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webrtc / src / 4b2be9739b6fff561e2477b9c3b4a47433116ccc / . / webrtc / api / peerconnection.cc
blob: bdd33d7d371003d15197f10579413fb253421263 [file] [log] [blame]
/*
* 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/logging.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/call.h"
#include "webrtc/media/sctp/sctpdataengine.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::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|.
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<
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:
ASSERT(false);
}
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);
}
}
}
} // namespace
namespace webrtc {
// 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_DCHECK(false);
}
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();
}
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),
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);
// Now destroy session_ before destroying other members,
// because its destruction fires signals (such as VoiceChannelDestroyed)
// which will trigger some final actions in PeerConnection...
session_.reset(nullptr);
// port_allocator_ lives on the network thread and should be destroyed there.
network_thread()->Invoke<void>(RTC_FROM_HERE,
[this] { port_allocator_.reset(nullptr); });
}
bool PeerConnection::Initialize(
const PeerConnectionInterface::RTCConfiguration& configuration,
std::unique_ptr<cricket::PortAllocator> allocator,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
PeerConnectionObserver* observer) {
TRACE_EVENT0("webrtc", "PeerConnection::Initialize");
RTC_DCHECK(observer != nullptr);
if (!observer) {
return false;
}
observer_ = observer;
port_allocator_ = std::move(allocator);
// The port allocator lives on the network thread and should be initialized
// there.
if (!network_thread()->Invoke<bool>(
RTC_FROM_HERE, rtc::Bind(&PeerConnection::InitializePortAllocator_n,
this, configuration))) {
return false;
}
media_controller_.reset(
factory_->CreateMediaController(configuration.media_config));
session_.reset(new WebRtcSession(
media_controller_.get(), factory_->network_thread(),
factory_->worker_thread(), factory_->signaling_thread(),
port_allocator_.get(),
std::unique_ptr<cricket::TransportController>(
factory_->CreateTransportController(
port_allocator_.get(),
configuration.redetermine_role_on_ice_restart))));
stats_.reset(new StatsCollector(this));
enable_ice_renomination_ = configuration.enable_ice_renomination;
// 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);
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);
signaling_thread()->Post(RTC_FROM_HERE, 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");
#ifdef HAVE_QUIC
if (session_->data_channel_type() == cricket::DCT_QUIC) {
// TODO(zhihuang): Handle case when config is NULL.
if (!config) {
LOG(LS_ERROR) << "Missing config for QUIC data channel.";
return nullptr;
}
// TODO(zhihuang): Allow unreliable or ordered QUIC data channels.
if (!config->reliable || config->ordered) {
LOG(LS_ERROR) << "QUIC data channel does not implement unreliable or "
"ordered delivery.";
return nullptr;
}
return session_->quic_data_transport()->CreateDataChannel(label, config);
}
#endif // HAVE_QUIC
bool first_datachannel = !HasDataChannels();
std::unique_ptr<InternalDataChannelInit> internal_config;
if (config) {
internal_config.reset(new InternalDataChannelInit(*config));
}
rtc::scoped_refptr<DataChannelInterface> channel(
InternalCreateDataChannel(label, internal_config.get()));
if (!channel.get()) {
return nullptr;
}
// Trigger the onRenegotiationNeeded event for every new RTP DataChannel, or
// the first SCTP DataChannel.
if (session_->data_channel_type() == cricket::DCT_RTP || first_datachannel) {
observer_->OnRenegotiationNeeded();
}
return DataChannelProxy::Create(signaling_thread(), channel.get());
}
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
if (!VERIFY(observer != nullptr)) {
LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
return;
}
RTCOfferAnswerOptions options;
bool value;
size_t mandatory_constraints = 0;
if (FindConstraint(constraints,
MediaConstraintsInterface::kOfferToReceiveAudio,
&value,
&mandatory_constraints)) {
options.offer_to_receive_audio =
value ? RTCOfferAnswerOptions::kOfferToReceiveMediaTrue : 0;
}
if (FindConstraint(constraints,
MediaConstraintsInterface::kOfferToReceiveVideo,
&value,
&mandatory_constraints)) {
options.offer_to_receive_video =
value ? RTCOfferAnswerOptions::kOfferToReceiveMediaTrue : 0;
}
if (FindConstraint(constraints,
MediaConstraintsInterface::kVoiceActivityDetection,
&value,
&mandatory_constraints)) {
options.voice_activity_detection = value;
}
if (FindConstraint(constraints,
MediaConstraintsInterface::kIceRestart,
&value,
&mandatory_constraints)) {
options.ice_restart = value;
}
if (FindConstraint(constraints,
MediaConstraintsInterface::kUseRtpMux,
&value,
&mandatory_constraints)) {
options.use_rtp_mux = value;
}
CreateOffer(observer, options);
}
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
if (!VERIFY(observer != nullptr)) {
LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
return;
}
cricket::MediaSessionOptions session_options;
if (!GetOptionsForOffer(options, &session_options)) {
std::string error = "CreateOffer called with invalid options.";
LOG(LS_ERROR) << error;
PostCreateSessionDescriptionFailure(observer, error);
return;
}
session_->CreateOffer(observer, options, session_options);
}
void PeerConnection::CreateAnswer(
CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateAnswer");
if (!VERIFY(observer != nullptr)) {
LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
return;
}
cricket::MediaSessionOptions session_options;
if (!GetOptionsForAnswer(constraints, &session_options)) {
std::string error = "CreateAnswer called with invalid constraints.";
LOG(LS_ERROR) << error;
PostCreateSessionDescriptionFailure(observer, error);
return;
}
session_->CreateAnswer(observer, session_options);
}
void PeerConnection::CreateAnswer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateAnswer");
if (!VERIFY(observer != nullptr)) {
LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
return;
}
cricket::MediaSessionOptions session_options;
if (!GetOptionsForAnswer(options, &session_options)) {
std::string error = "CreateAnswer called with invalid options.";
LOG(LS_ERROR) << error;
PostCreateSessionDescriptionFailure(observer, error);
return;
}
session_->CreateAnswer(observer, session_options);
}
void PeerConnection::SetLocalDescription(
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) {
TRACE_EVENT0("webrtc", "PeerConnection::SetLocalDescription");
if (IsClosed()) {
return;
}
if (!VERIFY(observer != nullptr)) {
LOG(LS_ERROR) << "SetLocalDescription - observer is NULL.";
return;
}
if (!desc) {
PostSetSessionDescriptionFailure(observer, "SessionDescription is NULL.");
return;
}
// Update stats here so that we have the most recent stats for tracks and
// streams that might be removed by updating the session description.
stats_->UpdateStats(kStatsOutputLevelStandard);
std::string error;
if (!session_->SetLocalDescription(desc, &error)) {
PostSetSessionDescriptionFailure(observer, error);
return;
}
// If setting the description decided our SSL role, allocate any necessary
// SCTP sids.
rtc::SSLRole role;
if (session_->data_channel_type() == cricket::DCT_SCTP &&
session_->GetSslRole(session_->data_channel(), &role)) {
AllocateSctpSids(role);
}
// Update state and SSRC of local MediaStreams and DataChannels based on the
// local session description.
const cricket::ContentInfo* audio_content =
GetFirstAudioContent(desc->description());
if (audio_content) {
if (audio_content->rejected) {
RemoveTracks(cricket::MEDIA_TYPE_AUDIO);
} else {
const cricket::AudioContentDescription* audio_desc =
static_cast<const cricket::AudioContentDescription*>(
audio_content->description);
UpdateLocalTracks(audio_desc->streams(), audio_desc->type());
}
}
const cricket::ContentInfo* video_content =
GetFirstVideoContent(desc->description());
if (video_content) {
if (video_content->rejected) {
RemoveTracks(cricket::MEDIA_TYPE_VIDEO);
} else {
const cricket::VideoContentDescription* video_desc =
static_cast<const cricket::VideoContentDescription*>(
video_content->description);
UpdateLocalTracks(video_desc->streams(), video_desc->type());
}
}
const cricket::ContentInfo* data_content =
GetFirstDataContent(desc->description());
if (data_content) {
const cricket::DataContentDescription* data_desc =
static_cast<const cricket::DataContentDescription*>(
data_content->description);
if (rtc::starts_with(data_desc->protocol().data(),
cricket::kMediaProtocolRtpPrefix)) {
UpdateLocalRtpDataChannels(data_desc->streams());
}
}
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
signaling_thread()->Post(RTC_FROM_HERE, this,
MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
// MaybeStartGathering needs to be called after posting
// MSG_SET_SESSIONDESCRIPTION_SUCCESS, so that we don't signal any candidates
// before signaling that SetLocalDescription completed.
session_->MaybeStartGathering();
}
void PeerConnection::SetRemoteDescription(
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) {
TRACE_EVENT0("webrtc", "PeerConnection::SetRemoteDescription");
if (IsClosed()) {
return;
}
if (!VERIFY(observer != nullptr)) {
LOG(LS_ERROR) << "SetRemoteDescription - observer is NULL.";
return;
}
if (!desc) {
PostSetSessionDescriptionFailure(observer, "SessionDescription is NULL.");
return;
}
// Update stats here so that we have the most recent stats for tracks and
// streams that might be removed by updating the session description.
stats_->UpdateStats(kStatsOutputLevelStandard);
std::string error;
if (!session_->SetRemoteDescription(desc, &error)) {
PostSetSessionDescriptionFailure(observer, error);
return;
}
// If setting the description decided our SSL role, allocate any necessary
// SCTP sids.
rtc::SSLRole role;
if (session_->data_channel_type() == cricket::DCT_SCTP &&
session_->GetSslRole(session_->data_channel(), &role)) {
AllocateSctpSids(role);
}
const cricket::SessionDescription* remote_desc = desc->description();
const cricket::ContentInfo* audio_content = GetFirstAudioContent(remote_desc);
const cricket::ContentInfo* video_content = GetFirstVideoContent(remote_desc);
const cricket::AudioContentDescription* audio_desc =
GetFirstAudioContentDescription(remote_desc);
const cricket::VideoContentDescription* video_desc =
GetFirstVideoContentDescription(remote_desc);
const cricket::DataContentDescription* data_desc =
GetFirstDataContentDescription(remote_desc);
// Check if the descriptions include streams, just in case the peer supports
// MSID, but doesn't indicate so with "a=msid-semantic".
if (remote_desc->msid_supported() ||
(audio_desc && !audio_desc->streams().empty()) ||
(video_desc && !video_desc->streams().empty())) {
remote_peer_supports_msid_ = true;
}
// We wait to signal new streams until we finish processing the description,
// since only at that point will new streams have all their tracks.
rtc::scoped_refptr<StreamCollection> new_streams(StreamCollection::Create());
// Find all audio rtp streams and create corresponding remote AudioTracks
// and MediaStreams.
if (audio_content) {
if (audio_content->rejected) {
RemoveTracks(cricket::MEDIA_TYPE_AUDIO);
} else {
bool default_audio_track_needed =
!remote_peer_supports_msid_ &&
MediaContentDirectionHasSend(audio_desc->direction());
UpdateRemoteStreamsList(GetActiveStreams(audio_desc),
default_audio_track_needed, audio_desc->type(),
new_streams);
}
}
// Find all video rtp streams and create corresponding remote VideoTracks
// and MediaStreams.
if (video_content) {
if (video_content->rejected) {
RemoveTracks(cricket::MEDIA_TYPE_VIDEO);
} else {
bool default_video_track_needed =
!remote_peer_supports_msid_ &&
MediaContentDirectionHasSend(video_desc->direction());
UpdateRemoteStreamsList(GetActiveStreams(video_desc),
default_video_track_needed, video_desc->type(),
new_streams);
}
}
// Update the DataChannels with the information from the remote peer.
if (data_desc) {
if (rtc::starts_with(data_desc->protocol().data(),
cricket::kMediaProtocolRtpPrefix)) {
UpdateRemoteRtpDataChannels(GetActiveStreams(data_desc));
}
}
// Iterate new_streams and notify the observer about new MediaStreams.
for (size_t i = 0; i < new_streams->count(); ++i) {
MediaStreamInterface* new_stream = new_streams->at(i);
stats_->AddStream(new_stream);
// Call both the raw pointer and scoped_refptr versions of the method
// for compatibility.
observer_->OnAddStream(new_stream);
observer_->OnAddStream(
rtc::scoped_refptr<MediaStreamInterface>(new_stream));
}
UpdateEndedRemoteMediaStreams();
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
signaling_thread()->Post(RTC_FROM_HERE, this,
MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
}
bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration) {
TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
if (port_allocator_) {
if (!network_thread()->Invoke<bool>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnection::ReconfigurePortAllocator_n, this,
configuration))) {
return false;
}
}
// TODO(deadbeef): Shouldn't have to hop to the worker thread twice...
session_->SetIceConfig(session_->ParseIceConfig(configuration));
enable_ice_renomination_ = configuration.enable_ice_renomination;
return true;
}
bool PeerConnection::AddIceCandidate(
const IceCandidateInterface* ice_candidate) {
TRACE_EVENT0("webrtc", "PeerConnection::AddIceCandidate");
if (IsClosed()) {
return false;
}
return session_->ProcessIceMessage(ice_candidate);
}
bool PeerConnection::RemoveIceCandidates(
const std::vector<cricket::Candidate>& candidates) {
TRACE_EVENT0("webrtc", "PeerConnection::RemoveIceCandidates");
return session_->RemoveRemoteIceCandidates(candidates);
}
void PeerConnection::RegisterUMAObserver(UMAObserver* observer) {
TRACE_EVENT0("webrtc", "PeerConnection::RegisterUmaObserver");
uma_observer_ = observer;
if (session_) {
session_->set_metrics_observer(uma_observer_);
}
// Send information about IPv4/IPv6 status.
if (uma_observer_ && port_allocator_) {
if (port_allocator_->flags() & cricket::PORTALLOCATOR_ENABLE_IPV6) {
uma_observer_->IncrementEnumCounter(
kEnumCounterAddressFamily, kPeerConnection_IPv6,
kPeerConnectionAddressFamilyCounter_Max);
} else {
uma_observer_->IncrementEnumCounter(
kEnumCounterAddressFamily, kPeerConnection_IPv4,
kPeerConnectionAddressFamilyCounter_Max);
}
}
}
bool PeerConnection::StartRtcEventLog(rtc::PlatformFile file,
int64_t max_size_bytes) {
return factory_->worker_thread()->Invoke<bool>(
RTC_FROM_HERE, rtc::Bind(&PeerConnection::StartRtcEventLog_w, this, file,
max_size_bytes));
}
void PeerConnection::StopRtcEventLog() {
factory_->worker_thread()->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&PeerConnection::StopRtcEventLog_w, this));
}
const SessionDescriptionInterface* PeerConnection::local_description() const {
return session_->local_description();
}
const SessionDescriptionInterface* PeerConnection::remote_description() const {
return session_->remote_description();
}
void PeerConnection::Close() {
TRACE_EVENT0("webrtc", "PeerConnection::Close");
// Update stats here so that we have the most recent stats for tracks and
// streams before the channels are closed.
stats_->UpdateStats(kStatsOutputLevelStandard);
session_->Close();
}
void PeerConnection::OnSessionStateChange(WebRtcSession* /*session*/,
WebRtcSession::State state) {
switch (state) {
case WebRtcSession::STATE_INIT:
ChangeSignalingState(PeerConnectionInterface::kStable);
break;
case WebRtcSession::STATE_SENTOFFER:
ChangeSignalingState(PeerConnectionInterface::kHaveLocalOffer);
break;
case WebRtcSession::STATE_SENTPRANSWER:
ChangeSignalingState(PeerConnectionInterface::kHaveLocalPrAnswer);
break;
case WebRtcSession::STATE_RECEIVEDOFFER:
ChangeSignalingState(PeerConnectionInterface::kHaveRemoteOffer);
break;
case WebRtcSession::STATE_RECEIVEDPRANSWER:
ChangeSignalingState(PeerConnectionInterface::kHaveRemotePrAnswer);
break;
case WebRtcSession::STATE_INPROGRESS:
ChangeSignalingState(PeerConnectionInterface::kStable);
break;
case WebRtcSession::STATE_CLOSED:
ChangeSignalingState(PeerConnectionInterface::kClosed);
break;
default:
break;
}
}
void PeerConnection::OnMessage(rtc::Message* msg) {
switch (msg->message_id) {
case MSG_SET_SESSIONDESCRIPTION_SUCCESS: {
SetSessionDescriptionMsg* param =
static_cast<SetSessionDescriptionMsg*>(msg->pdata);
param->observer->OnSuccess();
delete param;
break;
}
case MSG_SET_SESSIONDESCRIPTION_FAILED: {
SetSessionDescriptionMsg* param =
static_cast<SetSessionDescriptionMsg*>(msg->pdata);
param->observer->OnFailure(param->error);
delete param;
break;
}
case MSG_CREATE_SESSIONDESCRIPTION_FAILED: {
CreateSessionDescriptionMsg* param =
static_cast<CreateSessionDescriptionMsg*>(msg->pdata);
param->observer->OnFailure(param->error);
delete param;
break;
}
case MSG_GETSTATS: {
GetStatsMsg* param = static_cast<GetStatsMsg*>(msg->pdata);
StatsReports reports;
stats_->GetStats(param->track, &reports);
param->observer->OnComplete(reports);
delete param;
break;
}
case MSG_FREE_DATACHANNELS: {
sctp_data_channels_to_free_.clear();
break;
}
default:
RTC_DCHECK(false && "Not implemented");
break;
}
}
void PeerConnection::CreateAudioReceiver(MediaStreamInterface* stream,
const std::string& track_id,
uint32_t ssrc) {
receivers_.push_back(
RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(), new AudioRtpReceiver(stream, track_id, ssrc,
session_->voice_channel())));
}
void PeerConnection::CreateVideoReceiver(MediaStreamInterface* stream,
const std::string& track_id,
uint32_t ssrc) {
receivers_.push_back(
RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(),
new VideoRtpReceiver(stream, track_id, factory_->worker_thread(),
ssrc, session_->video_channel())));
}
// 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 = 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();
}
session_options->bundle_enabled =
session_options->bundle_enabled &&
(session_options->has_audio() || session_options->has_video() ||
session_options->has_data());
// 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->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 = 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_);
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.
// 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->crypto_options = factory_->options().crypto_options;
}
bool PeerConnection::GetOptionsForAnswer(
const MediaConstraintsInterface* constraints,
cricket::MediaSessionOptions* session_options) {
InitializeOptionsForAnswer(session_options);
if (!ParseConstraintsForAnswer(constraints, session_options)) {
return false;
}
session_options->rtcp_cname = rtcp_cname_;
FinishOptionsForAnswer(session_options);
return true;
}
bool PeerConnection::GetOptionsForAnswer(
const RTCOfferAnswerOptions& options,
cricket::MediaSessionOptions* session_options) {
InitializeOptionsForAnswer(session_options);
if (!ExtractMediaSessionOptions(options, false, session_options)) {
return false;
}
session_options->rtcp_cname = rtcp_cname_;
FinishOptionsForAnswer(session_options);
return true;
}
void PeerConnection::RemoveTracks(cricket::MediaType media_type) {
UpdateLocalTracks(std::vector<cricket::StreamParams>(), media_type);
UpdateRemoteStreamsList(std::vector<cricket::StreamParams>(), false,
media_type, nullptr);
}
void PeerConnection::UpdateRemoteStreamsList(
const cricket::StreamParamsVec& streams,
bool default_track_needed,
cricket::MediaType media_type,
StreamCollection* new_streams) {
TrackInfos* current_tracks = GetRemoteTracks(media_type);
// Find removed tracks. I.e., tracks where the track id or ssrc don't match
// the new StreamParam.
auto track_it = current_tracks->begin();
while (track_it != current_tracks->end()) {
const TrackInfo& info = *track_it;
const cricket::StreamParams* params =
cricket::GetStreamBySsrc(streams, info.ssrc);
bool track_exists = params && params->id == info.track_id;
// If this is a default track, and we still need it, don't remove it.
if ((info.stream_label == kDefaultStreamLabel && default_track_needed) ||
track_exists) {
++track_it;
} else {
OnRemoteTrackRemoved(info.stream_label, info.track_id, media_type);
track_it = current_tracks->erase(track_it);
}
}
// Find new and active tracks.
for (const cricket::StreamParams& params : streams) {
// The sync_label is the MediaStream label and the |stream.id| is the
// track id.
const std::string& stream_label = params.sync_label;
const std::string& track_id = params.id;
uint32_t ssrc = params.first_ssrc();
rtc::scoped_refptr<MediaStreamInterface> stream =
remote_streams_->find(stream_label);
if (!stream) {
// This is a new MediaStream. Create a new remote MediaStream.
stream = MediaStreamProxy::Create(rtc::Thread::Current(),
MediaStream::Create(stream_label));
remote_streams_->AddStream(stream);
new_streams->AddStream(stream);
}
const TrackInfo* track_info =
FindTrackInfo(*current_tracks, stream_label, track_id);
if (!track_info) {
current_tracks->push_back(TrackInfo(stream_label, track_id, ssrc));
OnRemoteTrackSeen(stream_label, track_id, ssrc, media_type);
}
}
// Add default track if necessary.
if (default_track_needed) {
rtc::scoped_refptr<MediaStreamInterface> default_stream =
remote_streams_->find(kDefaultStreamLabel);
if (!default_stream) {
// Create the new default MediaStream.
default_stream = MediaStreamProxy::Create(
rtc::Thread::Current(), MediaStream::Create(kDefaultStreamLabel));
remote_streams_->AddStream(default_stream);
new_streams->AddStream(default_stream);
}
std::string default_track_id = (media_type == cricket::MEDIA_TYPE_AUDIO)
? kDefaultAudioTrackLabel
: kDefaultVideoTrackLabel;
const TrackInfo* default_track_info =
FindTrackInfo(*current_tracks, kDefaultStreamLabel, default_track_id);
if (!default_track_info) {
current_tracks->push_back(
TrackInfo(kDefaultStreamLabel, default_track_id, 0));
OnRemoteTrackSeen(kDefaultStreamLabel, default_track_id, 0, media_type);
}
}
}
void PeerConnection::OnRemoteTrackSeen(const std::string& stream_label,
const std::string& track_id,
uint32_t ssrc,
cricket::MediaType media_type) {
MediaStreamInterface* stream = remote_streams_->find(stream_label);
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
CreateAudioReceiver(stream, track_id, ssrc);
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
CreateVideoReceiver(stream, track_id, ssrc);
} else {
RTC_DCHECK(false && "Invalid media type");
}
}
void PeerConnection::OnRemoteTrackRemoved(const std::string& stream_label,
const std::string& track_id,
cricket::MediaType media_type) {
MediaStreamInterface* stream = remote_streams_->find(stream_label);
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
// When the MediaEngine audio channel is destroyed, the RemoteAudioSource
// will be notified which will end the AudioRtpReceiver::track().
DestroyReceiver(track_id);
rtc::scoped_refptr<AudioTrackInterface> audio_track =
stream->FindAudioTrack(track_id);
if (audio_track) {
stream->RemoveTrack(audio_track);
}
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
// Stopping or destroying a VideoRtpReceiver will end the
// VideoRtpReceiver::track().
DestroyReceiver(track_id);
rtc::scoped_refptr<VideoTrackInterface> video_track =
stream->FindVideoTrack(track_id);
if (video_track) {
// There's no guarantee the track is still available, e.g. the track may
// have been removed from the stream by an application.
stream->RemoveTrack(video_track);
}
} else {
ASSERT(false && "Invalid media type");
}
}
void PeerConnection::UpdateEndedRemoteMediaStreams() {
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_to_remove;
for (size_t i = 0; i < remote_streams_->count(); ++i) {
MediaStreamInterface* stream = remote_streams_->at(i);
if (stream->GetAudioTracks().empty() && stream->GetVideoTracks().empty()) {
streams_to_remove.push_back(stream);
}
}
for (auto& stream : streams_to_remove) {
remote_streams_->RemoveStream(stream);
// Call both the raw pointer and scoped_refptr versions of the method
// for compatibility.
observer_->OnRemoveStream(stream.get());
observer_->OnRemoveStream(std::move(stream));
}
}
void PeerConnection::UpdateLocalTracks(
const std::vector<cricket::StreamParams>& streams,
cricket::MediaType media_type) {
TrackInfos* current_tracks = GetLocalTracks(media_type);
// Find removed tracks. I.e., tracks where the track id, stream label or ssrc
// don't match the new StreamParam.
TrackInfos::iterator track_it = current_tracks->begin();
while (track_it != current_tracks->end()) {
const TrackInfo& info = *track_it;
const cricket::StreamParams* params =
cricket::GetStreamBySsrc(streams, info.ssrc);
if (!params || params->id != info.track_id ||
params->sync_label != info.stream_label) {
OnLocalTrackRemoved(info.stream_label, info.track_id, info.ssrc,
media_type);
track_it = current_tracks->erase(track_it);
} else {
++track_it;
}
}
// Find new and active tracks.
for (const cricket::StreamParams& params : streams) {
// The sync_label is the MediaStream label and the |stream.id| is the
// track id.
const std::string& stream_label = params.sync_label;
const std::string& track_id = params.id;
uint32_t ssrc = params.first_ssrc();
const TrackInfo* track_info =
FindTrackInfo(*current_tracks, stream_label, track_id);
if (!track_info) {
current_tracks->push_back(TrackInfo(stream_label, track_id, ssrc));
OnLocalTrackSeen(stream_label, track_id, params.first_ssrc(), media_type);
}
}
}
void PeerConnection::OnLocalTrackSeen(const std::string& stream_label,
const std::string& track_id,
uint32_t ssrc,
cricket::MediaType media_type) {
RtpSenderInternal* sender = FindSenderById(track_id);
if (!sender) {
LOG(LS_WARNING) << "An unknown RtpSender with id " << track_id
<< " has been configured in the local description.";
return;
}
if (sender->media_type() != media_type) {
LOG(LS_WARNING) << "An RtpSender has been configured in the local"
<< " description with an unexpected media type.";
return;
}
sender->set_stream_id(stream_label);
sender->SetSsrc(ssrc);
}
void PeerConnection::OnLocalTrackRemoved(const std::string& stream_label,
const std::string& track_id,
uint32_t ssrc,
cricket::MediaType media_type) {
RtpSenderInternal* sender = FindSenderById(track_id);
if (!sender) {
// This is the normal case. I.e., RemoveStream has been called and the
// SessionDescriptions has been renegotiated.
return;
}
// A sender has been removed from the SessionDescription but it's still
// associated with the PeerConnection. This only occurs if the SDP doesn't
// match with the calls to CreateSender, AddStream and RemoveStream.
if (sender->media_type() != media_type) {
LOG(LS_WARNING) << "An RtpSender has been configured in the local"
<< " description with an unexpected media type.";
return;
}
sender->SetSsrc(0);
}
void PeerConnection::UpdateLocalRtpDataChannels(
const cricket::StreamParamsVec& streams) {
std::vector<std::string> existing_channels;
// Find new and active data channels.
for (const cricket::StreamParams& params : streams) {
// |it->sync_label| is actually the data channel label. The reason is that
// we use the same naming of data channels as we do for
// MediaStreams and Tracks.
// For MediaStreams, the sync_label is the MediaStream label and the
// track label is the same as |streamid|.
const std::string& channel_label = params.sync_label;
auto data_channel_it = rtp_data_channels_.find(channel_label);
if (!VERIFY(data_channel_it != rtp_data_channels_.end())) {
continue;
}
// Set the SSRC the data channel should use for sending.
data_channel_it->second->SetSendSsrc(params.first_ssrc());
existing_channels.push_back(data_channel_it->first);
}
UpdateClosingRtpDataChannels(existing_channels, true);
}
void PeerConnection::UpdateRemoteRtpDataChannels(
const cricket::StreamParamsVec& streams) {
std::vector<std::string> existing_channels;
// Find new and active data channels.
for (const cricket::StreamParams& params : streams) {
// The data channel label is either the mslabel or the SSRC if the mslabel
// does not exist. Ex a=ssrc:444330170 mslabel:test1.
std::string label = params.sync_label.empty()
? rtc::ToString(params.first_ssrc())
: params.sync_label;
auto data_channel_it = rtp_data_channels_.find(label);
if (data_channel_it == rtp_data_channels_.end()) {
// This is a new data channel.
CreateRemoteRtpDataChannel(label, params.first_ssrc());
} else {
data_channel_it->second->SetReceiveSsrc(params.first_ssrc());
}
existing_channels.push_back(label);
}
UpdateClosingRtpDataChannels(existing_channels, false);
}
void PeerConnection::UpdateClosingRtpDataChannels(
const std::vector<std::string>& active_channels,
bool is_local_update) {
auto it = rtp_data_channels_.begin();
while (it != rtp_data_channels_.end()) {
DataChannel* data_channel = it->second;
if (std::find(active_channels.begin(), active_channels.end(),
data_channel->label()) != active_channels.end()) {
++it;
continue;
}
if (is_local_update) {
data_channel->SetSendSsrc(0);
} else {
data_channel->RemotePeerRequestClose();
}
if (data_channel->state() == DataChannel::kClosed) {
rtp_data_channels_.erase(it);
it = rtp_data_channels_.begin();
} else {
++it;
}
}
}
void PeerConnection::CreateRemoteRtpDataChannel(const std::string& label,
uint32_t remote_ssrc) {
rtc::scoped_refptr<DataChannel> channel(
InternalCreateDataChannel(label, nullptr));
if (!channel.get()) {
LOG(LS_WARNING) << "Remote peer requested a DataChannel but"
<< "CreateDataChannel failed.";
return;
}
channel->SetReceiveSsrc(remote_ssrc);
rtc::scoped_refptr<DataChannelInterface> proxy_channel =
DataChannelProxy::Create(signaling_thread(), channel);
// Call both the raw pointer and scoped_refptr versions of the method
// for compatibility.
observer_->OnDataChannel(proxy_channel.get());
observer_->OnDataChannel(std::move(proxy_channel));
}
rtc::scoped_refptr<DataChannel> PeerConnection::InternalCreateDataChannel(
const std::string& label,
const InternalDataChannelInit* config) {
if (IsClosed()) {
return nullptr;
}
if (session_->data_channel_type() == cricket::DCT_NONE) {
LOG(LS_ERROR)
<< "InternalCreateDataChannel: Data is not supported in this call.";
return nullptr;
}
InternalDataChannelInit new_config =
config ? (*config) : InternalDataChannelInit();
if (session_->data_channel_type() == cricket::DCT_SCTP) {
if (new_config.id < 0) {
rtc::SSLRole role;
if ((session_->GetSslRole(session_->data_channel(), &role)) &&
!sid_allocator_.AllocateSid(role, &new_config.id)) {
LOG(LS_ERROR) << "No id can be allocated for the SCTP data channel.";
return nullptr;
}
} else if (!sid_allocator_.ReserveSid(new_config.id)) {
LOG(LS_ERROR) << "Failed to create a SCTP data channel "
<< "because the id is already in use or out of range.";
return nullptr;
}
}
rtc::scoped_refptr<DataChannel> channel(DataChannel::Create(
session_.get(), session_->data_channel_type(), label, new_config));
if (!channel) {
sid_allocator_.ReleaseSid(new_config.id);
return nullptr;
}
if (channel->data_channel_type() == cricket::DCT_RTP) {
if (rtp_data_channels_.find(channel->label()) != rtp_data_channels_.end()) {
LOG(LS_ERROR) << "DataChannel with label " << channel->label()
<< " already exists.";
return nullptr;
}
rtp_data_channels_[channel->label()] = channel;
} else {
RTC_DCHECK(channel->data_channel_type() == cricket::DCT_SCTP);
sctp_data_channels_.push_back(channel);
channel->SignalClosed.connect(this,
&PeerConnection::OnSctpDataChannelClosed);
}
return channel;
}
bool PeerConnection::HasDataChannels() const {
#ifdef HAVE_QUIC
return !rtp_data_channels_.empty() || !sctp_data_channels_.empty() ||
(session_->quic_data_transport() &&
session_->quic_data_transport()->HasDataChannels());
#else
return !rtp_data_channels_.empty() || !sctp_data_channels_.empty();
#endif // HAVE_QUIC
}
void PeerConnection::AllocateSctpSids(rtc::SSLRole role) {
for (const auto& channel : sctp_data_channels_) {
if (channel->id() < 0) {
int sid;
if (!sid_allocator_.AllocateSid(role, &sid)) {
LOG(LS_ERROR) << "Failed to allocate SCTP sid.";
continue;
}
channel->SetSctpSid(sid);
}
}
}
void PeerConnection::OnSctpDataChannelClosed(DataChannel* channel) {
RTC_DCHECK(signaling_thread()->IsCurrent());
for (auto it = sctp_data_channels_.begin(); it != sctp_data_channels_.end();
++it) {
if (it->get() == channel) {
if (channel->id() >= 0) {
sid_allocator_.ReleaseSid(channel->id());
}
// Since this method is triggered by a signal from the DataChannel,
// we can't free it directly here; we need to free it asynchronously.
sctp_data_channels_to_free_.push_back(*it);
sctp_data_channels_.erase(it);
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_FREE_DATACHANNELS,
nullptr);
return;
}
}
}
void PeerConnection::OnVoiceChannelCreated() {
SetChannelOnSendersAndReceivers<AudioRtpSender, AudioRtpReceiver>(
session_->voice_channel(), senders_, receivers_,
cricket::MEDIA_TYPE_AUDIO);
}
void PeerConnection::OnVoiceChannelDestroyed() {
SetChannelOnSendersAndReceivers<AudioRtpSender, AudioRtpReceiver,
cricket::VoiceChannel>(
nullptr, senders_, receivers_, cricket::MEDIA_TYPE_AUDIO);
}
void PeerConnection::OnVideoChannelCreated() {
SetChannelOnSendersAndReceivers<VideoRtpSender, VideoRtpReceiver>(
session_->video_channel(), senders_, receivers_,
cricket::MEDIA_TYPE_VIDEO);
}
void PeerConnection::OnVideoChannelDestroyed() {
SetChannelOnSendersAndReceivers<VideoRtpSender, VideoRtpReceiver,
cricket::VideoChannel>(
nullptr, senders_, receivers_, cricket::MEDIA_TYPE_VIDEO);
}
void PeerConnection::OnDataChannelCreated() {
for (const auto& channel : sctp_data_channels_) {
channel->OnTransportChannelCreated();
}
}
void PeerConnection::OnDataChannelDestroyed() {
// Use a temporary copy of the RTP/SCTP DataChannel list because the
// DataChannel may callback to us and try to modify the list.
std::map<std::string, rtc::scoped_refptr<DataChannel>> temp_rtp_dcs;
temp_rtp_dcs.swap(rtp_data_channels_);
for (const auto& kv : temp_rtp_dcs) {
kv.second->OnTransportChannelDestroyed();
}
std::vector<rtc::scoped_refptr<DataChannel>> temp_sctp_dcs;
temp_sctp_dcs.swap(sctp_data_channels_);
for (const auto& channel : temp_sctp_dcs) {
channel->OnTransportChannelDestroyed();
}
}
void PeerConnection::OnDataChannelOpenMessage(
const std::string& label,
const InternalDataChannelInit& config) {
rtc::scoped_refptr<DataChannel> channel(
InternalCreateDataChannel(label, &config));
if (!channel.get()) {
LOG(LS_ERROR) << "Failed to create DataChannel from the OPEN message.";
return;
}
rtc::scoped_refptr<DataChannelInterface> proxy_channel =
DataChannelProxy::Create(signaling_thread(), channel);
// Call both the raw pointer and scoped_refptr versions of the method
// for compatibility.
observer_->OnDataChannel(proxy_channel.get());
observer_->OnDataChannel(std::move(proxy_channel));
}
RtpSenderInternal* PeerConnection::FindSenderById(const std::string& id) {
auto it = std::find_if(
senders_.begin(), senders_.end(),
[id](const rtc::scoped_refptr<
RtpSenderProxyWithInternal<RtpSenderInternal>>& sender) {
return sender->id() == id;
});
return it != senders_.end() ? (*it)->internal() : nullptr;
}
std::vector<
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>::iterator
PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) {
return std::find_if(
senders_.begin(), senders_.end(),
[track](const rtc::scoped_refptr<
RtpSenderProxyWithInternal<RtpSenderInternal>>& sender) {
return sender->track() == track;
});
}
std::vector<rtc::scoped_refptr<
RtpReceiverProxyWithInternal<RtpReceiverInternal>>>::iterator
PeerConnection::FindReceiverForTrack(const std::string& track_id) {
return std::find_if(
receivers_.begin(), receivers_.end(),
[track_id](const rtc::scoped_refptr<
RtpReceiverProxyWithInternal<RtpReceiverInternal>>& receiver) {
return receiver->id() == track_id;
});
}
PeerConnection::TrackInfos* PeerConnection::GetRemoteTracks(
cricket::MediaType media_type) {
RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
media_type == cricket::MEDIA_TYPE_VIDEO);
return (media_type == cricket::MEDIA_TYPE_AUDIO) ? &remote_audio_tracks_
: &remote_video_tracks_;
}
PeerConnection::TrackInfos* PeerConnection::GetLocalTracks(
cricket::MediaType media_type) {
RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
media_type == cricket::MEDIA_TYPE_VIDEO);
return (media_type == cricket::MEDIA_TYPE_AUDIO) ? &local_audio_tracks_
: &local_video_tracks_;
}
const PeerConnection::TrackInfo* PeerConnection::FindTrackInfo(
const PeerConnection::TrackInfos& infos,
const std::string& stream_label,
const std::string track_id) const {
for (const TrackInfo& track_info : infos) {
if (track_info.stream_label == stream_label &&
track_info.track_id == track_id) {
return &track_info;
}
}
return nullptr;
}
DataChannel* PeerConnection::FindDataChannelBySid(int sid) const {
for (const auto& channel : sctp_data_channels_) {
if (channel->id() == sid) {
return channel;
}
}
return nullptr;
}
bool PeerConnection::InitializePortAllocator_n(
const RTCConfiguration& configuration) {
cricket::ServerAddresses stun_servers;
std::vector<cricket::RelayServerConfig> turn_servers;
if (!ParseIceServers(configuration.servers, &stun_servers, &turn_servers)) {
return false;
}
port_allocator_->Initialize();
// To handle both internal and externally created port allocator, we will
// enable BUNDLE here.
int portallocator_flags = port_allocator_->flags();
portallocator_flags |= cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
cricket::PORTALLOCATOR_ENABLE_IPV6;
// If the disable-IPv6 flag was specified, we'll not override it
// by experiment.
if (configuration.disable_ipv6) {
portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
} else if (webrtc::field_trial::FindFullName("WebRTC-IPv6Default") ==
"Disabled") {
portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
}
if (configuration.tcp_candidate_policy == kTcpCandidatePolicyDisabled) {
portallocator_flags |= cricket::PORTALLOCATOR_DISABLE_TCP;
LOG(LS_INFO) << "TCP candidates are disabled.";
}
if (configuration.candidate_network_policy ==
kCandidateNetworkPolicyLowCost) {
portallocator_flags |= cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS;
LOG(LS_INFO) << "Do not gather candidates on high-cost networks";
}
port_allocator_->set_flags(portallocator_flags);
// No step delay is used while allocating ports.
port_allocator_->set_step_delay(cricket::kMinimumStepDelay);
port_allocator_->set_candidate_filter(
ConvertIceTransportTypeToCandidateFilter(configuration.type));
// Call this last since it may create pooled allocator sessions using the
// properties set above.
port_allocator_->SetConfiguration(stun_servers, turn_servers,
configuration.ice_candidate_pool_size,
configuration.prune_turn_ports);
return true;
}
bool PeerConnection::ReconfigurePortAllocator_n(
const RTCConfiguration& configuration) {
cricket::ServerAddresses stun_servers;
std::vector<cricket::RelayServerConfig> turn_servers;
if (!ParseIceServers(configuration.servers, &stun_servers, &turn_servers)) {
return false;
}
port_allocator_->set_candidate_filter(
ConvertIceTransportTypeToCandidateFilter(configuration.type));
// Call this last since it may create pooled allocator sessions using the
// candidate filter set above.
port_allocator_->SetConfiguration(stun_servers, turn_servers,
configuration.ice_candidate_pool_size,
configuration.prune_turn_ports);
return true;
}
bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file,
int64_t max_size_bytes) {
return media_controller_->call_w()->StartEventLog(file, max_size_bytes);
}
void PeerConnection::StopRtcEventLog_w() {
media_controller_->call_w()->StopEventLog();
}
} // namespace webrtc