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webrtc / src / webrtc / 816f59158d1c4f747a6cc1496b76e7789cdc58e0 / . / pc / peerconnection.cc
blob: 6112357ce1dd586adad63634300bd08a810dde52 [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/pc/peerconnection.h"
#include <algorithm>
#include <utility>
#include <vector>
#include "webrtc/api/jsepicecandidate.h"
#include "webrtc/api/jsepsessiondescription.h"
#include "webrtc/api/mediaconstraintsinterface.h"
#include "webrtc/api/mediastreamproxy.h"
#include "webrtc/api/mediastreamtrackproxy.h"
#include "webrtc/call/call.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/sctp/sctptransport.h"
#include "webrtc/pc/audiotrack.h"
#include "webrtc/pc/channelmanager.h"
#include "webrtc/pc/dtmfsender.h"
#include "webrtc/pc/mediastream.h"
#include "webrtc/pc/mediastreamobserver.h"
#include "webrtc/pc/remoteaudiosource.h"
#include "webrtc/pc/rtpreceiver.h"
#include "webrtc/pc/rtpsender.h"
#include "webrtc/pc/streamcollection.h"
#include "webrtc/pc/videocapturertracksource.h"
#include "webrtc/pc/videotrack.h"
#include "webrtc/rtc_base/bind.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/logging.h"
#include "webrtc/rtc_base/stringencode.h"
#include "webrtc/rtc_base/stringutils.h"
#include "webrtc/rtc_base/trace_event.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/field_trial.h"
namespace {
using webrtc::DataChannel;
using webrtc::MediaConstraintsInterface;
using webrtc::MediaStreamInterface;
using webrtc::PeerConnectionInterface;
using webrtc::RTCError;
using webrtc::RTCErrorType;
using webrtc::RtpSenderInternal;
using webrtc::RtpSenderInterface;
using webrtc::RtpSenderProxy;
using webrtc::RtpSenderProxyWithInternal;
using webrtc::StreamCollection;
static const char kDefaultStreamLabel[] = "default";
static const char kDefaultAudioTrackLabel[] = "defaulta0";
static const char kDefaultVideoTrackLabel[] = "defaultv0";
// The length of RTCP CNAMEs.
static const int kRtcpCnameLength = 16;
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;
};
// 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 options to |[audio/video]_media_description_options| from |senders|.
void AddRtpSenderOptions(
const std::vector<rtc::scoped_refptr<
RtpSenderProxyWithInternal<RtpSenderInternal>>>& senders,
cricket::MediaDescriptionOptions* audio_media_description_options,
cricket::MediaDescriptionOptions* video_media_description_options) {
for (const auto& sender : senders) {
if (sender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
if (audio_media_description_options) {
audio_media_description_options->AddAudioSender(
sender->id(), sender->internal()->stream_id());
}
} else {
RTC_DCHECK(sender->media_type() == cricket::MEDIA_TYPE_VIDEO);
if (video_media_description_options) {
video_media_description_options->AddVideoSender(
sender->id(), sender->internal()->stream_id(), 1);
}
}
}
}
// Add options to |session_options| from |rtp_data_channels|.
void AddRtpDataChannelOptions(
const std::map<std::string, rtc::scoped_refptr<DataChannel>>&
rtp_data_channels,
cricket::MediaDescriptionOptions* data_media_description_options) {
if (!data_media_description_options) {
return;
}
// 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) {
// Legacy RTP data channels are signaled with the track/stream ID set to
// the data channel's label.
data_media_description_options->AddRtpDataChannel(channel->label(),
channel->label());
}
}
}
uint32_t ConvertIceTransportTypeToCandidateFilter(
PeerConnectionInterface::IceTransportsType type) {
switch (type) {
case PeerConnectionInterface::kNone:
return cricket::CF_NONE;
case PeerConnectionInterface::kRelay:
return cricket::CF_RELAY;
case PeerConnectionInterface::kNoHost:
return (cricket::CF_ALL & ~cricket::CF_HOST);
case PeerConnectionInterface::kAll:
return cricket::CF_ALL;
default:
RTC_NOTREACHED();
}
return cricket::CF_NONE;
}
// Helper method to set a voice/video channel on all applicable senders
// and receivers when one is created/destroyed by WebRtcSession.
//
// Used by On(Voice|Video)Channel(Created|Destroyed)
template <class SENDER,
class RECEIVER,
class CHANNEL,
class SENDERS,
class RECEIVERS>
void SetChannelOnSendersAndReceivers(CHANNEL* channel,
SENDERS& senders,
RECEIVERS& receivers,
cricket::MediaType media_type) {
for (auto& sender : senders) {
if (sender->media_type() == media_type) {
static_cast<SENDER*>(sender->internal())->SetChannel(channel);
}
}
for (auto& receiver : receivers) {
if (receiver->media_type() == media_type) {
if (!channel) {
receiver->internal()->Stop();
}
static_cast<RECEIVER*>(receiver->internal())->SetChannel(channel);
}
}
}
// Helper to set an error and return from a method.
bool SafeSetError(webrtc::RTCErrorType type, webrtc::RTCError* error) {
if (error) {
error->set_type(type);
}
return type == webrtc::RTCErrorType::NONE;
}
bool SafeSetError(webrtc::RTCError error, webrtc::RTCError* error_out) {
if (error_out) {
*error_out = std::move(error);
}
return error.ok();
}
} // namespace
namespace webrtc {
bool PeerConnectionInterface::RTCConfiguration::operator==(
const PeerConnectionInterface::RTCConfiguration& o) const {
// This static_assert prevents us from accidentally breaking operator==.
// Note: Order matters! Fields must be ordered the same as RTCConfiguration.
struct stuff_being_tested_for_equality {
IceServers servers;
IceTransportsType type;
BundlePolicy bundle_policy;
RtcpMuxPolicy rtcp_mux_policy;
std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
int ice_candidate_pool_size;
bool disable_ipv6;
bool disable_ipv6_on_wifi;
int max_ipv6_networks;
bool enable_rtp_data_channel;
rtc::Optional<int> screencast_min_bitrate;
rtc::Optional<bool> combined_audio_video_bwe;
rtc::Optional<bool> enable_dtls_srtp;
TcpCandidatePolicy tcp_candidate_policy;
CandidateNetworkPolicy candidate_network_policy;
int audio_jitter_buffer_max_packets;
bool audio_jitter_buffer_fast_accelerate;
int ice_connection_receiving_timeout;
int ice_backup_candidate_pair_ping_interval;
ContinualGatheringPolicy continual_gathering_policy;
bool prioritize_most_likely_ice_candidate_pairs;
struct cricket::MediaConfig media_config;
bool enable_quic;
bool prune_turn_ports;
bool presume_writable_when_fully_relayed;
bool enable_ice_renomination;
bool redetermine_role_on_ice_restart;
rtc::Optional<int> ice_check_min_interval;
rtc::Optional<rtc::IntervalRange> ice_regather_interval_range;
};
static_assert(sizeof(stuff_being_tested_for_equality) == sizeof(*this),
"Did you add something to RTCConfiguration and forget to "
"update operator==?");
return type == o.type && servers == o.servers &&
bundle_policy == o.bundle_policy &&
rtcp_mux_policy == o.rtcp_mux_policy &&
tcp_candidate_policy == o.tcp_candidate_policy &&
candidate_network_policy == o.candidate_network_policy &&
audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
audio_jitter_buffer_fast_accelerate ==
o.audio_jitter_buffer_fast_accelerate &&
ice_connection_receiving_timeout ==
o.ice_connection_receiving_timeout &&
ice_backup_candidate_pair_ping_interval ==
o.ice_backup_candidate_pair_ping_interval &&
continual_gathering_policy == o.continual_gathering_policy &&
certificates == o.certificates &&
prioritize_most_likely_ice_candidate_pairs ==
o.prioritize_most_likely_ice_candidate_pairs &&
media_config == o.media_config && disable_ipv6 == o.disable_ipv6 &&
disable_ipv6_on_wifi == o.disable_ipv6_on_wifi &&
max_ipv6_networks == o.max_ipv6_networks &&
enable_rtp_data_channel == o.enable_rtp_data_channel &&
enable_quic == o.enable_quic &&
screencast_min_bitrate == o.screencast_min_bitrate &&
combined_audio_video_bwe == o.combined_audio_video_bwe &&
enable_dtls_srtp == o.enable_dtls_srtp &&
ice_candidate_pool_size == o.ice_candidate_pool_size &&
prune_turn_ports == o.prune_turn_ports &&
presume_writable_when_fully_relayed ==
o.presume_writable_when_fully_relayed &&
enable_ice_renomination == o.enable_ice_renomination &&
redetermine_role_on_ice_restart == o.redetermine_role_on_ice_restart &&
ice_check_min_interval == o.ice_check_min_interval &&
ice_regather_interval_range == o.ice_regather_interval_range;
}
bool PeerConnectionInterface::RTCConfiguration::operator!=(
const PeerConnectionInterface::RTCConfiguration& o) const {
return !(*this == o);
}
// Generate a RTCP CNAME when a PeerConnection is created.
std::string GenerateRtcpCname() {
std::string cname;
if (!rtc::CreateRandomString(kRtcpCnameLength, &cname)) {
LOG(LS_ERROR) << "Failed to generate CNAME.";
RTC_NOTREACHED();
}
return cname;
}
bool ValidateOfferAnswerOptions(
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options) {
return IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_audio) &&
IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_video);
}
// From |rtc_options|, fill parts of |session_options| shared by all generated
// m= sections (in other words, nothing that involves a map/array).
void ExtractSharedMediaSessionOptions(
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
cricket::MediaSessionOptions* session_options) {
session_options->vad_enabled = rtc_options.voice_activity_detection;
session_options->bundle_enabled = rtc_options.use_rtp_mux;
}
bool ConvertConstraintsToOfferAnswerOptions(
const MediaConstraintsInterface* constraints,
PeerConnectionInterface::RTCOfferAnswerOptions* offer_answer_options) {
if (!constraints) {
return true;
}
bool value = false;
size_t mandatory_constraints_satisfied = 0;
if (FindConstraint(constraints,
MediaConstraintsInterface::kOfferToReceiveAudio, &value,
&mandatory_constraints_satisfied)) {
offer_answer_options->offer_to_receive_audio =
value ? PeerConnectionInterface::RTCOfferAnswerOptions::
kOfferToReceiveMediaTrue
: 0;
}
if (FindConstraint(constraints,
MediaConstraintsInterface::kOfferToReceiveVideo, &value,
&mandatory_constraints_satisfied)) {
offer_answer_options->offer_to_receive_video =
value ? PeerConnectionInterface::RTCOfferAnswerOptions::
kOfferToReceiveMediaTrue
: 0;
}
if (FindConstraint(constraints,
MediaConstraintsInterface::kVoiceActivityDetection, &value,
&mandatory_constraints_satisfied)) {
offer_answer_options->voice_activity_detection = value;
}
if (FindConstraint(constraints, MediaConstraintsInterface::kUseRtpMux, &value,
&mandatory_constraints_satisfied)) {
offer_answer_options->use_rtp_mux = value;
}
if (FindConstraint(constraints, MediaConstraintsInterface::kIceRestart,
&value, &mandatory_constraints_satisfied)) {
offer_answer_options->ice_restart = value;
}
return mandatory_constraints_satisfied == constraints->GetMandatory().size();
}
PeerConnection::PeerConnection(PeerConnectionFactory* factory,
std::unique_ptr<RtcEventLog> event_log,
std::unique_ptr<Call> call)
: factory_(factory),
observer_(NULL),
uma_observer_(NULL),
event_log_(std::move(event_log)),
signaling_state_(kStable),
ice_connection_state_(kIceConnectionNew),
ice_gathering_state_(kIceGatheringNew),
rtcp_cname_(GenerateRtcpCname()),
local_streams_(StreamCollection::Create()),
remote_streams_(StreamCollection::Create()),
call_(std::move(call)) {}
PeerConnection::~PeerConnection() {
TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
RTC_DCHECK(signaling_thread()->IsCurrent());
// Need to detach RTP senders/receivers from WebRtcSession,
// since it's about to be destroyed.
for (const auto& sender : senders_) {
sender->internal()->Stop();
}
for (const auto& receiver : receivers_) {
receiver->internal()->Stop();
}
// Destroy stats_ because it depends on session_.
stats_.reset(nullptr);
if (stats_collector_) {
stats_collector_->WaitForPendingRequest();
stats_collector_ = nullptr;
}
// Now destroy session_ before destroying other members,
// because its destruction fires signals (such as VoiceChannelDestroyed)
// which will trigger some final actions in PeerConnection...
session_.reset(nullptr);
// port_allocator_ lives on the network thread and should be destroyed there.
network_thread()->Invoke<void>(RTC_FROM_HERE,
[this] { port_allocator_.reset(); });
// call_ must be destroyed on the worker thread.
factory_->worker_thread()->Invoke<void>(RTC_FROM_HERE,
[this] { call_.reset(); });
}
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");
RTCError config_error = ValidateConfiguration(configuration);
if (!config_error.ok()) {
LOG(LS_ERROR) << "Invalid configuration: " << config_error.message();
return false;
}
if (!allocator) {
LOG(LS_ERROR) << "PeerConnection initialized without a PortAllocator? "
<< "This shouldn't happen if using PeerConnectionFactory.";
return false;
}
if (!observer) {
// TODO(deadbeef): Why do we do this?
LOG(LS_ERROR) << "PeerConnection initialized without a "
<< "PeerConnectionObserver";
return false;
}
observer_ = observer;
port_allocator_ = std::move(allocator);
// The port allocator lives on the network thread and should be initialized
// there.
if (!network_thread()->Invoke<bool>(
RTC_FROM_HERE, rtc::Bind(&PeerConnection::InitializePortAllocator_n,
this, configuration))) {
return false;
}
session_.reset(new WebRtcSession(
call_.get(), factory_->channel_manager(), configuration.media_config,
event_log_.get(),
factory_->network_thread(),
factory_->worker_thread(), factory_->signaling_thread(),
port_allocator_.get(),
std::unique_ptr<cricket::TransportController>(
factory_->CreateTransportController(
port_allocator_.get(),
configuration.redetermine_role_on_ice_restart)),
#ifdef HAVE_SCTP
std::unique_ptr<cricket::SctpTransportInternalFactory>(
new cricket::SctpTransportFactory(factory_->network_thread()))
#else
nullptr
#endif
));
stats_.reset(new StatsCollector(this));
stats_collector_ = RTCStatsCollector::Create(this);
// Initialize the WebRtcSession. It creates transport channels etc.
if (!session_->Initialize(factory_->options(), std::move(cert_generator),
configuration)) {
return false;
}
// Register PeerConnection as receiver of local ice candidates.
// All the callbacks will be posted to the application from PeerConnection.
session_->RegisterIceObserver(this);
session_->SignalState.connect(this, &PeerConnection::OnSessionStateChange);
session_->SignalVoiceChannelCreated.connect(
this, &PeerConnection::OnVoiceChannelCreated);
session_->SignalVoiceChannelDestroyed.connect(
this, &PeerConnection::OnVoiceChannelDestroyed);
session_->SignalVideoChannelCreated.connect(
this, &PeerConnection::OnVideoChannelCreated);
session_->SignalVideoChannelDestroyed.connect(
this, &PeerConnection::OnVideoChannelDestroyed);
session_->SignalDataChannelCreated.connect(
this, &PeerConnection::OnDataChannelCreated);
session_->SignalDataChannelDestroyed.connect(
this, &PeerConnection::OnDataChannelDestroyed);
session_->SignalDataChannelOpenMessage.connect(
this, &PeerConnection::OnDataChannelOpenMessage);
configuration_ = configuration;
return true;
}
RTCError PeerConnection::ValidateConfiguration(
const RTCConfiguration& config) const {
if (config.ice_regather_interval_range &&
config.continual_gathering_policy == GATHER_ONCE) {
return RTCError(RTCErrorType::INVALID_PARAMETER,
"ice_regather_interval_range specified but continual "
"gathering policy is GATHER_ONCE");
}
return RTCError::OK();
}
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()) {
AddAudioTrack(track.get(), local_stream);
}
for (const auto& track : local_stream->GetVideoTracks()) {
AddVideoTrack(track.get(), local_stream);
}
stats_->AddStream(local_stream);
observer_->OnRenegotiationNeeded();
return true;
}
void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
TRACE_EVENT0("webrtc", "PeerConnection::RemoveStream");
if (!IsClosed()) {
for (const auto& track : local_stream->GetAudioTracks()) {
RemoveAudioTrack(track.get(), local_stream);
}
for (const auto& track : local_stream->GetVideoTracks()) {
RemoveVideoTrack(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 nullptr;
}
auto it = FindSenderForTrack(track);
if (it == senders_.end()) {
LOG(LS_ERROR) << "CreateDtmfSender called with a non-added track.";
return nullptr;
}
return (*it)->GetDtmfSender();
}
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 (!observer) {
LOG(LS_ERROR) << "GetStats - observer is NULL.";
return false;
}
stats_->UpdateStats(level);
// The StatsCollector is used to tell if a track is valid because it may
// remember tracks that the PeerConnection previously removed.
if (track && !stats_->IsValidTrack(track->id())) {
LOG(LS_WARNING) << "GetStats is called with an invalid track: "
<< track->id();
return false;
}
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_GETSTATS,
new GetStatsMsg(observer, track));
return true;
}
void PeerConnection::GetStats(RTCStatsCollectorCallback* callback) {
RTC_DCHECK(stats_collector_);
stats_collector_->GetStatsReport(callback);
}
PeerConnectionInterface::SignalingState PeerConnection::signaling_state() {
return signaling_state_;
}
PeerConnectionInterface::IceConnectionState
PeerConnection::ice_connection_state() {
return ice_connection_state_;
}
PeerConnectionInterface::IceGatheringState
PeerConnection::ice_gathering_state() {
return ice_gathering_state_;
}
rtc::scoped_refptr<DataChannelInterface>
PeerConnection::CreateDataChannel(
const std::string& label,
const DataChannelInit* config) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateDataChannel");
#ifdef HAVE_QUIC
if (session_->data_channel_type() == cricket::DCT_QUIC) {
// TODO(zhihuang): Handle case when config is NULL.
if (!config) {
LOG(LS_ERROR) << "Missing config for QUIC data channel.";
return nullptr;
}
// TODO(zhihuang): Allow unreliable or ordered QUIC data channels.
if (!config->reliable || config->ordered) {
LOG(LS_ERROR) << "QUIC data channel does not implement unreliable or "
"ordered delivery.";
return nullptr;
}
return session_->quic_data_transport()->CreateDataChannel(label, config);
}
#endif // HAVE_QUIC
bool first_datachannel = !HasDataChannels();
std::unique_ptr<InternalDataChannelInit> internal_config;
if (config) {
internal_config.reset(new InternalDataChannelInit(*config));
}
rtc::scoped_refptr<DataChannelInterface> channel(
InternalCreateDataChannel(label, internal_config.get()));
if (!channel.get()) {
return nullptr;
}
// Trigger the onRenegotiationNeeded event for every new RTP DataChannel, or
// the first SCTP DataChannel.
if (session_->data_channel_type() == cricket::DCT_RTP || first_datachannel) {
observer_->OnRenegotiationNeeded();
}
return DataChannelProxy::Create(signaling_thread(), channel.get());
}
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
if (!observer) {
LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
return;
}
PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options;
// Always create an offer even if |ConvertConstraintsToOfferAnswerOptions|
// returns false for now. Because |ConvertConstraintsToOfferAnswerOptions|
// compares the mandatory fields parsed with the mandatory fields added in the
// |constraints| and some downstream applications might create offers with
// mandatory fields which would not be parsed in the helper method. For
// example, in Chromium/remoting, |kEnableDtlsSrtp| is added to the
// |constraints| as a mandatory field but it is not parsed.
ConvertConstraintsToOfferAnswerOptions(constraints, &offer_answer_options);
CreateOffer(observer, offer_answer_options);
}
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
if (!observer) {
LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
return;
}
if (!ValidateOfferAnswerOptions(options)) {
std::string error = "CreateOffer called with invalid options.";
LOG(LS_ERROR) << error;
PostCreateSessionDescriptionFailure(observer, error);
return;
}
cricket::MediaSessionOptions session_options;
GetOptionsForOffer(options, &session_options);
session_->CreateOffer(observer, options, session_options);
}
void PeerConnection::CreateAnswer(
CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateAnswer");
if (!observer) {
LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
return;
}
if (!session_->remote_description() ||
session_->remote_description()->type() !=
SessionDescriptionInterface::kOffer) {
std::string error = "CreateAnswer called without remote offer.";
LOG(LS_ERROR) << error;
PostCreateSessionDescriptionFailure(observer, error);
return;
}
PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options;
if (!ConvertConstraintsToOfferAnswerOptions(constraints,
&offer_answer_options)) {
std::string error = "CreateAnswer called with invalid constraints.";
LOG(LS_ERROR) << error;
PostCreateSessionDescriptionFailure(observer, error);
return;
}
cricket::MediaSessionOptions session_options;
GetOptionsForAnswer(offer_answer_options, &session_options);
session_->CreateAnswer(observer, session_options);
}
void PeerConnection::CreateAnswer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {
TRACE_EVENT0("webrtc", "PeerConnection::CreateAnswer");
if (!observer) {
LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
return;
}
cricket::MediaSessionOptions session_options;
GetOptionsForAnswer(options, &session_options);
session_->CreateAnswer(observer, session_options);
}
void PeerConnection::SetLocalDescription(
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) {
TRACE_EVENT0("webrtc", "PeerConnection::SetLocalDescription");
if (IsClosed()) {
return;
}
if (!observer) {
LOG(LS_ERROR) << "SetLocalDescription - observer is NULL.";
return;
}
if (!desc) {
PostSetSessionDescriptionFailure(observer, "SessionDescription is NULL.");
return;
}
// Update stats here so that we have the most recent stats for tracks and
// streams that might be removed by updating the session description.
stats_->UpdateStats(kStatsOutputLevelStandard);
std::string error;
if (!session_->SetLocalDescription(desc, &error)) {
PostSetSessionDescriptionFailure(observer, error);
return;
}
// If setting the description decided our SSL role, allocate any necessary
// SCTP sids.
rtc::SSLRole role;
if (session_->data_channel_type() == cricket::DCT_SCTP &&
session_->GetSctpSslRole(&role)) {
AllocateSctpSids(role);
}
// Update state and SSRC of local MediaStreams and DataChannels based on the
// local session description.
const cricket::ContentInfo* audio_content =
GetFirstAudioContent(desc->description());
if (audio_content) {
if (audio_content->rejected) {
RemoveTracks(cricket::MEDIA_TYPE_AUDIO);
} else {
const cricket::AudioContentDescription* audio_desc =
static_cast<const cricket::AudioContentDescription*>(
audio_content->description);
UpdateLocalTracks(audio_desc->streams(), audio_desc->type());
}
}
const cricket::ContentInfo* video_content =
GetFirstVideoContent(desc->description());
if (video_content) {
if (video_content->rejected) {
RemoveTracks(cricket::MEDIA_TYPE_VIDEO);
} else {
const cricket::VideoContentDescription* video_desc =
static_cast<const cricket::VideoContentDescription*>(
video_content->description);
UpdateLocalTracks(video_desc->streams(), video_desc->type());
}
}
const cricket::ContentInfo* data_content =
GetFirstDataContent(desc->description());
if (data_content) {
const cricket::DataContentDescription* data_desc =
static_cast<const cricket::DataContentDescription*>(
data_content->description);
if (rtc::starts_with(data_desc->protocol().data(),
cricket::kMediaProtocolRtpPrefix)) {
UpdateLocalRtpDataChannels(data_desc->streams());
}
}
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
signaling_thread()->Post(RTC_FROM_HERE, this,
MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
// According to JSEP, after setLocalDescription, changing the candidate pool
// size is not allowed, and changing the set of ICE servers will not result
// in new candidates being gathered.
port_allocator_->FreezeCandidatePool();
// 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();
if (desc->type() == SessionDescriptionInterface::kAnswer) {
// TODO(deadbeef): We already had to hop to the network thread for
// MaybeStartGathering...
network_thread()->Invoke<void>(
RTC_FROM_HERE,
rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
port_allocator_.get()));
}
}
void PeerConnection::SetRemoteDescription(
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) {
TRACE_EVENT0("webrtc", "PeerConnection::SetRemoteDescription");
if (IsClosed()) {
return;
}
if (!observer) {
LOG(LS_ERROR) << "SetRemoteDescription - observer is NULL.";
return;
}
if (!desc) {
PostSetSessionDescriptionFailure(observer, "SessionDescription is NULL.");
return;
}
// Update stats here so that we have the most recent stats for tracks and
// streams that might be removed by updating the session description.
stats_->UpdateStats(kStatsOutputLevelStandard);
std::string error;
if (!session_->SetRemoteDescription(desc, &error)) {
PostSetSessionDescriptionFailure(observer, error);
return;
}
// If setting the description decided our SSL role, allocate any necessary
// SCTP sids.
rtc::SSLRole role;
if (session_->data_channel_type() == cricket::DCT_SCTP &&
session_->GetSctpSslRole(&role)) {
AllocateSctpSids(role);
}
const cricket::SessionDescription* remote_desc = desc->description();
const cricket::ContentInfo* audio_content = GetFirstAudioContent(remote_desc);
const cricket::ContentInfo* video_content = GetFirstVideoContent(remote_desc);
const cricket::AudioContentDescription* audio_desc =
GetFirstAudioContentDescription(remote_desc);
const cricket::VideoContentDescription* video_desc =
GetFirstVideoContentDescription(remote_desc);
const cricket::DataContentDescription* data_desc =
GetFirstDataContentDescription(remote_desc);
// Check if the descriptions include streams, just in case the peer supports
// MSID, but doesn't indicate so with "a=msid-semantic".
if (remote_desc->msid_supported() ||
(audio_desc && !audio_desc->streams().empty()) ||
(video_desc && !video_desc->streams().empty())) {
remote_peer_supports_msid_ = true;
}
// We wait to signal new streams until we finish processing the description,
// since only at that point will new streams have all their tracks.
rtc::scoped_refptr<StreamCollection> new_streams(StreamCollection::Create());
// Find all audio rtp streams and create corresponding remote AudioTracks
// and MediaStreams.
if (audio_content) {
if (audio_content->rejected) {
RemoveTracks(cricket::MEDIA_TYPE_AUDIO);
} else {
bool default_audio_track_needed =
!remote_peer_supports_msid_ &&
MediaContentDirectionHasSend(audio_desc->direction());
UpdateRemoteStreamsList(GetActiveStreams(audio_desc),
default_audio_track_needed, audio_desc->type(),
new_streams);
}
}
// Find all video rtp streams and create corresponding remote VideoTracks
// and MediaStreams.
if (video_content) {
if (video_content->rejected) {
RemoveTracks(cricket::MEDIA_TYPE_VIDEO);
} else {
bool default_video_track_needed =
!remote_peer_supports_msid_ &&
MediaContentDirectionHasSend(video_desc->direction());
UpdateRemoteStreamsList(GetActiveStreams(video_desc),
default_video_track_needed, video_desc->type(),
new_streams);
}
}
// Update the DataChannels with the information from the remote peer.
if (data_desc) {
if (rtc::starts_with(data_desc->protocol().data(),
cricket::kMediaProtocolRtpPrefix)) {
UpdateRemoteRtpDataChannels(GetActiveStreams(data_desc));
}
}
// Iterate new_streams and notify the observer about new MediaStreams.
for (size_t i = 0; i < new_streams->count(); ++i) {
MediaStreamInterface* new_stream = new_streams->at(i);
stats_->AddStream(new_stream);
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);
if (desc->type() == SessionDescriptionInterface::kAnswer) {
// TODO(deadbeef): We already had to hop to the network thread for
// MaybeStartGathering...
network_thread()->Invoke<void>(
RTC_FROM_HERE,
rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
port_allocator_.get()));
}
}
PeerConnectionInterface::RTCConfiguration PeerConnection::GetConfiguration() {
return configuration_;
}
bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration,
RTCError* error) {
TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
if (session_->local_description() &&
configuration.ice_candidate_pool_size !=
configuration_.ice_candidate_pool_size) {
LOG(LS_ERROR) << "Can't change candidate pool size after calling "
"SetLocalDescription.";
return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
}
// The simplest (and most future-compatible) way to tell if the config was
// modified in an invalid way is to copy each property we do support
// modifying, then use operator==. There are far more properties we don't
// support modifying than those we do, and more could be added.
RTCConfiguration modified_config = configuration_;
modified_config.servers = configuration.servers;
modified_config.type = configuration.type;
modified_config.ice_candidate_pool_size =
configuration.ice_candidate_pool_size;
modified_config.prune_turn_ports = configuration.prune_turn_ports;
modified_config.ice_check_min_interval = configuration.ice_check_min_interval;
if (configuration != modified_config) {
LOG(LS_ERROR) << "Modifying the configuration in an unsupported way.";
return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
}
// Validate the modified configuration.
RTCError validate_error = ValidateConfiguration(modified_config);
if (!validate_error.ok()) {
return SafeSetError(std::move(validate_error), error);
}
// Note that this isn't possible through chromium, since it's an unsigned
// short in WebIDL.
if (configuration.ice_candidate_pool_size < 0 ||
configuration.ice_candidate_pool_size > UINT16_MAX) {
return SafeSetError(RTCErrorType::INVALID_RANGE, error);
}
// Parse ICE servers before hopping to network thread.
cricket::ServerAddresses stun_servers;
std::vector<cricket::RelayServerConfig> turn_servers;
RTCErrorType parse_error =
ParseIceServers(configuration.servers, &stun_servers, &turn_servers);
if (parse_error != RTCErrorType::NONE) {
return SafeSetError(parse_error, error);
}
// In theory this shouldn't fail.
if (!network_thread()->Invoke<bool>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnection::ReconfigurePortAllocator_n, this,
stun_servers, turn_servers, modified_config.type,
modified_config.ice_candidate_pool_size,
modified_config.prune_turn_ports))) {
LOG(LS_ERROR) << "Failed to apply configuration to PortAllocator.";
return SafeSetError(RTCErrorType::INTERNAL_ERROR, error);
}
// As described in JSEP, calling setConfiguration with new ICE servers or
// candidate policy must set a "needs-ice-restart" bit so that the next offer
// triggers an ICE restart which will pick up the changes.
if (modified_config.servers != configuration_.servers ||
modified_config.type != configuration_.type ||
modified_config.prune_turn_ports != configuration_.prune_turn_ports) {
session_->SetNeedsIceRestartFlag();
}
if (modified_config.ice_check_min_interval !=
configuration_.ice_check_min_interval) {
session_->SetIceConfig(session_->ParseIceConfig(modified_config));
}
configuration_ = modified_config;
return SafeSetError(RTCErrorType::NONE, error);
}
bool PeerConnection::AddIceCandidate(
const IceCandidateInterface* ice_candidate) {
TRACE_EVENT0("webrtc", "PeerConnection::AddIceCandidate");
if (IsClosed()) {
return false;
}
return session_->ProcessIceMessage(ice_candidate);
}
bool PeerConnection::RemoveIceCandidates(
const std::vector<cricket::Candidate>& candidates) {
TRACE_EVENT0("webrtc", "PeerConnection::RemoveIceCandidates");
return session_->RemoveRemoteIceCandidates(candidates);
}
void PeerConnection::RegisterUMAObserver(UMAObserver* observer) {
TRACE_EVENT0("webrtc", "PeerConnection::RegisterUmaObserver");
uma_observer_ = observer;
if (session_) {
session_->set_metrics_observer(uma_observer_);
}
// Send information about IPv4/IPv6 status.
if (uma_observer_) {
port_allocator_->SetMetricsObserver(uma_observer_);
if (port_allocator_->flags() & cricket::PORTALLOCATOR_ENABLE_IPV6) {
uma_observer_->IncrementEnumCounter(
kEnumCounterAddressFamily, kPeerConnection_IPv6,
kPeerConnectionAddressFamilyCounter_Max);
} else {
uma_observer_->IncrementEnumCounter(
kEnumCounterAddressFamily, kPeerConnection_IPv4,
kPeerConnectionAddressFamilyCounter_Max);
}
}
}
RTCError PeerConnection::SetBitrate(const BitrateParameters& bitrate) {
rtc::Thread* worker_thread = factory_->worker_thread();
if (!worker_thread->IsCurrent()) {
return worker_thread->Invoke<RTCError>(
RTC_FROM_HERE, rtc::Bind(&PeerConnection::SetBitrate, this, bitrate));
}
const bool has_min = static_cast<bool>(bitrate.min_bitrate_bps);
const bool has_current = static_cast<bool>(bitrate.current_bitrate_bps);
const bool has_max = static_cast<bool>(bitrate.max_bitrate_bps);
if (has_min && *bitrate.min_bitrate_bps < 0) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"min_bitrate_bps <= 0");
}
if (has_current) {
if (has_min && *bitrate.current_bitrate_bps < *bitrate.min_bitrate_bps) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"current_bitrate_bps < min_bitrate_bps");
} else if (*bitrate.current_bitrate_bps < 0) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"curent_bitrate_bps < 0");
}
}
if (has_max) {
if (has_current &&
*bitrate.max_bitrate_bps < *bitrate.current_bitrate_bps) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"max_bitrate_bps < current_bitrate_bps");
} else if (has_min && *bitrate.max_bitrate_bps < *bitrate.min_bitrate_bps) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"max_bitrate_bps < min_bitrate_bps");
} else if (*bitrate.max_bitrate_bps < 0) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"max_bitrate_bps < 0");
}
}
Call::Config::BitrateConfigMask mask;
mask.min_bitrate_bps = bitrate.min_bitrate_bps;
mask.start_bitrate_bps = bitrate.current_bitrate_bps;
mask.max_bitrate_bps = bitrate.max_bitrate_bps;
RTC_DCHECK(call_.get());
call_->SetBitrateConfigMask(mask);
return RTCError::OK();
}
bool PeerConnection::StartRtcEventLog(rtc::PlatformFile file,
int64_t max_size_bytes) {
return factory_->worker_thread()->Invoke<bool>(
RTC_FROM_HERE, rtc::Bind(&PeerConnection::StartRtcEventLog_w, this, file,
max_size_bytes));
}
void PeerConnection::StopRtcEventLog() {
factory_->worker_thread()->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&PeerConnection::StopRtcEventLog_w, this));
}
const SessionDescriptionInterface* PeerConnection::local_description() const {
return session_->local_description();
}
const SessionDescriptionInterface* PeerConnection::remote_description() const {
return session_->remote_description();
}
const SessionDescriptionInterface* PeerConnection::current_local_description()
const {
return session_->current_local_description();
}
const SessionDescriptionInterface* PeerConnection::current_remote_description()
const {
return session_->current_remote_description();
}
const SessionDescriptionInterface* PeerConnection::pending_local_description()
const {
return session_->pending_local_description();
}
const SessionDescriptionInterface* PeerConnection::pending_remote_description()
const {
return session_->pending_remote_description();
}
void PeerConnection::Close() {
TRACE_EVENT0("webrtc", "PeerConnection::Close");
// Update stats here so that we have the most recent stats for tracks and
// streams before the channels are closed.
stats_->UpdateStats(kStatsOutputLevelStandard);
session_->Close();
network_thread()->Invoke<void>(
RTC_FROM_HERE,
rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
port_allocator_.get()));
factory_->worker_thread()->Invoke<void>(RTC_FROM_HERE,
[this] { call_.reset(); });
// The event log must outlive call (and any other object that uses it).
event_log_.reset();
}
void PeerConnection::OnSessionStateChange(WebRtcSession* /*session*/,
WebRtcSession::State state) {
switch (state) {
case WebRtcSession::STATE_INIT:
ChangeSignalingState(PeerConnectionInterface::kStable);
break;
case WebRtcSession::STATE_SENTOFFER:
ChangeSignalingState(PeerConnectionInterface::kHaveLocalOffer);
break;
case WebRtcSession::STATE_SENTPRANSWER:
ChangeSignalingState(PeerConnectionInterface::kHaveLocalPrAnswer);
break;
case WebRtcSession::STATE_RECEIVEDOFFER:
ChangeSignalingState(PeerConnectionInterface::kHaveRemoteOffer);
break;
case WebRtcSession::STATE_RECEIVEDPRANSWER:
ChangeSignalingState(PeerConnectionInterface::kHaveRemotePrAnswer);
break;
case WebRtcSession::STATE_INPROGRESS:
ChangeSignalingState(PeerConnectionInterface::kStable);
break;
case WebRtcSession::STATE_CLOSED:
ChangeSignalingState(PeerConnectionInterface::kClosed);
break;
default:
break;
}
}
void PeerConnection::OnMessage(rtc::Message* msg) {
switch (msg->message_id) {
case MSG_SET_SESSIONDESCRIPTION_SUCCESS: {
SetSessionDescriptionMsg* param =
static_cast<SetSessionDescriptionMsg*>(msg->pdata);
param->observer->OnSuccess();
delete param;
break;
}
case MSG_SET_SESSIONDESCRIPTION_FAILED: {
SetSessionDescriptionMsg* param =
static_cast<SetSessionDescriptionMsg*>(msg->pdata);
param->observer->OnFailure(param->error);
delete param;
break;
}
case MSG_CREATE_SESSIONDESCRIPTION_FAILED: {
CreateSessionDescriptionMsg* param =
static_cast<CreateSessionDescriptionMsg*>(msg->pdata);
param->observer->OnFailure(param->error);
delete param;
break;
}
case MSG_GETSTATS: {
GetStatsMsg* param = static_cast<GetStatsMsg*>(msg->pdata);
StatsReports reports;
stats_->GetStats(param->track, &reports);
param->observer->OnComplete(reports);
delete param;
break;
}
case MSG_FREE_DATACHANNELS: {
sctp_data_channels_to_free_.clear();
break;
}
default:
RTC_NOTREACHED() << "Not implemented";
break;
}
}
void PeerConnection::CreateAudioReceiver(MediaStreamInterface* stream,
const std::string& track_id,
uint32_t ssrc) {
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(),
new AudioRtpReceiver(track_id, ssrc, session_->voice_channel()));
stream->AddTrack(
static_cast<AudioTrackInterface*>(receiver->internal()->track().get()));
receivers_.push_back(receiver);
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams;
streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
observer_->OnAddTrack(receiver, streams);
}
void PeerConnection::CreateVideoReceiver(MediaStreamInterface* stream,
const std::string& track_id,
uint32_t ssrc) {
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(),
new VideoRtpReceiver(track_id, factory_->worker_thread(), ssrc,
session_->video_channel()));
stream->AddTrack(
static_cast<VideoTrackInterface*>(receiver->internal()->track().get()));
receivers_.push_back(receiver);
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams;
streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
observer_->OnAddTrack(receiver, streams);
}
// TODO(deadbeef): Keep RtpReceivers around even if track goes away in remote
// description.
void PeerConnection::DestroyReceiver(const std::string& track_id) {
auto it = FindReceiverForTrack(track_id);
if (it == receivers_.end()) {
LOG(LS_WARNING) << "RtpReceiver for track with id " << track_id
<< " doesn't exist.";
} else {
(*it)->internal()->Stop();
receivers_.erase(it);
}
}
void PeerConnection::AddAudioTrack(AudioTrackInterface* track,
MediaStreamInterface* stream) {
RTC_DCHECK(!IsClosed());
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::RemoveAudioTrack(AudioTrackInterface* track,
MediaStreamInterface* stream) {
RTC_DCHECK(!IsClosed());
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::AddVideoTrack(VideoTrackInterface* track,
MediaStreamInterface* stream) {
RTC_DCHECK(!IsClosed());
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::RemoveVideoTrack(VideoTrackInterface* track,
MediaStreamInterface* stream) {
RTC_DCHECK(!IsClosed());
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::OnIceConnectionStateChange(
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(
std::unique_ptr<IceCandidateInterface> candidate) {
RTC_DCHECK(signaling_thread()->IsCurrent());
if (IsClosed()) {
return;
}
observer_->OnIceCandidate(candidate.get());
}
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;
}
AddAudioTrack(track, stream);
observer_->OnRenegotiationNeeded();
}
void PeerConnection::OnAudioTrackRemoved(AudioTrackInterface* track,
MediaStreamInterface* stream) {
if (IsClosed()) {
return;
}
RemoveAudioTrack(track, stream);
observer_->OnRenegotiationNeeded();
}
void PeerConnection::OnVideoTrackAdded(VideoTrackInterface* track,
MediaStreamInterface* stream) {
if (IsClosed()) {
return;
}
AddVideoTrack(track, stream);
observer_->OnRenegotiationNeeded();
}
void PeerConnection::OnVideoTrackRemoved(VideoTrackInterface* track,
MediaStreamInterface* stream) {
if (IsClosed()) {
return;
}
RemoveVideoTrack(track, stream);
observer_->OnRenegotiationNeeded();
}
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);
}
void PeerConnection::GetOptionsForOffer(
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
cricket::MediaSessionOptions* session_options) {
ExtractSharedMediaSessionOptions(rtc_options, session_options);
// Figure out transceiver directional preferences.
bool send_audio = HasRtpSender(cricket::MEDIA_TYPE_AUDIO);
bool send_video = HasRtpSender(cricket::MEDIA_TYPE_VIDEO);
// By default, generate sendrecv/recvonly m= sections.
bool recv_audio = true;
bool recv_video = true;
// By default, only offer a new m= section if we have media to send with it.
bool offer_new_audio_description = send_audio;
bool offer_new_video_description = send_video;
bool offer_new_data_description = HasDataChannels();
// The "offer_to_receive_X" options allow those defaults to be overridden.
if (rtc_options.offer_to_receive_audio != RTCOfferAnswerOptions::kUndefined) {
recv_audio = (rtc_options.offer_to_receive_audio > 0);
offer_new_audio_description =
offer_new_audio_description || (rtc_options.offer_to_receive_audio > 0);
}
if (rtc_options.offer_to_receive_video != RTCOfferAnswerOptions::kUndefined) {
recv_video = (rtc_options.offer_to_receive_video > 0);
offer_new_video_description =
offer_new_video_description || (rtc_options.offer_to_receive_video > 0);
}
rtc::Optional<size_t> audio_index;
rtc::Optional<size_t> video_index;
rtc::Optional<size_t> data_index;
// If a current description exists, generate m= sections in the same order,
// using the first audio/video/data section that appears and rejecting
// extraneous ones.
if (session_->local_description()) {
GenerateMediaDescriptionOptions(
session_->local_description(),
cricket::RtpTransceiverDirection(send_audio, recv_audio),
cricket::RtpTransceiverDirection(send_video, recv_video), &audio_index,
&video_index, &data_index, session_options);
}
// Add audio/video/data m= sections to the end if needed.
if (!audio_index && offer_new_audio_description) {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_AUDIO, cricket::CN_AUDIO,
cricket::RtpTransceiverDirection(send_audio, recv_audio), false));
audio_index = rtc::Optional<size_t>(
session_options->media_description_options.size() - 1);
}
if (!video_index && offer_new_video_description) {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_VIDEO, cricket::CN_VIDEO,
cricket::RtpTransceiverDirection(send_video, recv_video), false));
video_index = rtc::Optional<size_t>(
session_options->media_description_options.size() - 1);
}
if (!data_index && offer_new_data_description) {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_DATA, cricket::CN_DATA,
cricket::RtpTransceiverDirection(true, true), false));
data_index = rtc::Optional<size_t>(
session_options->media_description_options.size() - 1);
}
cricket::MediaDescriptionOptions* audio_media_description_options =
!audio_index ? nullptr
: &session_options->media_description_options[*audio_index];
cricket::MediaDescriptionOptions* video_media_description_options =
!video_index ? nullptr
: &session_options->media_description_options[*video_index];
cricket::MediaDescriptionOptions* data_media_description_options =
!data_index ? nullptr
: &session_options->media_description_options[*data_index];
// Apply ICE restart flag and renomination flag.
for (auto& options : session_options->media_description_options) {
options.transport_options.ice_restart = rtc_options.ice_restart;
options.transport_options.enable_ice_renomination =
configuration_.enable_ice_renomination;
}
AddRtpSenderOptions(senders_, audio_media_description_options,
video_media_description_options);
AddRtpDataChannelOptions(rtp_data_channels_, data_media_description_options);
// 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 (!rtp_data_channels_.empty() ||
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;
}
void PeerConnection::GetOptionsForAnswer(
const RTCOfferAnswerOptions& rtc_options,
cricket::MediaSessionOptions* session_options) {
ExtractSharedMediaSessionOptions(rtc_options, session_options);
// Figure out transceiver directional preferences.
bool send_audio = HasRtpSender(cricket::MEDIA_TYPE_AUDIO);
bool send_video = HasRtpSender(cricket::MEDIA_TYPE_VIDEO);
// By default, generate sendrecv/recvonly m= sections. The direction is also
// restricted by the direction in the offer.
bool recv_audio = true;
bool recv_video = true;
// The "offer_to_receive_X" options allow those defaults to be overridden.
if (rtc_options.offer_to_receive_audio != RTCOfferAnswerOptions::kUndefined) {
recv_audio = (rtc_options.offer_to_receive_audio > 0);
}
if (rtc_options.offer_to_receive_video != RTCOfferAnswerOptions::kUndefined) {
recv_video = (rtc_options.offer_to_receive_video > 0);
}
rtc::Optional<size_t> audio_index;
rtc::Optional<size_t> video_index;
rtc::Optional<size_t> data_index;
// There should be a pending remote description that's an offer...
RTC_DCHECK(session_->remote_description());
RTC_DCHECK(session_->remote_description()->type() ==
SessionDescriptionInterface::kOffer);
// Generate m= sections that match those in the offer.
// Note that mediasession.cc will handle intersection our preferred direction
// with the offered direction.
GenerateMediaDescriptionOptions(
session_->remote_description(),
cricket::RtpTransceiverDirection(send_audio, recv_audio),
cricket::RtpTransceiverDirection(send_video, recv_video), &audio_index,
&video_index, &data_index, session_options);
cricket::MediaDescriptionOptions* audio_media_description_options =
!audio_index ? nullptr
: &session_options->media_description_options[*audio_index];
cricket::MediaDescriptionOptions* video_media_description_options =
!video_index ? nullptr
: &session_options->media_description_options[*video_index];
cricket::MediaDescriptionOptions* data_media_description_options =
!data_index ? nullptr
: &session_options->media_description_options[*data_index];
// Apply ICE renomination flag.
for (auto& options : session_options->media_description_options) {
options.transport_options.enable_ice_renomination =
configuration_.enable_ice_renomination;
}
AddRtpSenderOptions(senders_, audio_media_description_options,
video_media_description_options);
AddRtpDataChannelOptions(rtp_data_channels_, data_media_description_options);
// 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 (!rtp_data_channels_.empty() ||
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;
}
void PeerConnection::GenerateMediaDescriptionOptions(
const SessionDescriptionInterface* session_desc,
cricket::RtpTransceiverDirection audio_direction,
cricket::RtpTransceiverDirection video_direction,
rtc::Optional<size_t>* audio_index,
rtc::Optional<size_t>* video_index,
rtc::Optional<size_t>* data_index,
cricket::MediaSessionOptions* session_options) {
for (const cricket::ContentInfo& content :
session_desc->description()->contents()) {
if (IsAudioContent(&content)) {
// If we already have an audio m= section, reject this extra one.
if (*audio_index) {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_AUDIO, content.name,
cricket::RtpTransceiverDirection(false, false), true));
} else {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_AUDIO, content.name, audio_direction,
!audio_direction.send && !audio_direction.recv));
*audio_index = rtc::Optional<size_t>(
session_options->media_description_options.size() - 1);
}
} else if (IsVideoContent(&content)) {
// If we already have an video m= section, reject this extra one.
if (*video_index) {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_VIDEO, content.name,
cricket::RtpTransceiverDirection(false, false), true));
} else {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_VIDEO, content.name, video_direction,
!video_direction.send && !video_direction.recv));
*video_index = rtc::Optional<size_t>(
session_options->media_description_options.size() - 1);
}
} else {
RTC_DCHECK(IsDataContent(&content));
// If we already have an data m= section, reject this extra one.
if (*data_index) {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_DATA, content.name,
cricket::RtpTransceiverDirection(false, false), true));
} else {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_DATA, content.name,
// Direction for data sections is meaningless, but legacy
// endpoints might expect sendrecv.
cricket::RtpTransceiverDirection(true, true), false));
*data_index = rtc::Optional<size_t>(
session_options->media_description_options.size() - 1);
}
}
}
}
void PeerConnection::RemoveTracks(cricket::MediaType media_type) {
UpdateLocalTracks(std::vector<cricket::StreamParams>(), media_type);
UpdateRemoteStreamsList(std::vector<cricket::StreamParams>(), false,
media_type, nullptr);
}
void PeerConnection::UpdateRemoteStreamsList(
const cricket::StreamParamsVec& streams,
bool default_track_needed,
cricket::MediaType media_type,
StreamCollection* new_streams) {
TrackInfos* current_tracks = GetRemoteTracks(media_type);
// Find removed tracks. I.e., tracks where the track id or ssrc don't match
// the new StreamParam.
auto track_it = current_tracks->begin();
while (track_it != current_tracks->end()) {
const TrackInfo& info = *track_it;
const cricket::StreamParams* params =
cricket::GetStreamBySsrc(streams, info.ssrc);
bool track_exists = params && params->id == info.track_id;
// If this is a default track, and we still need it, don't remove it.
if ((info.stream_label == kDefaultStreamLabel && default_track_needed) ||
track_exists) {
++track_it;
} else {
OnRemoteTrackRemoved(info.stream_label, info.track_id, media_type);
track_it = current_tracks->erase(track_it);
}
}
// Find new and active tracks.
for (const cricket::StreamParams& params : streams) {
// The sync_label is the MediaStream label and the |stream.id| is the
// track id.
const std::string& stream_label = params.sync_label;
const std::string& track_id = params.id;
uint32_t ssrc = params.first_ssrc();
rtc::scoped_refptr<MediaStreamInterface> stream =
remote_streams_->find(stream_label);
if (!stream) {
// This is a new MediaStream. Create a new remote MediaStream.
stream = MediaStreamProxy::Create(rtc::Thread::Current(),
MediaStream::Create(stream_label));
remote_streams_->AddStream(stream);
new_streams->AddStream(stream);
}
const TrackInfo* track_info =
FindTrackInfo(*current_tracks, stream_label, track_id);
if (!track_info) {
current_tracks->push_back(TrackInfo(stream_label, track_id, ssrc));
OnRemoteTrackSeen(stream_label, track_id, ssrc, media_type);
}
}
// Add default track if necessary.
if (default_track_needed) {
rtc::scoped_refptr<MediaStreamInterface> default_stream =
remote_streams_->find(kDefaultStreamLabel);
if (!default_stream) {
// Create the new default MediaStream.
default_stream = MediaStreamProxy::Create(
rtc::Thread::Current(), MediaStream::Create(kDefaultStreamLabel));
remote_streams_->AddStream(default_stream);
new_streams->AddStream(default_stream);
}
std::string default_track_id = (media_type == cricket::MEDIA_TYPE_AUDIO)
? kDefaultAudioTrackLabel
: kDefaultVideoTrackLabel;
const TrackInfo* default_track_info =
FindTrackInfo(*current_tracks, kDefaultStreamLabel, default_track_id);
if (!default_track_info) {
current_tracks->push_back(
TrackInfo(kDefaultStreamLabel, default_track_id, 0));
OnRemoteTrackSeen(kDefaultStreamLabel, default_track_id, 0, media_type);
}
}
}
void PeerConnection::OnRemoteTrackSeen(const std::string& stream_label,
const std::string& track_id,
uint32_t ssrc,
cricket::MediaType media_type) {
MediaStreamInterface* stream = remote_streams_->find(stream_label);
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
CreateAudioReceiver(stream, track_id, ssrc);
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
CreateVideoReceiver(stream, track_id, ssrc);
} else {
RTC_NOTREACHED() << "Invalid media type";
}
}
void PeerConnection::OnRemoteTrackRemoved(const std::string& stream_label,
const std::string& track_id,
cricket::MediaType media_type) {
MediaStreamInterface* stream = remote_streams_->find(stream_label);
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
// When the MediaEngine audio channel is destroyed, the RemoteAudioSource
// will be notified which will end the AudioRtpReceiver::track().
DestroyReceiver(track_id);
rtc::scoped_refptr<AudioTrackInterface> audio_track =
stream->FindAudioTrack(track_id);
if (audio_track) {
stream->RemoveTrack(audio_track);
}
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
// Stopping or destroying a VideoRtpReceiver will end the
// VideoRtpReceiver::track().
DestroyReceiver(track_id);
rtc::scoped_refptr<VideoTrackInterface> video_track =
stream->FindVideoTrack(track_id);
if (video_track) {
// There's no guarantee the track is still available, e.g. the track may
// have been removed from the stream by an application.
stream->RemoveTrack(video_track);
}
} else {
RTC_NOTREACHED() << "Invalid media type";
}
}
void PeerConnection::UpdateEndedRemoteMediaStreams() {
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_to_remove;
for (size_t i = 0; i < remote_streams_->count(); ++i) {
MediaStreamInterface* stream = remote_streams_->at(i);
if (stream->GetAudioTracks().empty() && stream->GetVideoTracks().empty()) {
streams_to_remove.push_back(stream);
}
}
for (auto& stream : streams_to_remove) {
remote_streams_->RemoveStream(stream);
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 (data_channel_it == rtp_data_channels_.end()) {
LOG(LS_ERROR) << "channel label not found";
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);
observer_->OnDataChannel(std::move(proxy_channel));
}
rtc::scoped_refptr<DataChannel> PeerConnection::InternalCreateDataChannel(
const std::string& label,
const InternalDataChannelInit* config) {
if (IsClosed()) {
return nullptr;
}
if (session_->data_channel_type() == cricket::DCT_NONE) {
LOG(LS_ERROR)
<< "InternalCreateDataChannel: Data is not supported in this call.";
return nullptr;
}
InternalDataChannelInit new_config =
config ? (*config) : InternalDataChannelInit();
if (session_->data_channel_type() == cricket::DCT_SCTP) {
if (new_config.id < 0) {
rtc::SSLRole role;
if ((session_->GetSctpSslRole(&role)) &&
!sid_allocator_.AllocateSid(role, &new_config.id)) {
LOG(LS_ERROR) << "No id can be allocated for the SCTP data channel.";
return nullptr;
}
} else if (!sid_allocator_.ReserveSid(new_config.id)) {
LOG(LS_ERROR) << "Failed to create a SCTP data channel "
<< "because the id is already in use or out of range.";
return nullptr;
}
}
rtc::scoped_refptr<DataChannel> channel(DataChannel::Create(
session_.get(), session_->data_channel_type(), label, new_config));
if (!channel) {
sid_allocator_.ReleaseSid(new_config.id);
return nullptr;
}
if (channel->data_channel_type() == cricket::DCT_RTP) {
if (rtp_data_channels_.find(channel->label()) != rtp_data_channels_.end()) {
LOG(LS_ERROR) << "DataChannel with label " << channel->label()
<< " already exists.";
return nullptr;
}
rtp_data_channels_[channel->label()] = channel;
} else {
RTC_DCHECK(channel->data_channel_type() == cricket::DCT_SCTP);
sctp_data_channels_.push_back(channel);
channel->SignalClosed.connect(this,
&PeerConnection::OnSctpDataChannelClosed);
}
SignalDataChannelCreated(channel.get());
return channel;
}
bool PeerConnection::HasDataChannels() const {
#ifdef HAVE_QUIC
return !rtp_data_channels_.empty() || !sctp_data_channels_.empty() ||
(session_->quic_data_transport() &&
session_->quic_data_transport()->HasDataChannels());
#else
return !rtp_data_channels_.empty() || !sctp_data_channels_.empty();
#endif // HAVE_QUIC
}
void PeerConnection::AllocateSctpSids(rtc::SSLRole role) {
for (const auto& channel : sctp_data_channels_) {
if (channel->id() < 0) {
int sid;
if (!sid_allocator_.AllocateSid(role, &sid)) {
LOG(LS_ERROR) << "Failed to allocate SCTP sid.";
continue;
}
channel->SetSctpSid(sid);
}
}
}
void PeerConnection::OnSctpDataChannelClosed(DataChannel* channel) {
RTC_DCHECK(signaling_thread()->IsCurrent());
for (auto it = sctp_data_channels_.begin(); it != sctp_data_channels_.end();
++it) {
if (it->get() == channel) {
if (channel->id() >= 0) {
sid_allocator_.ReleaseSid(channel->id());
}
// Since this method is triggered by a signal from the DataChannel,
// we can't free it directly here; we need to free it asynchronously.
sctp_data_channels_to_free_.push_back(*it);
sctp_data_channels_.erase(it);
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_FREE_DATACHANNELS,
nullptr);
return;
}
}
}
void PeerConnection::OnVoiceChannelCreated() {
SetChannelOnSendersAndReceivers<AudioRtpSender, AudioRtpReceiver>(
session_->voice_channel(), senders_, receivers_,
cricket::MEDIA_TYPE_AUDIO);
}
void PeerConnection::OnVoiceChannelDestroyed() {
SetChannelOnSendersAndReceivers<AudioRtpSender, AudioRtpReceiver,
cricket::VoiceChannel>(
nullptr, senders_, receivers_, cricket::MEDIA_TYPE_AUDIO);
}
void PeerConnection::OnVideoChannelCreated() {
SetChannelOnSendersAndReceivers<VideoRtpSender, VideoRtpReceiver>(
session_->video_channel(), senders_, receivers_,
cricket::MEDIA_TYPE_VIDEO);
}
void PeerConnection::OnVideoChannelDestroyed() {
SetChannelOnSendersAndReceivers<VideoRtpSender, VideoRtpReceiver,
cricket::VideoChannel>(
nullptr, senders_, receivers_, cricket::MEDIA_TYPE_VIDEO);
}
void PeerConnection::OnDataChannelCreated() {
for (const auto& channel : sctp_data_channels_) {
channel->OnTransportChannelCreated();
}
}
void PeerConnection::OnDataChannelDestroyed() {
// Use a temporary copy of the RTP/SCTP DataChannel list because the
// DataChannel may callback to us and try to modify the list.
std::map<std::string, rtc::scoped_refptr<DataChannel>> temp_rtp_dcs;
temp_rtp_dcs.swap(rtp_data_channels_);
for (const auto& kv : temp_rtp_dcs) {
kv.second->OnTransportChannelDestroyed();
}
std::vector<rtc::scoped_refptr<DataChannel>> temp_sctp_dcs;
temp_sctp_dcs.swap(sctp_data_channels_);
for (const auto& channel : temp_sctp_dcs) {
channel->OnTransportChannelDestroyed();
}
}
void PeerConnection::OnDataChannelOpenMessage(
const std::string& label,
const InternalDataChannelInit& config) {
rtc::scoped_refptr<DataChannel> channel(
InternalCreateDataChannel(label, &config));
if (!channel.get()) {
LOG(LS_ERROR) << "Failed to create DataChannel from the OPEN message.";
return;
}
rtc::scoped_refptr<DataChannelInterface> proxy_channel =
DataChannelProxy::Create(signaling_thread(), channel);
observer_->OnDataChannel(std::move(proxy_channel));
}
bool PeerConnection::HasRtpSender(cricket::MediaType type) const {
return std::find_if(
senders_.begin(), senders_.end(),
[type](const rtc::scoped_refptr<
RtpSenderProxyWithInternal<RtpSenderInternal>>& sender) {
return sender->media_type() == type;
}) != senders_.end();
}
RtpSenderInternal* PeerConnection::FindSenderById(const std::string& id) {
auto it = std::find_if(
senders_.begin(), senders_.end(),
[id](const rtc::scoped_refptr<
RtpSenderProxyWithInternal<RtpSenderInternal>>& sender) {
return sender->id() == id;
});
return it != senders_.end() ? (*it)->internal() : nullptr;
}
std::vector<
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>::iterator
PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) {
return std::find_if(
senders_.begin(), senders_.end(),
[track](const rtc::scoped_refptr<
RtpSenderProxyWithInternal<RtpSenderInternal>>& sender) {
return sender->track() == track;
});
}
std::vector<rtc::scoped_refptr<
RtpReceiverProxyWithInternal<RtpReceiverInternal>>>::iterator
PeerConnection::FindReceiverForTrack(const std::string& track_id) {
return std::find_if(
receivers_.begin(), receivers_.end(),
[track_id](const rtc::scoped_refptr<
RtpReceiverProxyWithInternal<RtpReceiverInternal>>& receiver) {
return receiver->id() == track_id;
});
}
PeerConnection::TrackInfos* PeerConnection::GetRemoteTracks(
cricket::MediaType media_type) {
RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
media_type == cricket::MEDIA_TYPE_VIDEO);
return (media_type == cricket::MEDIA_TYPE_AUDIO) ? &remote_audio_tracks_
: &remote_video_tracks_;
}
PeerConnection::TrackInfos* PeerConnection::GetLocalTracks(
cricket::MediaType media_type) {
RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
media_type == cricket::MEDIA_TYPE_VIDEO);
return (media_type == cricket::MEDIA_TYPE_AUDIO) ? &local_audio_tracks_
: &local_video_tracks_;
}
const PeerConnection::TrackInfo* PeerConnection::FindTrackInfo(
const PeerConnection::TrackInfos& infos,
const std::string& stream_label,
const std::string track_id) const {
for (const TrackInfo& track_info : infos) {
if (track_info.stream_label == stream_label &&
track_info.track_id == track_id) {
return &track_info;
}
}
return nullptr;
}
DataChannel* PeerConnection::FindDataChannelBySid(int sid) const {
for (const auto& channel : sctp_data_channels_) {
if (channel->id() == sid) {
return channel;
}
}
return nullptr;
}
bool PeerConnection::InitializePortAllocator_n(
const RTCConfiguration& configuration) {
cricket::ServerAddresses stun_servers;
std::vector<cricket::RelayServerConfig> turn_servers;
if (ParseIceServers(configuration.servers, &stun_servers, &turn_servers) !=
RTCErrorType::NONE) {
return false;
}
port_allocator_->Initialize();
// To handle both internal and externally created port allocator, we will
// enable BUNDLE here.
int portallocator_flags = port_allocator_->flags();
portallocator_flags |= cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
cricket::PORTALLOCATOR_ENABLE_IPV6 |
cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI;
// 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")
.find("Disabled") == 0) {
portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
}
if (configuration.disable_ipv6_on_wifi) {
portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI);
LOG(LS_INFO) << "IPv6 candidates on Wi-Fi are disabled.";
}
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));
port_allocator_->set_max_ipv6_networks(configuration.max_ipv6_networks);
// Call this last since it may create pooled allocator sessions using the
// properties set above.
port_allocator_->SetConfiguration(stun_servers, turn_servers,
configuration.ice_candidate_pool_size,
configuration.prune_turn_ports);
return true;
}
bool PeerConnection::ReconfigurePortAllocator_n(
const cricket::ServerAddresses& stun_servers,
const std::vector<cricket::RelayServerConfig>& turn_servers,
IceTransportsType type,
int candidate_pool_size,
bool prune_turn_ports) {
port_allocator_->set_candidate_filter(
ConvertIceTransportTypeToCandidateFilter(type));
// Call this last since it may create pooled allocator sessions using the
// candidate filter set above.
return port_allocator_->SetConfiguration(
stun_servers, turn_servers, candidate_pool_size, prune_turn_ports);
}
bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file,
int64_t max_size_bytes) {
if (!event_log_) {
return false;
}
return event_log_->StartLogging(file, max_size_bytes);
}
void PeerConnection::StopRtcEventLog_w() {
if (event_log_) {
event_log_->StopLogging();
}
}
} // namespace webrtc