC_NOTREACHED();
|
| - }
|
| - return cname;
|
| -}
|
| -
|
| -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();
|
| -}
|
| -
|
| -RTCErrorType 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 RTCErrorType::SYNTAX_ERROR;
|
| - }
|
| - RTCErrorType err =
|
| - ParseIceServerUrl(server, url, stun_servers, turn_servers);
|
| - if (err != RTCErrorType::NONE) {
|
| - return err;
|
| - }
|
| - }
|
| - } else if (!server.uri.empty()) {
|
| - // Fallback to old .uri if new .urls isn't present.
|
| - RTCErrorType err =
|
| - ParseIceServerUrl(server, server.uri, stun_servers, turn_servers);
|
| - if (err != RTCErrorType::NONE) {
|
| - return err;
|
| - }
|
| - } else {
|
| - LOG(LS_ERROR) << "Empty uri.";
|
| - return RTCErrorType::SYNTAX_ERROR;
|
| - }
|
| - }
|
| - // 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 RTCErrorType::NONE;
|
| -}
|
| -
|
| -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()),
|
| - 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");
|
| - if (!allocator) {
|
| - LOG(LS_ERROR) << "PeerConnection initialized without a PortAllocator? "
|
| - << "This shouldn't happen if using PeerConnectionFactory.";
|
| - return false;
|
| - }
|
| - if (!observer) {
|
| - // TODO(deadbeef): Why do we do this?
|
| - LOG(LS_ERROR) << "PeerConnection initialized without a "
|
| - << "PeerConnectionObserver";
|
| - 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)),
|
| -#ifdef HAVE_SCTP
|
| - std::unique_ptr<cricket::SctpTransportInternalFactory>(
|
| - new cricket::SctpTransportFactory(factory_->network_thread()))
|
| -#else
|
| - nullptr
|
| -#endif
|
| - ));
|
| -
|
| - 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_->GetSctpSslRole(&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_->GetSctpSslRole(&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,
|
| - RTCError* error) {
|
| - 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 SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
|
| - }
|
| -
|
| - // The simplest (and most future-compatible) way to tell if the config was
|
| - // modified in an invalid way is to copy each property we do support
|
| - // modifying, then use operator==. There are far more properties we don't
|
| - // support modifying than those we do, and more could be added.
|
| - RTCConfiguration modified_config = configuration_;
|
| - modified_config.servers = configuration.servers;
|
| - modified_config.type = configuration.type;
|
| - modified_config.ice_candidate_pool_size =
|
| - configuration.ice_candidate_pool_size;
|
| - modified_config.prune_turn_ports = configuration.prune_turn_ports;
|
| - if (configuration != modified_config) {
|
| - LOG(LS_ERROR) << "Modifying the configuration in an unsupported way.";
|
| - return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
|
| - }
|
| -
|
| - // Note that this isn't possible through chromium, since it's an unsigned
|
| - // short in WebIDL.
|
| - if (configuration.ice_candidate_pool_size < 0 ||
|
| - configuration.ice_candidate_pool_size > UINT16_MAX) {
|
| - return SafeSetError(RTCErrorType::INVALID_RANGE, error);
|
| - }
|
| -
|
| - // Parse ICE servers before hopping to network thread.
|
| - cricket::ServerAddresses stun_servers;
|
| - std::vector<cricket::RelayServerConfig> turn_servers;
|
| - RTCErrorType parse_error =
|
| - ParseIceServers(configuration.servers, &stun_servers, &turn_servers);
|
| - if (parse_error != RTCErrorType::NONE) {
|
| - return SafeSetError(parse_error, error);
|
| - }
|
| -
|
| - // In theory this shouldn't fail.
|
| - if (!network_thread()->Invoke<bool>(
|
| - RTC_FROM_HERE,
|
| - rtc::Bind(&PeerConnection::ReconfigurePortAllocator_n, this,
|
| - stun_servers, turn_servers, modified_config.type,
|
| - modified_config.ice_candidate_pool_size,
|
| - modified_config.prune_turn_ports))) {
|
| - LOG(LS_ERROR) << "Failed to apply configuration to PortAllocator.";
|
| - return SafeSetError(RTCErrorType::INTERNAL_ERROR, error);
|
| - }
|
| -
|
| - // 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 (modified_config.servers != configuration_.servers ||
|
| - modified_config.type != configuration_.type ||
|
| - modified_config.prune_turn_ports != configuration_.prune_turn_ports) {
|
| - session_->SetNeedsIceRestartFlag();
|
| - }
|
| - configuration_ = modified_config;
|
| - return SafeSetError(RTCErrorType::NONE, error);
|
| -}
|
| -
|
| -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_->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();
|
| -}
|
| -
|
| -const SessionDescriptionInterface* PeerConnection::current_local_description()
|
| - const {
|
| - return session_->current_local_description();
|
| -}
|
| -
|
| -const SessionDescriptionInterface* PeerConnection::current_remote_description()
|
| - const {
|
| - return session_->current_remote_description();
|
| -}
|
| -
|
| -const SessionDescriptionInterface* PeerConnection::pending_local_description()
|
| - const {
|
| - return session_->pending_local_description();
|
| -}
|
| -
|
| -const SessionDescriptionInterface* PeerConnection::pending_remote_description()
|
| - const {
|
| - return session_->pending_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_NOTREACHED() << "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_NOTREACHED() << "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 {
|
| - RTC_NOTREACHED() << "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_->GetSctpSslRole(&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) !=
|
| - RTCErrorType::NONE) {
|
| - 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 cricket::ServerAddresses& stun_servers,
|
| - const std::vector<cricket::RelayServerConfig>& turn_servers,
|
| - IceTransportsType type,
|
| - int candidate_pool_size,
|
| - bool prune_turn_ports) {
|
| - port_allocator_->set_candidate_filter(
|
| - ConvertIceTransportTypeToCandidateFilter(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, candidate_pool_size, 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
|
|
|
Chromium Code Reviews
Issue 2514883002:
Create //webrtc/api:libjingle_peerconnection_api + refactorings. (Closed)
