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Unified Diff: talk/app/webrtc/peerconnection.cc

Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed processing of api.gyp for Chromium builds Created 4 years, 10 months ago
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Index: talk/app/webrtc/peerconnection.cc
diff --git a/talk/app/webrtc/peerconnection.cc b/talk/app/webrtc/peerconnection.cc
deleted file mode 100644
index c423b0fadec634c7cf9e7f1f3b5ce27d7030d118..0000000000000000000000000000000000000000
--- a/talk/app/webrtc/peerconnection.cc
+++ /dev/null
@@ -1,2091 +0,0 @@
-/*
- * libjingle
- * Copyright 2012 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/app/webrtc/peerconnection.h"
-
-#include <algorithm>
-#include <cctype> // for isdigit
-#include <utility>
-#include <vector>
-
-#include "talk/app/webrtc/audiotrack.h"
-#include "talk/app/webrtc/dtmfsender.h"
-#include "talk/app/webrtc/jsepicecandidate.h"
-#include "talk/app/webrtc/jsepsessiondescription.h"
-#include "talk/app/webrtc/mediaconstraintsinterface.h"
-#include "talk/app/webrtc/mediastream.h"
-#include "talk/app/webrtc/mediastreamobserver.h"
-#include "talk/app/webrtc/mediastreamproxy.h"
-#include "talk/app/webrtc/mediastreamtrackproxy.h"
-#include "talk/app/webrtc/remoteaudiosource.h"
-#include "talk/app/webrtc/remotevideocapturer.h"
-#include "talk/app/webrtc/rtpreceiver.h"
-#include "talk/app/webrtc/rtpsender.h"
-#include "talk/app/webrtc/streamcollection.h"
-#include "talk/app/webrtc/videosource.h"
-#include "talk/app/webrtc/videotrack.h"
-#include "talk/session/media/channelmanager.h"
-#include "webrtc/base/arraysize.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/stringencode.h"
-#include "webrtc/base/stringutils.h"
-#include "webrtc/base/trace_event.h"
-#include "webrtc/media/sctp/sctpdataengine.h"
-#include "webrtc/p2p/client/basicportallocator.h"
-#include "webrtc/system_wrappers/include/field_trial.h"
-
-namespace {
-
-using webrtc::DataChannel;
-using webrtc::MediaConstraintsInterface;
-using webrtc::MediaStreamInterface;
-using webrtc::PeerConnectionInterface;
-using webrtc::RtpSenderInterface;
-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"};
-
-// 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|.
-bool 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(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(uri_transport_param, '=', &tokens);
- if (tokens[0] == kTransport) {
- // As per above grammar transport param will be consist of lower case
- // letters.
- if (!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 false;
- }
- }
- }
-
- 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 false;
- }
-
- // 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 false;
- }
- if (tokens.size() == kTurnHostTokensNum) {
- if (tokens[0].empty() || tokens[1].empty()) {
- LOG(LS_WARNING) << "Invalid user@hostname format: " << hoststring;
- return false;
- }
- 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_TCP;
- }
-
- std::string address;
- if (!ParseHostnameAndPortFromString(hoststring, &address, &port)) {
- LOG(WARNING) << "Invalid hostname format: " << uri_without_transport;
- return false;
- }
-
- if (port <= 0 || port > 0xffff) {
- LOG(WARNING) << "Invalid port: " << port;
- return false;
- }
-
- switch (service_type) {
- case STUN:
- case STUNS:
- stun_servers->insert(rtc::SocketAddress(address, port));
- break;
- case TURN:
- case TURNS: {
- bool secure = (service_type == TURNS);
- turn_servers->push_back(
- cricket::RelayServerConfig(address, port, username, server.password,
- turn_transport_type, secure));
- break;
- }
- case INVALID:
- default:
- LOG(WARNING) << "Configuration not supported: " << url;
- return false;
- }
- return true;
-}
-
-// 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<RtpSenderInterface>>& 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->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);
- }
- }
-}
-
-} // namespace
-
-namespace webrtc {
-
-// Factory class for creating remote MediaStreams and MediaStreamTracks.
-class RemoteMediaStreamFactory {
- public:
- explicit RemoteMediaStreamFactory(rtc::Thread* signaling_thread,
- cricket::ChannelManager* channel_manager)
- : signaling_thread_(signaling_thread),
- channel_manager_(channel_manager) {}
-
- rtc::scoped_refptr<MediaStreamInterface> CreateMediaStream(
- const std::string& stream_label) {
- return MediaStreamProxy::Create(signaling_thread_,
- MediaStream::Create(stream_label));
- }
-
- AudioTrackInterface* AddAudioTrack(uint32_t ssrc,
- AudioProviderInterface* provider,
- webrtc::MediaStreamInterface* stream,
- const std::string& track_id) {
- return AddTrack<AudioTrackInterface, AudioTrack, AudioTrackProxy>(
- stream, track_id, RemoteAudioSource::Create(ssrc, provider));
- }
-
- VideoTrackInterface* AddVideoTrack(webrtc::MediaStreamInterface* stream,
- const std::string& track_id) {
- return AddTrack<VideoTrackInterface, VideoTrack, VideoTrackProxy>(
- stream, track_id,
- VideoSource::Create(channel_manager_, new RemoteVideoCapturer(),
- nullptr, true)
- .get());
- }
-
- private:
- template <typename TI, typename T, typename TP, typename S>
- TI* AddTrack(MediaStreamInterface* stream,
- const std::string& track_id,
- const S& source) {
- rtc::scoped_refptr<TI> track(
- TP::Create(signaling_thread_, T::Create(track_id, source)));
- track->set_state(webrtc::MediaStreamTrackInterface::kLive);
- if (stream->AddTrack(track)) {
- return track;
- }
- return nullptr;
- }
-
- rtc::Thread* signaling_thread_;
- cricket::ChannelManager* channel_manager_;
-};
-
-bool ConvertRtcOptionsForOffer(
- const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
- 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 (rtc_options.offer_to_receive_audio != RTCOfferAnswerOptions::kUndefined) {
- session_options->recv_audio = (rtc_options.offer_to_receive_audio > 0);
- }
- if (rtc_options.offer_to_receive_video != RTCOfferAnswerOptions::kUndefined) {
- session_options->recv_video = (rtc_options.offer_to_receive_video > 0);
- }
-
- session_options->vad_enabled = rtc_options.voice_activity_detection;
- session_options->audio_transport_options.ice_restart =
- rtc_options.ice_restart;
- session_options->video_transport_options.ice_restart =
- rtc_options.ice_restart;
- session_options->data_transport_options.ice_restart = rtc_options.ice_restart;
- session_options->bundle_enabled = rtc_options.use_rtp_mux;
-
- 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;
- }
-
- if (FindConstraint(constraints, MediaConstraintsInterface::kIceRestart,
- &value, &mandatory_constraints_satisfied)) {
- session_options->audio_transport_options.ice_restart = value;
- session_options->video_transport_options.ice_restart = value;
- session_options->data_transport_options.ice_restart = value;
- } else {
- // kIceRestart defaults to false according to spec.
- session_options->audio_transport_options.ice_restart = false;
- session_options->video_transport_options.ice_restart = false;
- session_options->data_transport_options.ice_restart = false;
- }
-
- if (!constraints) {
- return true;
- }
- return mandatory_constraints_satisfied == constraints->GetMandatory().size();
-}
-
-bool ParseIceServers(const PeerConnectionInterface::IceServers& servers,
- cricket::ServerAddresses* stun_servers,
- std::vector<cricket::RelayServerConfig>* turn_servers) {
- for (const webrtc::PeerConnectionInterface::IceServer& server : servers) {
- if (!server.urls.empty()) {
- for (const std::string& url : server.urls) {
- if (url.empty()) {
- LOG(LS_ERROR) << "Empty uri.";
- return false;
- }
- if (!ParseIceServerUrl(server, url, stun_servers, turn_servers)) {
- return false;
- }
- }
- } else if (!server.uri.empty()) {
- // Fallback to old .uri if new .urls isn't present.
- if (!ParseIceServerUrl(server, server.uri, stun_servers, turn_servers)) {
- return false;
- }
- } else {
- LOG(LS_ERROR) << "Empty uri.";
- return false;
- }
- }
- // Candidates must have unique priorities, so that connectivity checks
- // are performed in a well-defined order.
- int priority = static_cast<int>(turn_servers->size() - 1);
- for (cricket::RelayServerConfig& turn_server : *turn_servers) {
- // First in the list gets highest priority.
- turn_server.priority = priority--;
- }
- return true;
-}
-
-PeerConnection::PeerConnection(PeerConnectionFactory* factory)
- : factory_(factory),
- observer_(NULL),
- uma_observer_(NULL),
- signaling_state_(kStable),
- ice_state_(kIceNew),
- ice_connection_state_(kIceConnectionNew),
- ice_gathering_state_(kIceGatheringNew),
- 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->Stop();
- }
- for (const auto& receiver : receivers_) {
- receiver->Stop();
- }
-}
-
-bool PeerConnection::Initialize(
- const PeerConnectionInterface::RTCConfiguration& configuration,
- const MediaConstraintsInterface* constraints,
- rtc::scoped_ptr<cricket::PortAllocator> allocator,
- rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
- PeerConnectionObserver* observer) {
- TRACE_EVENT0("webrtc", "PeerConnection::Initialize");
- RTC_DCHECK(observer != nullptr);
- if (!observer) {
- return false;
- }
- observer_ = observer;
-
- port_allocator_ = std::move(allocator);
-
- cricket::ServerAddresses stun_servers;
- std::vector<cricket::RelayServerConfig> turn_servers;
- if (!ParseIceServers(configuration.servers, &stun_servers, &turn_servers)) {
- return false;
- }
- port_allocator_->SetIceServers(stun_servers, turn_servers);
-
- // 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;
- bool value;
- // If IPv6 flag was specified, we'll not override it by experiment.
- if (FindConstraint(constraints, MediaConstraintsInterface::kEnableIPv6,
- &value, nullptr)) {
- if (!value) {
- 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.";
- }
-
- port_allocator_->set_flags(portallocator_flags);
- // No step delay is used while allocating ports.
- port_allocator_->set_step_delay(cricket::kMinimumStepDelay);
-
- media_controller_.reset(factory_->CreateMediaController());
-
- remote_stream_factory_.reset(new RemoteMediaStreamFactory(
- factory_->signaling_thread(), media_controller_->channel_manager()));
-
- session_.reset(
- new WebRtcSession(media_controller_.get(), factory_->signaling_thread(),
- factory_->worker_thread(), port_allocator_.get()));
- stats_.reset(new StatsCollector(this));
-
- // Initialize the WebRtcSession. It creates transport channels etc.
- if (!session_->Initialize(factory_->options(), constraints,
- std::move(dtls_identity_store), 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_->SignalVoiceChannelDestroyed.connect(
- this, &PeerConnection::OnVoiceChannelDestroyed);
- session_->SignalVideoChannelDestroyed.connect(
- this, &PeerConnection::OnVideoChannelDestroyed);
- session_->SignalDataChannelCreated.connect(
- this, &PeerConnection::OnDataChannelCreated);
- session_->SignalDataChannelDestroyed.connect(
- this, &PeerConnection::OnDataChannelDestroyed);
- session_->SignalDataChannelOpenMessage.connect(
- this, &PeerConnection::OnDataChannelOpenMessage);
- 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(rtc::scoped_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 rtc::scoped_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<RtpSenderInterface> new_sender;
- if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
- new_sender = RtpSenderProxy::Create(
- signaling_thread(),
- new AudioRtpSender(static_cast<AudioTrackInterface*>(track),
- session_.get(), stats_.get()));
- if (!streams.empty()) {
- new_sender->set_stream_id(streams[0]->label());
- }
- const TrackInfo* track_info = FindTrackInfo(
- local_audio_tracks_, new_sender->stream_id(), track->id());
- if (track_info) {
- new_sender->SetSsrc(track_info->ssrc);
- }
- } else if (track->kind() == MediaStreamTrackInterface::kVideoKind) {
- new_sender = RtpSenderProxy::Create(
- signaling_thread(),
- new VideoRtpSender(static_cast<VideoTrackInterface*>(track),
- session_.get()));
- if (!streams.empty()) {
- new_sender->set_stream_id(streams[0]->label());
- }
- const TrackInfo* track_info = FindTrackInfo(
- local_video_tracks_, new_sender->stream_id(), track->id());
- if (track_info) {
- new_sender->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)->Stop();
- senders_.erase(it);
-
- observer_->OnRenegotiationNeeded();
- return true;
-}
-
-rtc::scoped_refptr<DtmfSenderInterface> PeerConnection::CreateDtmfSender(
- AudioTrackInterface* track) {
- TRACE_EVENT0("webrtc", "PeerConnection::CreateDtmfSender");
- 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");
- rtc::scoped_refptr<RtpSenderInterface> new_sender;
- if (kind == MediaStreamTrackInterface::kAudioKind) {
- new_sender = RtpSenderProxy::Create(
- signaling_thread(), new AudioRtpSender(session_.get(), stats_.get()));
- } else if (kind == MediaStreamTrackInterface::kVideoKind) {
- new_sender = RtpSenderProxy::Create(signaling_thread(),
- new VideoRtpSender(session_.get()));
- } else {
- LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
- return new_sender;
- }
- if (!stream_id.empty()) {
- new_sender->set_stream_id(stream_id);
- }
- senders_.push_back(new_sender);
- return new_sender;
-}
-
-std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
- const {
- return senders_;
-}
-
-std::vector<rtc::scoped_refptr<RtpReceiverInterface>>
-PeerConnection::GetReceivers() const {
- return receivers_;
-}
-
-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);
- signaling_thread()->Post(this, MSG_GETSTATS,
- new GetStatsMsg(observer, track));
- return true;
-}
-
-PeerConnectionInterface::SignalingState PeerConnection::signaling_state() {
- return signaling_state_;
-}
-
-PeerConnectionInterface::IceState PeerConnection::ice_state() {
- return ice_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");
- bool first_datachannel = !HasDataChannels();
-
- rtc::scoped_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, constraints, session_options);
-}
-
-void PeerConnection::SetLocalDescription(
- SetSessionDescriptionObserver* observer,
- SessionDescriptionInterface* desc) {
- TRACE_EVENT0("webrtc", "PeerConnection::SetLocalDescription");
- if (!VERIFY(observer != nullptr)) {
- LOG(LS_ERROR) << "SetLocalDescription - observer is NULL.";
- return;
- }
- if (!desc) {
- PostSetSessionDescriptionFailure(observer, "SessionDescription is NULL.");
- return;
- }
- // Update stats here so that we have the most recent stats for tracks and
- // streams that might be removed by updating the session description.
- stats_->UpdateStats(kStatsOutputLevelStandard);
- std::string error;
- if (!session_->SetLocalDescription(desc, &error)) {
- PostSetSessionDescriptionFailure(observer, error);
- return;
- }
-
- // If setting the description decided our SSL role, allocate any necessary
- // SCTP sids.
- rtc::SSLRole role;
- if (session_->data_channel_type() == cricket::DCT_SCTP &&
- session_->GetSslRole(session_->data_channel(), &role)) {
- AllocateSctpSids(role);
- }
-
- // Update state and SSRC of local MediaStreams and DataChannels based on the
- // local session description.
- const cricket::ContentInfo* audio_content =
- GetFirstAudioContent(desc->description());
- if (audio_content) {
- if (audio_content->rejected) {
- RemoveTracks(cricket::MEDIA_TYPE_AUDIO);
- } else {
- const cricket::AudioContentDescription* audio_desc =
- static_cast<const cricket::AudioContentDescription*>(
- audio_content->description);
- UpdateLocalTracks(audio_desc->streams(), audio_desc->type());
- }
- }
-
- const cricket::ContentInfo* video_content =
- GetFirstVideoContent(desc->description());
- if (video_content) {
- if (video_content->rejected) {
- RemoveTracks(cricket::MEDIA_TYPE_VIDEO);
- } else {
- const cricket::VideoContentDescription* video_desc =
- static_cast<const cricket::VideoContentDescription*>(
- video_content->description);
- UpdateLocalTracks(video_desc->streams(), video_desc->type());
- }
- }
-
- const cricket::ContentInfo* data_content =
- GetFirstDataContent(desc->description());
- if (data_content) {
- const cricket::DataContentDescription* data_desc =
- static_cast<const cricket::DataContentDescription*>(
- data_content->description);
- if (rtc::starts_with(data_desc->protocol().data(),
- cricket::kMediaProtocolRtpPrefix)) {
- UpdateLocalRtpDataChannels(data_desc->streams());
- }
- }
-
- SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
- signaling_thread()->Post(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 (!VERIFY(observer != nullptr)) {
- LOG(LS_ERROR) << "SetRemoteDescription - observer is NULL.";
- return;
- }
- if (!desc) {
- PostSetSessionDescriptionFailure(observer, "SessionDescription is NULL.");
- return;
- }
- // Update stats here so that we have the most recent stats for tracks and
- // streams that might be removed by updating the session description.
- stats_->UpdateStats(kStatsOutputLevelStandard);
- std::string error;
- if (!session_->SetRemoteDescription(desc, &error)) {
- PostSetSessionDescriptionFailure(observer, error);
- return;
- }
-
- // If setting the description decided our SSL role, allocate any necessary
- // SCTP sids.
- rtc::SSLRole role;
- if (session_->data_channel_type() == cricket::DCT_SCTP &&
- session_->GetSslRole(session_->data_channel(), &role)) {
- AllocateSctpSids(role);
- }
-
- const cricket::SessionDescription* remote_desc = desc->description();
- const cricket::ContentInfo* audio_content = GetFirstAudioContent(remote_desc);
- const cricket::ContentInfo* video_content = GetFirstVideoContent(remote_desc);
- const cricket::AudioContentDescription* audio_desc =
- GetFirstAudioContentDescription(remote_desc);
- const cricket::VideoContentDescription* video_desc =
- GetFirstVideoContentDescription(remote_desc);
- const cricket::DataContentDescription* data_desc =
- GetFirstDataContentDescription(remote_desc);
-
- // Check if the descriptions include streams, just in case the peer supports
- // MSID, but doesn't indicate so with "a=msid-semantic".
- if (remote_desc->msid_supported() ||
- (audio_desc && !audio_desc->streams().empty()) ||
- (video_desc && !video_desc->streams().empty())) {
- remote_peer_supports_msid_ = true;
- }
-
- // We wait to signal new streams until we finish processing the description,
- // since only at that point will new streams have all their tracks.
- rtc::scoped_refptr<StreamCollection> new_streams(StreamCollection::Create());
-
- // Find all audio rtp streams and create corresponding remote AudioTracks
- // and MediaStreams.
- if (audio_content) {
- if (audio_content->rejected) {
- RemoveTracks(cricket::MEDIA_TYPE_AUDIO);
- } else {
- bool default_audio_track_needed =
- !remote_peer_supports_msid_ &&
- MediaContentDirectionHasSend(audio_desc->direction());
- UpdateRemoteStreamsList(GetActiveStreams(audio_desc),
- default_audio_track_needed, audio_desc->type(),
- new_streams);
- }
- }
-
- // Find all video rtp streams and create corresponding remote VideoTracks
- // and MediaStreams.
- if (video_content) {
- if (video_content->rejected) {
- RemoveTracks(cricket::MEDIA_TYPE_VIDEO);
- } else {
- bool default_video_track_needed =
- !remote_peer_supports_msid_ &&
- MediaContentDirectionHasSend(video_desc->direction());
- UpdateRemoteStreamsList(GetActiveStreams(video_desc),
- default_video_track_needed, video_desc->type(),
- new_streams);
- }
- }
-
- // Update the DataChannels with the information from the remote peer.
- if (data_desc) {
- if (rtc::starts_with(data_desc->protocol().data(),
- cricket::kMediaProtocolRtpPrefix)) {
- UpdateRemoteRtpDataChannels(GetActiveStreams(data_desc));
- }
- }
-
- // Iterate new_streams and notify the observer about new MediaStreams.
- for (size_t i = 0; i < new_streams->count(); ++i) {
- MediaStreamInterface* new_stream = new_streams->at(i);
- stats_->AddStream(new_stream);
- observer_->OnAddStream(new_stream);
- }
-
- UpdateEndedRemoteMediaStreams();
-
- SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
- signaling_thread()->Post(this, MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
-}
-
-bool PeerConnection::SetConfiguration(const RTCConfiguration& config) {
- TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
- if (port_allocator_) {
- cricket::ServerAddresses stun_servers;
- std::vector<cricket::RelayServerConfig> turn_servers;
- if (!ParseIceServers(config.servers, &stun_servers, &turn_servers)) {
- return false;
- }
- port_allocator_->SetIceServers(stun_servers, turn_servers);
- }
- session_->SetIceConfig(session_->ParseIceConfig(config));
- return session_->SetIceTransports(config.type);
-}
-
-bool PeerConnection::AddIceCandidate(
- const IceCandidateInterface* ice_candidate) {
- TRACE_EVENT0("webrtc", "PeerConnection::AddIceCandidate");
- return session_->ProcessIceMessage(ice_candidate);
-}
-
-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_) {
- 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);
- }
- }
-}
-
-const SessionDescriptionInterface* PeerConnection::local_description() const {
- return session_->local_description();
-}
-
-const SessionDescriptionInterface* PeerConnection::remote_description() const {
- return session_->remote_description();
-}
-
-void PeerConnection::Close() {
- TRACE_EVENT0("webrtc", "PeerConnection::Close");
- // Update stats here so that we have the most recent stats for tracks and
- // streams before the channels are closed.
- stats_->UpdateStats(kStatsOutputLevelStandard);
-
- session_->Close();
-}
-
-void PeerConnection::OnSessionStateChange(WebRtcSession* /*session*/,
- WebRtcSession::State state) {
- switch (state) {
- case WebRtcSession::STATE_INIT:
- ChangeSignalingState(PeerConnectionInterface::kStable);
- break;
- case WebRtcSession::STATE_SENTOFFER:
- ChangeSignalingState(PeerConnectionInterface::kHaveLocalOffer);
- break;
- case WebRtcSession::STATE_SENTPRANSWER:
- ChangeSignalingState(PeerConnectionInterface::kHaveLocalPrAnswer);
- break;
- case WebRtcSession::STATE_RECEIVEDOFFER:
- ChangeSignalingState(PeerConnectionInterface::kHaveRemoteOffer);
- break;
- case WebRtcSession::STATE_RECEIVEDPRANSWER:
- ChangeSignalingState(PeerConnectionInterface::kHaveRemotePrAnswer);
- break;
- case WebRtcSession::STATE_INPROGRESS:
- ChangeSignalingState(PeerConnectionInterface::kStable);
- break;
- case WebRtcSession::STATE_CLOSED:
- ChangeSignalingState(PeerConnectionInterface::kClosed);
- break;
- default:
- break;
- }
-}
-
-void PeerConnection::OnMessage(rtc::Message* msg) {
- switch (msg->message_id) {
- case MSG_SET_SESSIONDESCRIPTION_SUCCESS: {
- SetSessionDescriptionMsg* param =
- static_cast<SetSessionDescriptionMsg*>(msg->pdata);
- param->observer->OnSuccess();
- delete param;
- break;
- }
- case MSG_SET_SESSIONDESCRIPTION_FAILED: {
- SetSessionDescriptionMsg* param =
- static_cast<SetSessionDescriptionMsg*>(msg->pdata);
- param->observer->OnFailure(param->error);
- delete param;
- break;
- }
- case MSG_CREATE_SESSIONDESCRIPTION_FAILED: {
- CreateSessionDescriptionMsg* param =
- static_cast<CreateSessionDescriptionMsg*>(msg->pdata);
- param->observer->OnFailure(param->error);
- delete param;
- break;
- }
- case MSG_GETSTATS: {
- GetStatsMsg* param = static_cast<GetStatsMsg*>(msg->pdata);
- StatsReports reports;
- stats_->GetStats(param->track, &reports);
- param->observer->OnComplete(reports);
- delete param;
- break;
- }
- case MSG_FREE_DATACHANNELS: {
- sctp_data_channels_to_free_.clear();
- break;
- }
- default:
- RTC_DCHECK(false && "Not implemented");
- break;
- }
-}
-
-void PeerConnection::CreateAudioReceiver(MediaStreamInterface* stream,
- AudioTrackInterface* audio_track,
- uint32_t ssrc) {
- receivers_.push_back(RtpReceiverProxy::Create(
- signaling_thread(),
- new AudioRtpReceiver(audio_track, ssrc, session_.get())));
-}
-
-void PeerConnection::CreateVideoReceiver(MediaStreamInterface* stream,
- VideoTrackInterface* video_track,
- uint32_t ssrc) {
- receivers_.push_back(RtpReceiverProxy::Create(
- signaling_thread(),
- new VideoRtpReceiver(video_track, ssrc, session_.get())));
-}
-
-// TODO(deadbeef): Keep RtpReceivers around even if track goes away in remote
-// description.
-void PeerConnection::DestroyAudioReceiver(MediaStreamInterface* stream,
- AudioTrackInterface* audio_track) {
- auto it = FindReceiverForTrack(audio_track);
- if (it == receivers_.end()) {
- LOG(LS_WARNING) << "RtpReceiver for track with id " << audio_track->id()
- << " doesn't exist.";
- } else {
- (*it)->Stop();
- receivers_.erase(it);
- }
-}
-
-void PeerConnection::DestroyVideoReceiver(MediaStreamInterface* stream,
- VideoTrackInterface* video_track) {
- auto it = FindReceiverForTrack(video_track);
- if (it == receivers_.end()) {
- LOG(LS_WARNING) << "RtpReceiver for track with id " << video_track->id()
- << " doesn't exist.";
- } else {
- (*it)->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());
- observer_->OnIceCandidate(candidate);
-}
-
-void PeerConnection::OnIceConnectionReceivingChange(bool receiving) {
- RTC_DCHECK(signaling_thread()->IsCurrent());
- 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) {
- 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)->set_stream_id(stream->label());
- return;
- }
-
- // Normal case; we've never seen this track before.
- rtc::scoped_refptr<RtpSenderInterface> new_sender = RtpSenderProxy::Create(
- signaling_thread(),
- new AudioRtpSender(track, stream->label(), session_.get(), 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->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) {
- auto sender = FindSenderForTrack(track);
- if (sender == senders_.end()) {
- LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
- << " doesn't exist.";
- return;
- }
- (*sender)->Stop();
- senders_.erase(sender);
-}
-
-void PeerConnection::OnVideoTrackAdded(VideoTrackInterface* track,
- MediaStreamInterface* stream) {
- 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)->set_stream_id(stream->label());
- return;
- }
-
- // Normal case; we've never seen this track before.
- rtc::scoped_refptr<RtpSenderInterface> new_sender = RtpSenderProxy::Create(
- signaling_thread(),
- new VideoRtpSender(track, stream->label(), session_.get()));
- senders_.push_back(new_sender);
- const TrackInfo* track_info =
- FindTrackInfo(local_video_tracks_, stream->label(), track->id());
- if (track_info) {
- new_sender->SetSsrc(track_info->ssrc);
- }
-}
-
-void PeerConnection::OnVideoTrackRemoved(VideoTrackInterface* track,
- MediaStreamInterface* stream) {
- auto sender = FindSenderForTrack(track);
- if (sender == senders_.end()) {
- LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
- << " doesn't exist.";
- return;
- }
- (*sender)->Stop();
- senders_.erase(sender);
-}
-
-void PeerConnection::PostSetSessionDescriptionFailure(
- SetSessionDescriptionObserver* observer,
- const std::string& error) {
- SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
- msg->error = error;
- signaling_thread()->Post(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(this, MSG_CREATE_SESSIONDESCRIPTION_FAILED, msg);
-}
-
-bool PeerConnection::GetOptionsForOffer(
- const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
- cricket::MediaSessionOptions* session_options) {
- if (!ConvertRtcOptionsForOffer(rtc_options, 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();
- }
- session_options->bundle_enabled =
- session_options->bundle_enabled &&
- (session_options->has_audio() || session_options->has_video() ||
- session_options->has_data());
-
- if (session_->data_channel_type() == cricket::DCT_SCTP && HasDataChannels()) {
- session_options->data_channel_type = cricket::DCT_SCTP;
- }
- return true;
-}
-
-bool PeerConnection::GetOptionsForAnswer(
- const MediaConstraintsInterface* constraints,
- cricket::MediaSessionOptions* session_options) {
- session_options->recv_audio = false;
- session_options->recv_video = false;
- if (!ParseConstraintsForAnswer(constraints, session_options)) {
- return false;
- }
-
- AddSendStreams(session_options, senders_, rtp_data_channels_);
- session_options->bundle_enabled =
- session_options->bundle_enabled &&
- (session_options->has_audio() || session_options->has_video() ||
- session_options->has_data());
-
- // 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.
- if (session_->data_channel_type() == cricket::DCT_SCTP) {
- session_options->data_channel_type = cricket::DCT_SCTP;
- }
- 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 = remote_stream_factory_->CreateMediaStream(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 =
- remote_stream_factory_->CreateMediaStream(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) {
- AudioTrackInterface* audio_track = remote_stream_factory_->AddAudioTrack(
- ssrc, session_.get(), stream, track_id);
- CreateAudioReceiver(stream, audio_track, ssrc);
- } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
- VideoTrackInterface* video_track =
- remote_stream_factory_->AddVideoTrack(stream, track_id);
- CreateVideoReceiver(stream, video_track, ssrc);
- } else {
- RTC_DCHECK(false && "Invalid media type");
- }
-}
-
-void PeerConnection::OnRemoteTrackRemoved(const std::string& stream_label,
- const std::string& track_id,
- cricket::MediaType media_type) {
- MediaStreamInterface* stream = remote_streams_->find(stream_label);
-
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
- rtc::scoped_refptr<AudioTrackInterface> audio_track =
- stream->FindAudioTrack(track_id);
- if (audio_track) {
- audio_track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
- stream->RemoveTrack(audio_track);
- DestroyAudioReceiver(stream, audio_track);
- }
- } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
- rtc::scoped_refptr<VideoTrackInterface> video_track =
- stream->FindVideoTrack(track_id);
- if (video_track) {
- video_track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
- stream->RemoveTrack(video_track);
- DestroyVideoReceiver(stream, video_track);
- }
- } else {
- ASSERT(false && "Invalid media type");
- }
-}
-
-void PeerConnection::UpdateEndedRemoteMediaStreams() {
- std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_to_remove;
- for (size_t i = 0; i < remote_streams_->count(); ++i) {
- MediaStreamInterface* stream = remote_streams_->at(i);
- if (stream->GetAudioTracks().empty() && stream->GetVideoTracks().empty()) {
- streams_to_remove.push_back(stream);
- }
- }
-
- for (const auto& stream : streams_to_remove) {
- remote_streams_->RemoveStream(stream);
- observer_->OnRemoveStream(stream);
- }
-}
-
-void PeerConnection::EndRemoteTracks(cricket::MediaType media_type) {
- TrackInfos* current_tracks = GetRemoteTracks(media_type);
- for (TrackInfos::iterator track_it = current_tracks->begin();
- track_it != current_tracks->end(); ++track_it) {
- const TrackInfo& info = *track_it;
- MediaStreamInterface* stream = remote_streams_->find(info.stream_label);
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
- AudioTrackInterface* track = stream->FindAudioTrack(info.track_id);
- // There's no guarantee the track is still available, e.g. the track may
- // have been removed from the stream by javascript.
- if (track) {
- track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
- }
- }
- if (media_type == cricket::MEDIA_TYPE_VIDEO) {
- VideoTrackInterface* track = stream->FindVideoTrack(info.track_id);
- // There's no guarantee the track is still available, e.g. the track may
- // have been removed from the stream by javascript.
- if (track) {
- track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
- }
- }
- }
-}
-
-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) {
- RtpSenderInterface* 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) {
- RtpSenderInterface* 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);
- observer_->OnDataChannel(
- DataChannelProxy::Create(signaling_thread(), channel));
-}
-
-rtc::scoped_refptr<DataChannel> PeerConnection::InternalCreateDataChannel(
- const std::string& label,
- const InternalDataChannelInit* config) {
- if (IsClosed()) {
- return nullptr;
- }
- if (session_->data_channel_type() == cricket::DCT_NONE) {
- LOG(LS_ERROR)
- << "InternalCreateDataChannel: Data is not supported in this call.";
- return nullptr;
- }
- InternalDataChannelInit new_config =
- config ? (*config) : InternalDataChannelInit();
- if (session_->data_channel_type() == cricket::DCT_SCTP) {
- if (new_config.id < 0) {
- rtc::SSLRole role;
- if ((session_->GetSslRole(session_->data_channel(), &role)) &&
- !sid_allocator_.AllocateSid(role, &new_config.id)) {
- LOG(LS_ERROR) << "No id can be allocated for the SCTP data channel.";
- return nullptr;
- }
- } else if (!sid_allocator_.ReserveSid(new_config.id)) {
- LOG(LS_ERROR) << "Failed to create a SCTP data channel "
- << "because the id is already in use or out of range.";
- return nullptr;
- }
- }
-
- rtc::scoped_refptr<DataChannel> channel(DataChannel::Create(
- session_.get(), session_->data_channel_type(), label, new_config));
- if (!channel) {
- sid_allocator_.ReleaseSid(new_config.id);
- return nullptr;
- }
-
- if (channel->data_channel_type() == cricket::DCT_RTP) {
- if (rtp_data_channels_.find(channel->label()) != rtp_data_channels_.end()) {
- LOG(LS_ERROR) << "DataChannel with label " << channel->label()
- << " already exists.";
- return nullptr;
- }
- rtp_data_channels_[channel->label()] = channel;
- } else {
- RTC_DCHECK(channel->data_channel_type() == cricket::DCT_SCTP);
- sctp_data_channels_.push_back(channel);
- channel->SignalClosed.connect(this,
- &PeerConnection::OnSctpDataChannelClosed);
- }
-
- return channel;
-}
-
-bool PeerConnection::HasDataChannels() const {
- return !rtp_data_channels_.empty() || !sctp_data_channels_.empty();
-}
-
-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(this, MSG_FREE_DATACHANNELS, nullptr);
- return;
- }
- }
-}
-
-void PeerConnection::OnVoiceChannelDestroyed() {
- EndRemoteTracks(cricket::MEDIA_TYPE_AUDIO);
-}
-
-void PeerConnection::OnVideoChannelDestroyed() {
- EndRemoteTracks(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;
- }
-
- observer_->OnDataChannel(
- DataChannelProxy::Create(signaling_thread(), channel));
-}
-
-RtpSenderInterface* PeerConnection::FindSenderById(const std::string& id) {
- auto it =
- std::find_if(senders_.begin(), senders_.end(),
- [id](const rtc::scoped_refptr<RtpSenderInterface>& sender) {
- return sender->id() == id;
- });
- return it != senders_.end() ? it->get() : nullptr;
-}
-
-std::vector<rtc::scoped_refptr<RtpSenderInterface>>::iterator
-PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) {
- return std::find_if(
- senders_.begin(), senders_.end(),
- [track](const rtc::scoped_refptr<RtpSenderInterface>& sender) {
- return sender->track() == track;
- });
-}
-
-std::vector<rtc::scoped_refptr<RtpReceiverInterface>>::iterator
-PeerConnection::FindReceiverForTrack(MediaStreamTrackInterface* track) {
- return std::find_if(
- receivers_.begin(), receivers_.end(),
- [track](const rtc::scoped_refptr<RtpReceiverInterface>& receiver) {
- return receiver->track() == track;
- });
-}
-
-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;
-}
-
-} // namespace webrtc
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