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webrtc / src / / c06e374a55038becac4c9a24861532ae39988e11 / . / pc / peer_connection.cc
blob: 41963596a726a3e2eae856fb87e5cdae20ab7d44 [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 "pc/peer_connection.h"
#include <algorithm>
#include <limits>
#include <memory>
#include <queue>
#include <set>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
#include "api/jsep_ice_candidate.h"
#include "api/jsep_session_description.h"
#include "api/media_stream_proxy.h"
#include "api/media_stream_track_proxy.h"
#include "api/rtc_error.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtc_event_log_output_file.h"
#include "api/rtp_parameters.h"
#include "api/uma_metrics.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "call/call.h"
#include "logging/rtc_event_log/ice_logger.h"
#include "media/base/rid_description.h"
#include "media/sctp/sctp_transport.h"
#include "pc/audio_rtp_receiver.h"
#include "pc/audio_track.h"
#include "pc/channel.h"
#include "pc/channel_manager.h"
#include "pc/dtmf_sender.h"
#include "pc/media_stream.h"
#include "pc/media_stream_observer.h"
#include "pc/remote_audio_source.h"
#include "pc/rtp_media_utils.h"
#include "pc/rtp_receiver.h"
#include "pc/rtp_sender.h"
#include "pc/sctp_transport.h"
#include "pc/sctp_utils.h"
#include "pc/sdp_offer_answer.h"
#include "pc/sdp_utils.h"
#include "pc/stream_collection.h"
#include "pc/video_rtp_receiver.h"
#include "pc/video_track.h"
#include "rtc_base/bind.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/string_encode.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/task_utils/to_queued_task.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/metrics.h"
using cricket::ContentInfo;
using cricket::ContentInfos;
using cricket::MediaContentDescription;
using cricket::MediaProtocolType;
using cricket::RidDescription;
using cricket::RidDirection;
using cricket::SessionDescription;
using cricket::SimulcastDescription;
using cricket::SimulcastLayer;
using cricket::SimulcastLayerList;
using cricket::StreamParams;
using cricket::TransportInfo;
using cricket::LOCAL_PORT_TYPE;
using cricket::PRFLX_PORT_TYPE;
using cricket::RELAY_PORT_TYPE;
using cricket::STUN_PORT_TYPE;
namespace webrtc {
// Error messages
const char kSessionError[] = "Session error code: ";
const char kSessionErrorDesc[] = "Session error description: ";
namespace {
// UMA metric names.
const char kSimulcastNumberOfEncodings[] =
"WebRTC.PeerConnection.Simulcast.NumberOfSendEncodings";
static const char kDefaultStreamId[] = "default";
static const char kDefaultAudioSenderId[] = "defaulta0";
static const char kDefaultVideoSenderId[] = "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_REPORT_USAGE_PATTERN,
};
static const int REPORT_USAGE_PATTERN_DELAY_MS = 60000;
struct SetSessionDescriptionMsg : public rtc::MessageData {
explicit SetSessionDescriptionMsg(
webrtc::SetSessionDescriptionObserver* observer)
: observer(observer) {}
rtc::scoped_refptr<webrtc::SetSessionDescriptionObserver> observer;
RTCError error;
};
struct CreateSessionDescriptionMsg : public rtc::MessageData {
explicit CreateSessionDescriptionMsg(
webrtc::CreateSessionDescriptionObserver* observer)
: observer(observer) {}
rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserver> observer;
RTCError 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->id()) != nullptr) {
RTC_LOG(LS_ERROR) << "MediaStream with ID " << new_stream->id()
<< " is already added.";
return false;
}
return true;
}
// Add options to |session_options| from |rtp_data_channels|.
void AddRtpDataChannelOptions(
const std::map<std::string, rtc::scoped_refptr<RtpDataChannel>>&
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 RtpDataChannel* channel = kv.second;
if (channel->state() == RtpDataChannel::kConnecting ||
channel->state() == RtpDataChannel::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;
}
IceCandidatePairType GetIceCandidatePairCounter(
const cricket::Candidate& local,
const cricket::Candidate& remote) {
const auto& l = local.type();
const auto& r = remote.type();
const auto& host = LOCAL_PORT_TYPE;
const auto& srflx = STUN_PORT_TYPE;
const auto& relay = RELAY_PORT_TYPE;
const auto& prflx = PRFLX_PORT_TYPE;
if (l == host && r == host) {
bool local_hostname =
!local.address().hostname().empty() && local.address().IsUnresolvedIP();
bool remote_hostname = !remote.address().hostname().empty() &&
remote.address().IsUnresolvedIP();
bool local_private = IPIsPrivate(local.address().ipaddr());
bool remote_private = IPIsPrivate(remote.address().ipaddr());
if (local_hostname) {
if (remote_hostname) {
return kIceCandidatePairHostNameHostName;
} else if (remote_private) {
return kIceCandidatePairHostNameHostPrivate;
} else {
return kIceCandidatePairHostNameHostPublic;
}
} else if (local_private) {
if (remote_hostname) {
return kIceCandidatePairHostPrivateHostName;
} else if (remote_private) {
return kIceCandidatePairHostPrivateHostPrivate;
} else {
return kIceCandidatePairHostPrivateHostPublic;
}
} else {
if (remote_hostname) {
return kIceCandidatePairHostPublicHostName;
} else if (remote_private) {
return kIceCandidatePairHostPublicHostPrivate;
} else {
return kIceCandidatePairHostPublicHostPublic;
}
}
}
if (l == host && r == srflx)
return kIceCandidatePairHostSrflx;
if (l == host && r == relay)
return kIceCandidatePairHostRelay;
if (l == host && r == prflx)
return kIceCandidatePairHostPrflx;
if (l == srflx && r == host)
return kIceCandidatePairSrflxHost;
if (l == srflx && r == srflx)
return kIceCandidatePairSrflxSrflx;
if (l == srflx && r == relay)
return kIceCandidatePairSrflxRelay;
if (l == srflx && r == prflx)
return kIceCandidatePairSrflxPrflx;
if (l == relay && r == host)
return kIceCandidatePairRelayHost;
if (l == relay && r == srflx)
return kIceCandidatePairRelaySrflx;
if (l == relay && r == relay)
return kIceCandidatePairRelayRelay;
if (l == relay && r == prflx)
return kIceCandidatePairRelayPrflx;
if (l == prflx && r == host)
return kIceCandidatePairPrflxHost;
if (l == prflx && r == srflx)
return kIceCandidatePairPrflxSrflx;
if (l == prflx && r == relay)
return kIceCandidatePairPrflxRelay;
return kIceCandidatePairMax;
}
absl::optional<int> RTCConfigurationToIceConfigOptionalInt(
int rtc_configuration_parameter) {
if (rtc_configuration_parameter ==
webrtc::PeerConnectionInterface::RTCConfiguration::kUndefined) {
return absl::nullopt;
}
return rtc_configuration_parameter;
}
// Check if the changes of IceTransportsType motives an ice restart.
bool NeedIceRestart(bool surface_ice_candidates_on_ice_transport_type_changed,
PeerConnectionInterface::IceTransportsType current,
PeerConnectionInterface::IceTransportsType modified) {
if (current == modified) {
return false;
}
if (!surface_ice_candidates_on_ice_transport_type_changed) {
return true;
}
auto current_filter = ConvertIceTransportTypeToCandidateFilter(current);
auto modified_filter = ConvertIceTransportTypeToCandidateFilter(modified);
// If surface_ice_candidates_on_ice_transport_type_changed is true and we
// extend the filter, then no ice restart is needed.
return (current_filter & modified_filter) != current_filter;
}
} // namespace
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 disable_link_local_networks;
bool enable_rtp_data_channel;
absl::optional<int> screencast_min_bitrate;
absl::optional<bool> combined_audio_video_bwe;
absl::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 audio_jitter_buffer_min_delay_ms;
bool audio_jitter_buffer_enable_rtx_handling;
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 prune_turn_ports;
PortPrunePolicy turn_port_prune_policy;
bool presume_writable_when_fully_relayed;
bool enable_ice_renomination;
bool redetermine_role_on_ice_restart;
bool surface_ice_candidates_on_ice_transport_type_changed;
absl::optional<int> ice_check_interval_strong_connectivity;
absl::optional<int> ice_check_interval_weak_connectivity;
absl::optional<int> ice_check_min_interval;
absl::optional<int> ice_unwritable_timeout;
absl::optional<int> ice_unwritable_min_checks;
absl::optional<int> ice_inactive_timeout;
absl::optional<int> stun_candidate_keepalive_interval;
webrtc::TurnCustomizer* turn_customizer;
SdpSemantics sdp_semantics;
absl::optional<rtc::AdapterType> network_preference;
bool active_reset_srtp_params;
absl::optional<CryptoOptions> crypto_options;
bool offer_extmap_allow_mixed;
std::string turn_logging_id;
bool enable_implicit_rollback;
absl::optional<bool> allow_codec_switching;
};
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 &&
audio_jitter_buffer_min_delay_ms ==
o.audio_jitter_buffer_min_delay_ms &&
audio_jitter_buffer_enable_rtx_handling ==
o.audio_jitter_buffer_enable_rtx_handling &&
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 &&
disable_link_local_networks == o.disable_link_local_networks &&
enable_rtp_data_channel == o.enable_rtp_data_channel &&
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 &&
turn_port_prune_policy == o.turn_port_prune_policy &&
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 &&
surface_ice_candidates_on_ice_transport_type_changed ==
o.surface_ice_candidates_on_ice_transport_type_changed &&
ice_check_interval_strong_connectivity ==
o.ice_check_interval_strong_connectivity &&
ice_check_interval_weak_connectivity ==
o.ice_check_interval_weak_connectivity &&
ice_check_min_interval == o.ice_check_min_interval &&
ice_unwritable_timeout == o.ice_unwritable_timeout &&
ice_unwritable_min_checks == o.ice_unwritable_min_checks &&
ice_inactive_timeout == o.ice_inactive_timeout &&
stun_candidate_keepalive_interval ==
o.stun_candidate_keepalive_interval &&
turn_customizer == o.turn_customizer &&
sdp_semantics == o.sdp_semantics &&
network_preference == o.network_preference &&
active_reset_srtp_params == o.active_reset_srtp_params &&
crypto_options == o.crypto_options &&
offer_extmap_allow_mixed == o.offer_extmap_allow_mixed &&
turn_logging_id == o.turn_logging_id &&
enable_implicit_rollback == o.enable_implicit_rollback &&
allow_codec_switching == o.allow_codec_switching;
}
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)) {
RTC_LOG(LS_ERROR) << "Failed to generate CNAME.";
RTC_NOTREACHED();
}
return cname;
}
PeerConnection::PeerConnection(PeerConnectionFactory* factory,
std::unique_ptr<RtcEventLog> event_log,
std::unique_ptr<Call> call)
: factory_(factory),
event_log_(std::move(event_log)),
event_log_ptr_(event_log_.get()),
rtcp_cname_(GenerateRtcpCname()),
local_streams_(StreamCollection::Create()),
remote_streams_(StreamCollection::Create()),
call_(std::move(call)),
call_ptr_(call_.get()),
sdp_handler_(this),
data_channel_controller_(this) {}
PeerConnection::~PeerConnection() {
TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_.PrepareForShutdown();
// Need to stop transceivers before destroying the stats collector because
// AudioRtpSender has a reference to the StatsCollector it will update when
// stopping.
for (const auto& transceiver : transceivers_.List()) {
transceiver->StopInternal();
}
stats_.reset(nullptr);
if (stats_collector_) {
stats_collector_->WaitForPendingRequest();
stats_collector_ = nullptr;
}
// Don't destroy BaseChannels until after stats has been cleaned up so that
// the last stats request can still read from the channels.
DestroyAllChannels();
RTC_LOG(LS_INFO) << "Session: " << session_id() << " is destroyed.";
sdp_handler_.ResetSessionDescFactory();
transport_controller_.reset();
// port_allocator_ lives on the network thread and should be destroyed there.
network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
RTC_DCHECK_RUN_ON(network_thread());
port_allocator_.reset();
});
// call_ and event_log_ must be destroyed on the worker thread.
worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
RTC_DCHECK_RUN_ON(worker_thread());
call_safety_.reset();
call_.reset();
// The event log must outlive call (and any other object that uses it).
event_log_.reset();
});
// Process all pending notifications in the message queue. If we don't do
// this, requests will linger and not know they succeeded or failed.
rtc::MessageList list;
signaling_thread()->Clear(this, rtc::MQID_ANY, &list);
for (auto& msg : list) {
if (msg.message_id == MSG_CREATE_SESSIONDESCRIPTION_FAILED) {
// Processing CreateOffer() and CreateAnswer() messages ensures their
// observers are invoked even if the PeerConnection is destroyed early.
OnMessage(&msg);
} else {
// TODO(hbos): Consider processing all pending messages. This would mean
// that SetLocalDescription() and SetRemoteDescription() observers are
// informed of successes and failures; this is currently NOT the case.
delete msg.pdata;
}
}
}
void PeerConnection::DestroyAllChannels() {
// Destroy video channels first since they may have a pointer to a voice
// channel.
for (const auto& transceiver : transceivers_.List()) {
if (transceiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
DestroyTransceiverChannel(transceiver);
}
}
for (const auto& transceiver : transceivers_.List()) {
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
DestroyTransceiverChannel(transceiver);
}
}
DestroyDataChannelTransport();
}
bool PeerConnection::Initialize(
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies dependencies) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::Initialize");
RTCError config_error = ValidateConfiguration(configuration);
if (!config_error.ok()) {
RTC_LOG(LS_ERROR) << "Invalid configuration: " << config_error.message();
return false;
}
if (!dependencies.allocator) {
RTC_LOG(LS_ERROR)
<< "PeerConnection initialized without a PortAllocator? "
"This shouldn't happen if using PeerConnectionFactory.";
return false;
}
if (!dependencies.observer) {
// TODO(deadbeef): Why do we do this?
RTC_LOG(LS_ERROR) << "PeerConnection initialized without a "
"PeerConnectionObserver";
return false;
}
observer_ = dependencies.observer;
async_resolver_factory_ = std::move(dependencies.async_resolver_factory);
port_allocator_ = std::move(dependencies.allocator);
packet_socket_factory_ = std::move(dependencies.packet_socket_factory);
ice_transport_factory_ = std::move(dependencies.ice_transport_factory);
tls_cert_verifier_ = std::move(dependencies.tls_cert_verifier);
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 false;
}
// Add the turn logging id to all turn servers
for (cricket::RelayServerConfig& turn_server : turn_servers) {
turn_server.turn_logging_id = configuration.turn_logging_id;
}
// The port allocator lives on the network thread and should be initialized
// there.
const auto pa_result =
network_thread()->Invoke<InitializePortAllocatorResult>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnection::InitializePortAllocator_n, this,
stun_servers, turn_servers, configuration));
// If initialization was successful, note if STUN or TURN servers
// were supplied.
if (!stun_servers.empty()) {
NoteUsageEvent(UsageEvent::STUN_SERVER_ADDED);
}
if (!turn_servers.empty()) {
NoteUsageEvent(UsageEvent::TURN_SERVER_ADDED);
}
// Send information about IPv4/IPv6 status.
PeerConnectionAddressFamilyCounter address_family;
if (pa_result.enable_ipv6) {
address_family = kPeerConnection_IPv6;
} else {
address_family = kPeerConnection_IPv4;
}
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics", address_family,
kPeerConnectionAddressFamilyCounter_Max);
const PeerConnectionFactoryInterface::Options& options = factory_->options();
// RFC 3264: The numeric value of the session id and version in the
// o line MUST be representable with a "64 bit signed integer".
// Due to this constraint session id |session_id_| is max limited to
// LLONG_MAX.
session_id_ = rtc::ToString(rtc::CreateRandomId64() & LLONG_MAX);
JsepTransportController::Config config;
config.redetermine_role_on_ice_restart =
configuration.redetermine_role_on_ice_restart;
config.ssl_max_version = factory_->options().ssl_max_version;
config.disable_encryption = options.disable_encryption;
config.bundle_policy = configuration.bundle_policy;
config.rtcp_mux_policy = configuration.rtcp_mux_policy;
// TODO(bugs.webrtc.org/9891) - Remove options.crypto_options then remove this
// stub.
config.crypto_options = configuration.crypto_options.has_value()
? *configuration.crypto_options
: options.crypto_options;
config.transport_observer = this;
config.rtcp_handler = InitializeRtcpCallback();
config.event_log = event_log_ptr_;
#if defined(ENABLE_EXTERNAL_AUTH)
config.enable_external_auth = true;
#endif
config.active_reset_srtp_params = configuration.active_reset_srtp_params;
// Obtain a certificate from RTCConfiguration if any were provided (optional).
rtc::scoped_refptr<rtc::RTCCertificate> certificate;
if (!configuration.certificates.empty()) {
// TODO(hbos,torbjorng): Decide on certificate-selection strategy instead of
// just picking the first one. The decision should be made based on the DTLS
// handshake. The DTLS negotiations need to know about all certificates.
certificate = configuration.certificates[0];
}
if (options.disable_encryption) {
dtls_enabled_ = false;
} else {
// Enable DTLS by default if we have an identity store or a certificate.
dtls_enabled_ = (dependencies.cert_generator || certificate);
// |configuration| can override the default |dtls_enabled_| value.
if (configuration.enable_dtls_srtp) {
dtls_enabled_ = *(configuration.enable_dtls_srtp);
}
}
if (configuration.enable_rtp_data_channel) {
// Enable creation of RTP data channels if the kEnableRtpDataChannels is
// set. It takes precendence over the disable_sctp_data_channels
// PeerConnectionFactoryInterface::Options.
data_channel_controller_.set_data_channel_type(cricket::DCT_RTP);
} else {
// DTLS has to be enabled to use SCTP.
if (!options.disable_sctp_data_channels && dtls_enabled_) {
data_channel_controller_.set_data_channel_type(cricket::DCT_SCTP);
config.sctp_factory = factory_->sctp_transport_factory();
}
}
config.ice_transport_factory = ice_transport_factory_.get();
transport_controller_.reset(new JsepTransportController(
signaling_thread(), network_thread(), port_allocator_.get(),
async_resolver_factory_.get(), config));
transport_controller_->SignalStandardizedIceConnectionState.connect(
this, &PeerConnection::SetStandardizedIceConnectionState);
transport_controller_->SignalConnectionState.connect(
this, &PeerConnection::SetConnectionState);
transport_controller_->SignalIceGatheringState.connect(
this, &PeerConnection::OnTransportControllerGatheringState);
transport_controller_->SignalIceCandidatesGathered.connect(
this, &PeerConnection::OnTransportControllerCandidatesGathered);
transport_controller_->SignalIceCandidateError.connect(
this, &PeerConnection::OnTransportControllerCandidateError);
transport_controller_->SignalIceCandidatesRemoved.connect(
this, &PeerConnection::OnTransportControllerCandidatesRemoved);
transport_controller_->SignalDtlsHandshakeError.connect(
this, &PeerConnection::OnTransportControllerDtlsHandshakeError);
transport_controller_->SignalIceCandidatePairChanged.connect(
this, &PeerConnection::OnTransportControllerCandidateChanged);
transport_controller_->SignalIceConnectionState.AddReceiver(
[this](cricket::IceConnectionState s) {
RTC_DCHECK_RUN_ON(signaling_thread());
OnTransportControllerConnectionState(s);
});
stats_.reset(new StatsCollector(this));
stats_collector_ = RTCStatsCollector::Create(this);
configuration_ = configuration;
transport_controller_->SetIceConfig(ParseIceConfig(configuration));
video_options_.screencast_min_bitrate_kbps =
configuration.screencast_min_bitrate;
audio_options_.combined_audio_video_bwe =
configuration.combined_audio_video_bwe;
audio_options_.audio_jitter_buffer_max_packets =
configuration.audio_jitter_buffer_max_packets;
audio_options_.audio_jitter_buffer_fast_accelerate =
configuration.audio_jitter_buffer_fast_accelerate;
audio_options_.audio_jitter_buffer_min_delay_ms =
configuration.audio_jitter_buffer_min_delay_ms;
audio_options_.audio_jitter_buffer_enable_rtx_handling =
configuration.audio_jitter_buffer_enable_rtx_handling;
// Whether the certificate generator/certificate is null or not determines
// what PeerConnectionDescriptionFactory will do, so make sure that we give it
// the right instructions by clearing the variables if needed.
if (!dtls_enabled_) {
dependencies.cert_generator.reset();
certificate = nullptr;
} else if (certificate) {
// Favor generated certificate over the certificate generator.
dependencies.cert_generator.reset();
}
auto webrtc_session_desc_factory =
std::make_unique<WebRtcSessionDescriptionFactory>(
signaling_thread(), channel_manager(), this, session_id(),
std::move(dependencies.cert_generator), certificate,
&ssrc_generator_);
webrtc_session_desc_factory->SignalCertificateReady.connect(
this, &PeerConnection::OnCertificateReady);
if (options.disable_encryption) {
webrtc_session_desc_factory->SetSdesPolicy(cricket::SEC_DISABLED);
}
webrtc_session_desc_factory->set_enable_encrypted_rtp_header_extensions(
GetCryptoOptions().srtp.enable_encrypted_rtp_header_extensions);
webrtc_session_desc_factory->set_is_unified_plan(IsUnifiedPlan());
sdp_handler_.SetSessionDescFactory(std::move(webrtc_session_desc_factory));
// Add default audio/video transceivers for Plan B SDP.
if (!IsUnifiedPlan()) {
transceivers_.Add(RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
signaling_thread(), new RtpTransceiver(cricket::MEDIA_TYPE_AUDIO)));
transceivers_.Add(RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
signaling_thread(), new RtpTransceiver(cricket::MEDIA_TYPE_VIDEO)));
}
int delay_ms =
return_histogram_very_quickly_ ? 0 : REPORT_USAGE_PATTERN_DELAY_MS;
signaling_thread()->PostDelayed(RTC_FROM_HERE, delay_ms, this,
MSG_REPORT_USAGE_PATTERN, nullptr);
if (dependencies.video_bitrate_allocator_factory) {
video_bitrate_allocator_factory_ =
std::move(dependencies.video_bitrate_allocator_factory);
} else {
video_bitrate_allocator_factory_ =
CreateBuiltinVideoBitrateAllocatorFactory();
}
return true;
}
RTCError PeerConnection::ValidateConfiguration(
const RTCConfiguration& config) const {
return cricket::P2PTransportChannel::ValidateIceConfig(
ParseIceConfig(config));
}
rtc::scoped_refptr<StreamCollectionInterface> PeerConnection::local_streams() {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(!IsUnifiedPlan()) << "local_streams is not available with Unified "
"Plan SdpSemantics. Please use GetSenders "
"instead.";
return local_streams_;
}
rtc::scoped_refptr<StreamCollectionInterface> PeerConnection::remote_streams() {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(!IsUnifiedPlan()) << "remote_streams is not available with Unified "
"Plan SdpSemantics. Please use GetReceivers "
"instead.";
return remote_streams_;
}
bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(!IsUnifiedPlan()) << "AddStream is not available with Unified Plan "
"SdpSemantics. Please use AddTrack instead.";
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);
sdp_handler_.UpdateNegotiationNeeded();
return true;
}
void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(!IsUnifiedPlan()) << "RemoveStream is not available with Unified "
"Plan SdpSemantics. Please use RemoveTrack "
"instead.";
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()->id().compare(local_stream->id()) == 0;
}),
stream_observers_.end());
if (IsClosed()) {
return;
}
sdp_handler_.UpdateNegotiationNeeded();
}
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::AddTrack(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::AddTrack");
if (!track) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "Track is null.");
}
if (!(track->kind() == MediaStreamTrackInterface::kAudioKind ||
track->kind() == MediaStreamTrackInterface::kVideoKind)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Track has invalid kind: " + track->kind());
}
if (IsClosed()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
"PeerConnection is closed.");
}
if (FindSenderForTrack(track)) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"Sender already exists for track " + track->id() + ".");
}
auto sender_or_error =
(IsUnifiedPlan() ? AddTrackUnifiedPlan(track, stream_ids)
: AddTrackPlanB(track, stream_ids));
if (sender_or_error.ok()) {
sdp_handler_.UpdateNegotiationNeeded();
stats_->AddTrack(track);
}
return sender_or_error;
}
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>
PeerConnection::AddTrackPlanB(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids) {
if (stream_ids.size() > 1u) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
"AddTrack with more than one stream is not "
"supported with Plan B semantics.");
}
std::vector<std::string> adjusted_stream_ids = stream_ids;
if (adjusted_stream_ids.empty()) {
adjusted_stream_ids.push_back(rtc::CreateRandomUuid());
}
cricket::MediaType media_type =
(track->kind() == MediaStreamTrackInterface::kAudioKind
? cricket::MEDIA_TYPE_AUDIO
: cricket::MEDIA_TYPE_VIDEO);
auto new_sender =
CreateSender(media_type, track->id(), track, adjusted_stream_ids, {});
if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
new_sender->internal()->SetMediaChannel(voice_media_channel());
GetAudioTransceiver()->internal()->AddSender(new_sender);
const RtpSenderInfo* sender_info =
FindSenderInfo(local_audio_sender_infos_,
new_sender->internal()->stream_ids()[0], track->id());
if (sender_info) {
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
}
} else {
RTC_DCHECK_EQ(MediaStreamTrackInterface::kVideoKind, track->kind());
new_sender->internal()->SetMediaChannel(video_media_channel());
GetVideoTransceiver()->internal()->AddSender(new_sender);
const RtpSenderInfo* sender_info =
FindSenderInfo(local_video_sender_infos_,
new_sender->internal()->stream_ids()[0], track->id());
if (sender_info) {
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
}
}
return rtc::scoped_refptr<RtpSenderInterface>(new_sender);
}
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>
PeerConnection::AddTrackUnifiedPlan(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids) {
auto transceiver = FindFirstTransceiverForAddedTrack(track);
if (transceiver) {
RTC_LOG(LS_INFO) << "Reusing an existing "
<< cricket::MediaTypeToString(transceiver->media_type())
<< " transceiver for AddTrack.";
if (transceiver->stopping()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"The existing transceiver is stopping.");
}
if (transceiver->direction() == RtpTransceiverDirection::kRecvOnly) {
transceiver->internal()->set_direction(
RtpTransceiverDirection::kSendRecv);
} else if (transceiver->direction() == RtpTransceiverDirection::kInactive) {
transceiver->internal()->set_direction(
RtpTransceiverDirection::kSendOnly);
}
transceiver->sender()->SetTrack(track);
transceiver->internal()->sender_internal()->set_stream_ids(stream_ids);
transceiver->internal()->set_reused_for_addtrack(true);
} else {
cricket::MediaType media_type =
(track->kind() == MediaStreamTrackInterface::kAudioKind
? cricket::MEDIA_TYPE_AUDIO
: cricket::MEDIA_TYPE_VIDEO);
RTC_LOG(LS_INFO) << "Adding " << cricket::MediaTypeToString(media_type)
<< " transceiver in response to a call to AddTrack.";
std::string sender_id = track->id();
// Avoid creating a sender with an existing ID by generating a random ID.
// This can happen if this is the second time AddTrack has created a sender
// for this track.
if (FindSenderById(sender_id)) {
sender_id = rtc::CreateRandomUuid();
}
auto sender = CreateSender(media_type, sender_id, track, stream_ids, {});
auto receiver = CreateReceiver(media_type, rtc::CreateRandomUuid());
transceiver = CreateAndAddTransceiver(sender, receiver);
transceiver->internal()->set_created_by_addtrack(true);
transceiver->internal()->set_direction(RtpTransceiverDirection::kSendRecv);
}
return transceiver->sender();
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::FindFirstTransceiverForAddedTrack(
rtc::scoped_refptr<MediaStreamTrackInterface> track) {
RTC_DCHECK(track);
for (auto transceiver : transceivers_.List()) {
if (!transceiver->sender()->track() &&
cricket::MediaTypeToString(transceiver->media_type()) ==
track->kind() &&
!transceiver->internal()->has_ever_been_used_to_send() &&
!transceiver->stopped()) {
return transceiver;
}
}
return nullptr;
}
bool PeerConnection::RemoveTrack(RtpSenderInterface* sender) {
TRACE_EVENT0("webrtc", "PeerConnection::RemoveTrack");
return RemoveTrackNew(sender).ok();
}
RTCError PeerConnection::RemoveTrackNew(
rtc::scoped_refptr<RtpSenderInterface> sender) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!sender) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "Sender is null.");
}
if (IsClosed()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
"PeerConnection is closed.");
}
if (IsUnifiedPlan()) {
auto transceiver = FindTransceiverBySender(sender);
if (!transceiver || !sender->track()) {
return RTCError::OK();
}
sender->SetTrack(nullptr);
if (transceiver->direction() == RtpTransceiverDirection::kSendRecv) {
transceiver->internal()->set_direction(
RtpTransceiverDirection::kRecvOnly);
} else if (transceiver->direction() == RtpTransceiverDirection::kSendOnly) {
transceiver->internal()->set_direction(
RtpTransceiverDirection::kInactive);
}
} else {
bool removed;
if (sender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
removed = GetAudioTransceiver()->internal()->RemoveSender(sender);
} else {
RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, sender->media_type());
removed = GetVideoTransceiver()->internal()->RemoveSender(sender);
}
if (!removed) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"Couldn't find sender " + sender->id() + " to remove.");
}
}
sdp_handler_.UpdateNegotiationNeeded();
return RTCError::OK();
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::FindTransceiverBySender(
rtc::scoped_refptr<RtpSenderInterface> sender) {
return transceivers_.FindBySender(sender);
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track) {
return AddTransceiver(track, RtpTransceiverInit());
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(IsUnifiedPlan())
<< "AddTransceiver is only available with Unified Plan SdpSemantics";
if (!track) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "track is null");
}
cricket::MediaType media_type;
if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
media_type = cricket::MEDIA_TYPE_AUDIO;
} else if (track->kind() == MediaStreamTrackInterface::kVideoKind) {
media_type = cricket::MEDIA_TYPE_VIDEO;
} else {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Track kind is not audio or video");
}
return AddTransceiver(media_type, track, init);
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(cricket::MediaType media_type) {
return AddTransceiver(media_type, RtpTransceiverInit());
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(cricket::MediaType media_type,
const RtpTransceiverInit& init) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(IsUnifiedPlan())
<< "AddTransceiver is only available with Unified Plan SdpSemantics";
if (!(media_type == cricket::MEDIA_TYPE_AUDIO ||
media_type == cricket::MEDIA_TYPE_VIDEO)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"media type is not audio or video");
}
return AddTransceiver(media_type, nullptr, init);
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(
cricket::MediaType media_type,
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init,
bool update_negotiation_needed) {
RTC_DCHECK((media_type == cricket::MEDIA_TYPE_AUDIO ||
media_type == cricket::MEDIA_TYPE_VIDEO));
if (track) {
RTC_DCHECK_EQ(media_type,
(track->kind() == MediaStreamTrackInterface::kAudioKind
? cricket::MEDIA_TYPE_AUDIO
: cricket::MEDIA_TYPE_VIDEO));
}
RTC_HISTOGRAM_COUNTS_LINEAR(kSimulcastNumberOfEncodings,
init.send_encodings.size(), 0, 7, 8);
size_t num_rids = absl::c_count_if(init.send_encodings,
[](const RtpEncodingParameters& encoding) {
return !encoding.rid.empty();
});
if (num_rids > 0 && num_rids != init.send_encodings.size()) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"RIDs must be provided for either all or none of the send encodings.");
}
if (num_rids > 0 && absl::c_any_of(init.send_encodings,
[](const RtpEncodingParameters& encoding) {
return !IsLegalRsidName(encoding.rid);
})) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Invalid RID value provided.");
}
if (absl::c_any_of(init.send_encodings,
[](const RtpEncodingParameters& encoding) {
return encoding.ssrc.has_value();
})) {
LOG_AND_RETURN_ERROR(
RTCErrorType::UNSUPPORTED_PARAMETER,
"Attempted to set an unimplemented parameter of RtpParameters.");
}
RtpParameters parameters;
parameters.encodings = init.send_encodings;
// Encodings are dropped from the tail if too many are provided.
if (parameters.encodings.size() > kMaxSimulcastStreams) {
parameters.encodings.erase(
parameters.encodings.begin() + kMaxSimulcastStreams,
parameters.encodings.end());
}
// Single RID should be removed.
if (parameters.encodings.size() == 1 &&
!parameters.encodings[0].rid.empty()) {
RTC_LOG(LS_INFO) << "Removing RID: " << parameters.encodings[0].rid << ".";
parameters.encodings[0].rid.clear();
}
// If RIDs were not provided, they are generated for simulcast scenario.
if (parameters.encodings.size() > 1 && num_rids == 0) {
rtc::UniqueStringGenerator rid_generator;
for (RtpEncodingParameters& encoding : parameters.encodings) {
encoding.rid = rid_generator();
}
}
if (UnimplementedRtpParameterHasValue(parameters)) {
LOG_AND_RETURN_ERROR(
RTCErrorType::UNSUPPORTED_PARAMETER,
"Attempted to set an unimplemented parameter of RtpParameters.");
}
auto result = cricket::CheckRtpParametersValues(parameters);
if (!result.ok()) {
LOG_AND_RETURN_ERROR(result.type(), result.message());
}
RTC_LOG(LS_INFO) << "Adding " << cricket::MediaTypeToString(media_type)
<< " transceiver in response to a call to AddTransceiver.";
// Set the sender ID equal to the track ID if the track is specified unless
// that sender ID is already in use.
std::string sender_id =
(track && !FindSenderById(track->id()) ? track->id()
: rtc::CreateRandomUuid());
auto sender = CreateSender(media_type, sender_id, track, init.stream_ids,
parameters.encodings);
auto receiver = CreateReceiver(media_type, rtc::CreateRandomUuid());
auto transceiver = CreateAndAddTransceiver(sender, receiver);
transceiver->internal()->set_direction(init.direction);
if (update_negotiation_needed) {
sdp_handler_.UpdateNegotiationNeeded();
}
return rtc::scoped_refptr<RtpTransceiverInterface>(transceiver);
}
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
PeerConnection::CreateSender(
cricket::MediaType media_type,
const std::string& id,
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids,
const std::vector<RtpEncodingParameters>& send_encodings) {
RTC_DCHECK_RUN_ON(signaling_thread());
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender;
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
RTC_DCHECK(!track ||
(track->kind() == MediaStreamTrackInterface::kAudioKind));
sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(),
AudioRtpSender::Create(worker_thread(), id, stats_.get(), this));
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
} else {
RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
RTC_DCHECK(!track ||
(track->kind() == MediaStreamTrackInterface::kVideoKind));
sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(), VideoRtpSender::Create(worker_thread(), id, this));
NoteUsageEvent(UsageEvent::VIDEO_ADDED);
}
bool set_track_succeeded = sender->SetTrack(track);
RTC_DCHECK(set_track_succeeded);
sender->internal()->set_stream_ids(stream_ids);
sender->internal()->set_init_send_encodings(send_encodings);
return sender;
}
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
PeerConnection::CreateReceiver(cricket::MediaType media_type,
const std::string& receiver_id) {
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
receiver;
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(), new AudioRtpReceiver(worker_thread(), receiver_id,
std::vector<std::string>({})));
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
} else {
RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(), new VideoRtpReceiver(worker_thread(), receiver_id,
std::vector<std::string>({})));
NoteUsageEvent(UsageEvent::VIDEO_ADDED);
}
return receiver;
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::CreateAndAddTransceiver(
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender,
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
receiver) {
RTC_DCHECK_RUN_ON(signaling_thread());
// Ensure that the new sender does not have an ID that is already in use by
// another sender.
// Allow receiver IDs to conflict since those come from remote SDP (which
// could be invalid, but should not cause a crash).
RTC_DCHECK(!FindSenderById(sender->id()));
auto transceiver = RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
signaling_thread(),
new RtpTransceiver(
sender, receiver, channel_manager(),
sender->media_type() == cricket::MEDIA_TYPE_AUDIO
? channel_manager()->GetSupportedAudioRtpHeaderExtensions()
: channel_manager()->GetSupportedVideoRtpHeaderExtensions()));
transceivers_.Add(transceiver);
transceiver->internal()->SignalNegotiationNeeded.connect(
this, &PeerConnection::OnNegotiationNeeded);
return transceiver;
}
void PeerConnection::OnNegotiationNeeded() {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(!IsClosed());
sdp_handler_.UpdateNegotiationNeeded();
}
rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
const std::string& kind,
const std::string& stream_id) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(!IsUnifiedPlan()) << "CreateSender is not available with Unified "
"Plan SdpSemantics. Please use AddTransceiver "
"instead.";
TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
if (IsClosed()) {
return nullptr;
}
// Internally we need to have one stream with Plan B semantics, so we
// generate a random stream ID if not specified.
std::vector<std::string> stream_ids;
if (stream_id.empty()) {
stream_ids.push_back(rtc::CreateRandomUuid());
RTC_LOG(LS_INFO)
<< "No stream_id specified for sender. Generated stream ID: "
<< stream_ids[0];
} else {
stream_ids.push_back(stream_id);
}
// TODO(steveanton): Move construction of the RtpSenders to RtpTransceiver.
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender;
if (kind == MediaStreamTrackInterface::kAudioKind) {
auto audio_sender = AudioRtpSender::Create(
worker_thread(), rtc::CreateRandomUuid(), stats_.get(), this);
audio_sender->SetMediaChannel(voice_media_channel());
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(), audio_sender);
GetAudioTransceiver()->internal()->AddSender(new_sender);
} else if (kind == MediaStreamTrackInterface::kVideoKind) {
auto video_sender =
VideoRtpSender::Create(worker_thread(), rtc::CreateRandomUuid(), this);
video_sender->SetMediaChannel(video_media_channel());
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(), video_sender);
GetVideoTransceiver()->internal()->AddSender(new_sender);
} else {
RTC_LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
return nullptr;
}
new_sender->internal()->set_stream_ids(stream_ids);
return new_sender;
}
std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
std::vector<rtc::scoped_refptr<RtpSenderInterface>> ret;
for (const auto& sender : GetSendersInternal()) {
ret.push_back(sender);
}
return ret;
}
std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
PeerConnection::GetSendersInternal() const {
RTC_DCHECK_RUN_ON(signaling_thread());
std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
all_senders;
for (const auto& transceiver : transceivers_.List()) {
if (IsUnifiedPlan() && transceiver->internal()->stopped())
continue;
auto senders = transceiver->internal()->senders();
all_senders.insert(all_senders.end(), senders.begin(), senders.end());
}
return all_senders;
}
std::vector<rtc::scoped_refptr<RtpReceiverInterface>>
PeerConnection::GetReceivers() const {
RTC_DCHECK_RUN_ON(signaling_thread());
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> ret;
for (const auto& receiver : GetReceiversInternal()) {
ret.push_back(receiver);
}
return ret;
}
std::vector<
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
PeerConnection::GetReceiversInternal() const {
std::vector<
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
all_receivers;
for (const auto& transceiver : transceivers_.List()) {
if (IsUnifiedPlan() && transceiver->internal()->stopped())
continue;
auto receivers = transceiver->internal()->receivers();
all_receivers.insert(all_receivers.end(), receivers.begin(),
receivers.end());
}
return all_receivers;
}
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::GetTransceivers() const {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(IsUnifiedPlan())
<< "GetTransceivers is only supported with Unified Plan SdpSemantics.";
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> all_transceivers;
for (const auto& transceiver : transceivers_.List()) {
all_transceivers.push_back(transceiver);
}
return all_transceivers;
}
bool PeerConnection::GetStats(StatsObserver* observer,
MediaStreamTrackInterface* track,
StatsOutputLevel level) {
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
RTC_DCHECK_RUN_ON(signaling_thread());
if (!observer) {
RTC_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())) {
RTC_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) {
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(stats_collector_);
RTC_DCHECK(callback);
stats_collector_->GetStatsReport(callback);
}
void PeerConnection::GetStats(
rtc::scoped_refptr<RtpSenderInterface> selector,
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(callback);
RTC_DCHECK(stats_collector_);
rtc::scoped_refptr<RtpSenderInternal> internal_sender;
if (selector) {
for (const auto& proxy_transceiver : transceivers_.List()) {
for (const auto& proxy_sender :
proxy_transceiver->internal()->senders()) {
if (proxy_sender == selector) {
internal_sender = proxy_sender->internal();
break;
}
}
if (internal_sender)
break;
}
}
// If there is no |internal_sender| then |selector| is either null or does not
// belong to the PeerConnection (in Plan B, senders can be removed from the
// PeerConnection). This means that "all the stats objects representing the
// selector" is an empty set. Invoking GetStatsReport() with a null selector
// produces an empty stats report.
stats_collector_->GetStatsReport(internal_sender, callback);
}
void PeerConnection::GetStats(
rtc::scoped_refptr<RtpReceiverInterface> selector,
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(callback);
RTC_DCHECK(stats_collector_);
rtc::scoped_refptr<RtpReceiverInternal> internal_receiver;
if (selector) {
for (const auto& proxy_transceiver : transceivers_.List()) {
for (const auto& proxy_receiver :
proxy_transceiver->internal()->receivers()) {
if (proxy_receiver == selector) {
internal_receiver = proxy_receiver->internal();
break;
}
}
if (internal_receiver)
break;
}
}
// If there is no |internal_receiver| then |selector| is either null or does
// not belong to the PeerConnection (in Plan B, receivers can be removed from
// the PeerConnection). This means that "all the stats objects representing
// the selector" is an empty set. Invoking GetStatsReport() with a null
// selector produces an empty stats report.
stats_collector_->GetStatsReport(internal_receiver, callback);
}
PeerConnectionInterface::SignalingState PeerConnection::signaling_state() {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_.signaling_state();
}
PeerConnectionInterface::IceConnectionState
PeerConnection::ice_connection_state() {
RTC_DCHECK_RUN_ON(signaling_thread());
return ice_connection_state_;
}
PeerConnectionInterface::IceConnectionState
PeerConnection::standardized_ice_connection_state() {
RTC_DCHECK_RUN_ON(signaling_thread());
return standardized_ice_connection_state_;
}
PeerConnectionInterface::PeerConnectionState
PeerConnection::peer_connection_state() {
RTC_DCHECK_RUN_ON(signaling_thread());
return connection_state_;
}
PeerConnectionInterface::IceGatheringState
PeerConnection::ice_gathering_state() {
RTC_DCHECK_RUN_ON(signaling_thread());
return ice_gathering_state_;
}
absl::optional<bool> PeerConnection::can_trickle_ice_candidates() {
RTC_DCHECK_RUN_ON(signaling_thread());
const SessionDescriptionInterface* description = current_remote_description();
if (!description) {
description = pending_remote_description();
}
if (!description) {
return absl::nullopt;
}
// TODO(bugs.webrtc.org/7443): Change to retrieve from session-level option.
if (description->description()->transport_infos().size() < 1) {
return absl::nullopt;
}
return description->description()->transport_infos()[0].description.HasOption(
"trickle");
}
rtc::scoped_refptr<DataChannelInterface> PeerConnection::CreateDataChannel(
const std::string& label,
const DataChannelInit* config) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::CreateDataChannel");
bool first_datachannel = !data_channel_controller_.HasDataChannels();
std::unique_ptr<InternalDataChannelInit> internal_config;
if (config) {
internal_config.reset(new InternalDataChannelInit(*config));
}
rtc::scoped_refptr<DataChannelInterface> channel(
data_channel_controller_.InternalCreateDataChannelWithProxy(
label, internal_config.get()));
if (!channel.get()) {
return nullptr;
}
// Trigger the onRenegotiationNeeded event for every new RTP DataChannel, or
// the first SCTP DataChannel.
if (data_channel_type() == cricket::DCT_RTP || first_datachannel) {
sdp_handler_.UpdateNegotiationNeeded();
}
NoteUsageEvent(UsageEvent::DATA_ADDED);
return channel;
}
void PeerConnection::RestartIce() {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_.RestartIce();
}
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_.CreateOffer(observer, options);
}
void PeerConnection::CreateAnswer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_.CreateAnswer(observer, options);
}
RTCError PeerConnection::HandleLegacyOfferOptions(
const RTCOfferAnswerOptions& options) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(IsUnifiedPlan());
if (options.offer_to_receive_audio == 0) {
RemoveRecvDirectionFromReceivingTransceiversOfType(
cricket::MEDIA_TYPE_AUDIO);
} else if (options.offer_to_receive_audio == 1) {
AddUpToOneReceivingTransceiverOfType(cricket::MEDIA_TYPE_AUDIO);
} else if (options.offer_to_receive_audio > 1) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
"offer_to_receive_audio > 1 is not supported.");
}
if (options.offer_to_receive_video == 0) {
RemoveRecvDirectionFromReceivingTransceiversOfType(
cricket::MEDIA_TYPE_VIDEO);
} else if (options.offer_to_receive_video == 1) {
AddUpToOneReceivingTransceiverOfType(cricket::MEDIA_TYPE_VIDEO);
} else if (options.offer_to_receive_video > 1) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
"offer_to_receive_video > 1 is not supported.");
}
return RTCError::OK();
}
void PeerConnection::RemoveRecvDirectionFromReceivingTransceiversOfType(
cricket::MediaType media_type) {
for (const auto& transceiver : GetReceivingTransceiversOfType(media_type)) {
RtpTransceiverDirection new_direction =
RtpTransceiverDirectionWithRecvSet(transceiver->direction(), false);
if (new_direction != transceiver->direction()) {
RTC_LOG(LS_INFO) << "Changing " << cricket::MediaTypeToString(media_type)
<< " transceiver (MID="
<< transceiver->mid().value_or("<not set>") << ") from "
<< RtpTransceiverDirectionToString(
transceiver->direction())
<< " to "
<< RtpTransceiverDirectionToString(new_direction)
<< " since CreateOffer specified offer_to_receive=0";
transceiver->internal()->set_direction(new_direction);
}
}
}
void PeerConnection::AddUpToOneReceivingTransceiverOfType(
cricket::MediaType media_type) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (GetReceivingTransceiversOfType(media_type).empty()) {
RTC_LOG(LS_INFO)
<< "Adding one recvonly " << cricket::MediaTypeToString(media_type)
<< " transceiver since CreateOffer specified offer_to_receive=1";
RtpTransceiverInit init;
init.direction = RtpTransceiverDirection::kRecvOnly;
AddTransceiver(media_type, nullptr, init,
/*update_negotiation_needed=*/false);
}
}
std::vector<rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
PeerConnection::GetReceivingTransceiversOfType(cricket::MediaType media_type) {
std::vector<
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
receiving_transceivers;
for (const auto& transceiver : transceivers_.List()) {
if (!transceiver->stopped() && transceiver->media_type() == media_type &&
RtpTransceiverDirectionHasRecv(transceiver->direction())) {
receiving_transceivers.push_back(transceiver);
}
}
return receiving_transceivers;
}
void PeerConnection::SetLocalDescription(
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc_ptr) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_.SetLocalDescription(observer, desc_ptr);
}
void PeerConnection::SetLocalDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_.SetLocalDescription(std::move(desc), observer);
}
void PeerConnection::SetLocalDescription(
SetSessionDescriptionObserver* observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_.SetLocalDescription(observer);
}
void PeerConnection::SetLocalDescription(
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_.SetLocalDescription(observer);
}
void PeerConnection::RemoveStoppedTransceivers() {
RTC_DCHECK_RUN_ON(signaling_thread());
// 3.2.10.1: For each transceiver in the connection's set of transceivers
// run the following steps:
if (!IsUnifiedPlan())
return;
// Traverse a copy of the transceiver list.
auto transceiver_list = transceivers_.List();
for (auto transceiver : transceiver_list) {
// 3.2.10.1.1: If transceiver is stopped, associated with an m= section
// and the associated m= section is rejected in
// connection.[[CurrentLocalDescription]] or
// connection.[[CurrentRemoteDescription]], remove the
// transceiver from the connection's set of transceivers.
if (!transceiver->stopped()) {
continue;
}
const ContentInfo* local_content =
sdp_handler_.FindMediaSectionForTransceiver(transceiver,
local_description());
const ContentInfo* remote_content =
sdp_handler_.FindMediaSectionForTransceiver(transceiver,
remote_description());
if ((local_content && local_content->rejected) ||
(remote_content && remote_content->rejected)) {
RTC_LOG(LS_INFO) << "Dissociating transceiver"
<< " since the media section is being recycled.";
transceiver->internal()->set_mid(absl::nullopt);
transceiver->internal()->set_mline_index(absl::nullopt);
transceivers_.Remove(transceiver);
continue;
}
if (!local_content && !remote_content) {
// TODO(bugs.webrtc.org/11973): Consider if this should be removed already
// See https://github.com/w3c/webrtc-pc/issues/2576
RTC_LOG(LS_INFO)
<< "Dropping stopped transceiver that was never associated";
transceivers_.Remove(transceiver);
continue;
}
}
}
void PeerConnection::SetRemoteDescription(
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc_ptr) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_.SetRemoteDescription(observer, desc_ptr);
}
void PeerConnection::SetRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_.SetRemoteDescription(std::move(desc), observer);
}
void PeerConnection::ProcessRemovalOfRemoteTrack(
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
transceiver,
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>* remove_list,
std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams) {
RTC_DCHECK(transceiver->mid());
RTC_LOG(LS_INFO) << "Processing the removal of a track for MID="
<< *transceiver->mid();
std::vector<rtc::scoped_refptr<MediaStreamInterface>> previous_streams =
transceiver->internal()->receiver_internal()->streams();
// This will remove the remote track from the streams.
transceiver->internal()->receiver_internal()->set_stream_ids({});
remove_list->push_back(transceiver);
RemoveRemoteStreamsIfEmpty(previous_streams, removed_streams);
}
void PeerConnection::RemoveRemoteStreamsIfEmpty(
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& remote_streams,
std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams) {
RTC_DCHECK_RUN_ON(signaling_thread());
// TODO(https://crbug.com/webrtc/9480): When we use stream IDs instead of
// streams, see if the stream was removed by checking if this was the last
// receiver with that stream ID.
for (const auto& remote_stream : remote_streams) {
if (remote_stream->GetAudioTracks().empty() &&
remote_stream->GetVideoTracks().empty()) {
remote_streams_->RemoveStream(remote_stream);
removed_streams->push_back(remote_stream);
}
}
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::GetAssociatedTransceiver(const std::string& mid) const {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(IsUnifiedPlan());
return transceivers_.FindByMid(mid);
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::GetTransceiverByMLineIndex(size_t mline_index) const {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(IsUnifiedPlan());
return transceivers_.FindByMLineIndex(mline_index);
}
PeerConnectionInterface::RTCConfiguration PeerConnection::GetConfiguration() {
RTC_DCHECK_RUN_ON(signaling_thread());
return configuration_;
}
RTCError PeerConnection::SetConfiguration(
const RTCConfiguration& configuration) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
if (IsClosed()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
"SetConfiguration: PeerConnection is closed.");
}
// 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.
if (local_description() && configuration.ice_candidate_pool_size !=
configuration_.ice_candidate_pool_size) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
"Can't change candidate pool size after calling "
"SetLocalDescription.");
}
if (local_description() &&
configuration.crypto_options != configuration_.crypto_options) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
"Can't change crypto_options after calling "
"SetLocalDescription.");
}
// 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.turn_port_prune_policy = configuration.turn_port_prune_policy;
modified_config.surface_ice_candidates_on_ice_transport_type_changed =
configuration.surface_ice_candidates_on_ice_transport_type_changed;
modified_config.ice_check_min_interval = configuration.ice_check_min_interval;
modified_config.ice_check_interval_strong_connectivity =
configuration.ice_check_interval_strong_connectivity;
modified_config.ice_check_interval_weak_connectivity =
configuration.ice_check_interval_weak_connectivity;
modified_config.ice_unwritable_timeout = configuration.ice_unwritable_timeout;
modified_config.ice_unwritable_min_checks =
configuration.ice_unwritable_min_checks;
modified_config.ice_inactive_timeout = configuration.ice_inactive_timeout;
modified_config.stun_candidate_keepalive_interval =
configuration.stun_candidate_keepalive_interval;
modified_config.turn_customizer = configuration.turn_customizer;
modified_config.network_preference = configuration.network_preference;
modified_config.active_reset_srtp_params =
configuration.active_reset_srtp_params;
modified_config.turn_logging_id = configuration.turn_logging_id;
modified_config.allow_codec_switching = configuration.allow_codec_switching;
if (configuration != modified_config) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
"Modifying the configuration in an unsupported way.");
}
// Validate the modified configuration.
RTCError validate_error = ValidateConfiguration(modified_config);
if (!validate_error.ok()) {
return validate_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 > static_cast<int>(UINT16_MAX)) {
return RTCError(RTCErrorType::INVALID_RANGE);
}
// 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 RTCError(parse_error);
}
// Add the turn logging id to all turn servers
for (cricket::RelayServerConfig& turn_server : turn_servers) {
turn_server.turn_logging_id = configuration.turn_logging_id;
}
// Note if STUN or TURN servers were supplied.
if (!stun_servers.empty()) {
NoteUsageEvent(UsageEvent::STUN_SERVER_ADDED);
}
if (!turn_servers.empty()) {
NoteUsageEvent(UsageEvent::TURN_SERVER_ADDED);
}
// 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.GetTurnPortPrunePolicy(),
modified_config.turn_customizer,
modified_config.stun_candidate_keepalive_interval,
static_cast<bool>(local_description())))) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to apply configuration to PortAllocator.");
}
// 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 ||
NeedIceRestart(
configuration_.surface_ice_candidates_on_ice_transport_type_changed,
configuration_.type, modified_config.type) ||
modified_config.GetTurnPortPrunePolicy() !=
configuration_.GetTurnPortPrunePolicy()) {
transport_controller_->SetNeedsIceRestartFlag();
}
transport_controller_->SetIceConfig(ParseIceConfig(modified_config));
if (configuration_.active_reset_srtp_params !=
modified_config.active_reset_srtp_params) {
transport_controller_->SetActiveResetSrtpParams(
modified_config.active_reset_srtp_params);
}
if (modified_config.allow_codec_switching.has_value()) {
std::vector<cricket::VideoMediaChannel*> channels;
for (const auto& transceiver : transceivers_.List()) {
if (transceiver->media_type() != cricket::MEDIA_TYPE_VIDEO)
continue;
auto* video_channel = static_cast<cricket::VideoChannel*>(
transceiver->internal()->channel());
if (video_channel)
channels.push_back(video_channel->media_channel());
}
worker_thread()->Invoke<void>(
RTC_FROM_HERE,
[channels = std::move(channels),
allow_codec_switching = *modified_config.allow_codec_switching]() {
for (auto* ch : channels)
ch->SetVideoCodecSwitchingEnabled(allow_codec_switching);
});
}
configuration_ = modified_config;
return RTCError::OK();
}
bool PeerConnection::AddIceCandidate(
const IceCandidateInterface* ice_candidate) {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_.AddIceCandidate(ice_candidate);
}
void PeerConnection::AddIceCandidate(
std::unique_ptr<IceCandidateInterface> candidate,
std::function<void(RTCError)> callback) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_.AddIceCandidate(std::move(candidate), callback);
}
bool PeerConnection::RemoveIceCandidates(
const std::vector<cricket::Candidate>& candidates) {
TRACE_EVENT0("webrtc", "PeerConnection::RemoveIceCandidates");
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_.RemoveIceCandidates(candidates);
}
RTCError PeerConnection::SetBitrate(const BitrateSettings& bitrate) {
if (!worker_thread()->IsCurrent()) {
return worker_thread()->Invoke<RTCError>(
RTC_FROM_HERE, [&]() { return SetBitrate(bitrate); });
}
RTC_DCHECK_RUN_ON(worker_thread());
const bool has_min = bitrate.min_bitrate_bps.has_value();
const bool has_start = bitrate.start_bitrate_bps.has_value();
const bool has_max = bitrate.max_bitrate_bps.has_value();
if (has_min && *bitrate.min_bitrate_bps < 0) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"min_bitrate_bps <= 0");
}
if (has_start) {
if (has_min && *bitrate.start_bitrate_bps < *bitrate.min_bitrate_bps) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"start_bitrate_bps < min_bitrate_bps");
} else if (*bitrate.start_bitrate_bps < 0) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"curent_bitrate_bps < 0");
}
}
if (has_max) {
if (has_start && *bitrate.max_bitrate_bps < *bitrate.start_bitrate_bps) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"max_bitrate_bps < start_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");
}
}
RTC_DCHECK(call_.get());
call_->SetClientBitratePreferences(bitrate);
return RTCError::OK();
}
void PeerConnection::SetAudioPlayout(bool playout) {
if (!worker_thread()->IsCurrent()) {
worker_thread()->Invoke<void>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnection::SetAudioPlayout, this, playout));
return;
}
auto audio_state =
factory_->channel_manager()->media_engine()->voice().GetAudioState();
audio_state->SetPlayout(playout);
}
void PeerConnection::SetAudioRecording(bool recording) {
if (!worker_thread()->IsCurrent()) {
worker_thread()->Invoke<void>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnection::SetAudioRecording, this, recording));
return;
}
auto audio_state =
factory_->channel_manager()->media_engine()->voice().GetAudioState();
audio_state->SetRecording(recording);
}
std::unique_ptr<rtc::SSLCertificate>
PeerConnection::GetRemoteAudioSSLCertificate() {
std::unique_ptr<rtc::SSLCertChain> chain = GetRemoteAudioSSLCertChain();
if (!chain || !chain->GetSize()) {
return nullptr;
}
return chain->Get(0).Clone();
}
std::unique_ptr<rtc::SSLCertChain>
PeerConnection::GetRemoteAudioSSLCertChain() {
RTC_DCHECK_RUN_ON(signaling_thread());
auto audio_transceiver = GetFirstAudioTransceiver();
if (!audio_transceiver || !audio_transceiver->internal()->channel()) {
return nullptr;
}
return transport_controller_->GetRemoteSSLCertChain(
audio_transceiver->internal()->channel()->transport_name());
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::GetFirstAudioTransceiver() const {
for (auto transceiver : transceivers_.List()) {
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
return transceiver;
}
}
return nullptr;
}
void PeerConnection::AddAdaptationResource(
rtc::scoped_refptr<Resource> resource) {
if (!worker_thread()->IsCurrent()) {
return worker_thread()->Invoke<void>(RTC_FROM_HERE, [this, resource]() {
return AddAdaptationResource(resource);
});
}
RTC_DCHECK_RUN_ON(worker_thread());
if (!call_) {
// The PeerConnection has been closed.
return;
}
call_->AddAdaptationResource(resource);
}
bool PeerConnection::StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms) {
return worker_thread()->Invoke<bool>(
RTC_FROM_HERE,
[this, output = std::move(output), output_period_ms]() mutable {
return StartRtcEventLog_w(std::move(output), output_period_ms);
});
}
bool PeerConnection::StartRtcEventLog(
std::unique_ptr<RtcEventLogOutput> output) {
int64_t output_period_ms = webrtc::RtcEventLog::kImmediateOutput;
if (absl::StartsWith(factory_->trials().Lookup("WebRTC-RtcEventLogNewFormat"),
"Enabled")) {
output_period_ms = 5000;
}
return StartRtcEventLog(std::move(output), output_period_ms);
}
void PeerConnection::StopRtcEventLog() {
worker_thread()->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&PeerConnection::StopRtcEventLog_w, this));
}
rtc::scoped_refptr<DtlsTransportInterface>
PeerConnection::LookupDtlsTransportByMid(const std::string& mid) {
RTC_DCHECK_RUN_ON(signaling_thread());
return transport_controller_->LookupDtlsTransportByMid(mid);
}
rtc::scoped_refptr<DtlsTransport>
PeerConnection::LookupDtlsTransportByMidInternal(const std::string& mid) {
RTC_DCHECK_RUN_ON(signaling_thread());
return transport_controller_->LookupDtlsTransportByMid(mid);
}
rtc::scoped_refptr<SctpTransportInterface> PeerConnection::GetSctpTransport()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!sctp_mid_s_) {
return nullptr;
}
return transport_controller_->GetSctpTransport(*sctp_mid_s_);
}
const SessionDescriptionInterface* PeerConnection::local_description() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_.local_description();
}
const SessionDescriptionInterface* PeerConnection::remote_description() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_.remote_description();
}
const SessionDescriptionInterface* PeerConnection::current_local_description()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_.current_local_description();
}
const SessionDescriptionInterface* PeerConnection::current_remote_description()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_.current_remote_description();
}
const SessionDescriptionInterface* PeerConnection::pending_local_description()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_.pending_local_description();
}
const SessionDescriptionInterface* PeerConnection::pending_remote_description()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_.pending_remote_description();
}
void PeerConnection::Close() {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::Close");
if (IsClosed()) {
return;
}
// Update stats here so that we have the most recent stats for tracks and
// streams before the channels are closed.
stats_->UpdateStats(kStatsOutputLevelStandard);
ice_connection_state_ = PeerConnectionInterface::kIceConnectionClosed;
Observer()->OnIceConnectionChange(ice_connection_state_);
standardized_ice_connection_state_ =
PeerConnectionInterface::IceConnectionState::kIceConnectionClosed;
connection_state_ = PeerConnectionInterface::PeerConnectionState::kClosed;
Observer()->OnConnectionChange(connection_state_);
sdp_handler_.Close();
NoteUsageEvent(UsageEvent::CLOSE_CALLED);
for (const auto& transceiver : transceivers_.List()) {
transceiver->internal()->SetPeerConnectionClosed();
if (!transceiver->stopped())
transceiver->StopInternal();
}
// Ensure that all asynchronous stats requests are completed before destroying
// the transport controller below.
if (stats_collector_) {
stats_collector_->WaitForPendingRequest();
}
// Don't destroy BaseChannels until after stats has been cleaned up so that
// the last stats request can still read from the channels.
DestroyAllChannels();
// The event log is used in the transport controller, which must be outlived
// by the former. CreateOffer by the peer connection is implemented
// asynchronously and if the peer connection is closed without resetting the
// WebRTC session description factory, the session description factory would
// call the transport controller.
sdp_handler_.ResetSessionDescFactory();
transport_controller_.reset();
network_thread()->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
port_allocator_.get()));
worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
RTC_DCHECK_RUN_ON(worker_thread());
call_safety_.reset();
call_.reset();
// The event log must outlive call (and any other object that uses it).
event_log_.reset();
});
ReportUsagePattern();
// The .h file says that observer can be discarded after close() returns.
// Make sure this is true.
observer_ = nullptr;
}
void PeerConnection::OnMessage(rtc::Message* msg) {
RTC_DCHECK_RUN_ON(signaling_thread());
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(std::move(param->error));
delete param;
break;
}
case MSG_CREATE_SESSIONDESCRIPTION_FAILED: {
CreateSessionDescriptionMsg* param =
static_cast<CreateSessionDescriptionMsg*>(msg->pdata);
param->observer->OnFailure(std::move(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_REPORT_USAGE_PATTERN: {
ReportUsagePattern();
break;
}
default:
RTC_NOTREACHED() << "Not implemented";
break;
}
}
cricket::VoiceMediaChannel* PeerConnection::voice_media_channel() const {
RTC_DCHECK(!IsUnifiedPlan());
auto* voice_channel = static_cast<cricket::VoiceChannel*>(
GetAudioTransceiver()->internal()->channel());
if (voice_channel) {
return voice_channel->media_channel();
} else {
return nullptr;
}
}
cricket::VideoMediaChannel* PeerConnection::video_media_channel() const {
RTC_DCHECK(!IsUnifiedPlan());
auto* video_channel = static_cast<cricket::VideoChannel*>(
GetVideoTransceiver()->internal()->channel());
if (video_channel) {
return video_channel->media_channel();
} else {
return nullptr;
}
}
void PeerConnection::CreateAudioReceiver(
MediaStreamInterface* stream,
const RtpSenderInfo& remote_sender_info) {
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams;
streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
// TODO(https://crbug.com/webrtc/9480): When we remove remote_streams(), use
// the constructor taking stream IDs instead.
auto* audio_receiver = new AudioRtpReceiver(
worker_thread(), remote_sender_info.sender_id, streams);
audio_receiver->SetMediaChannel(voice_media_channel());
if (remote_sender_info.sender_id == kDefaultAudioSenderId) {
audio_receiver->SetupUnsignaledMediaChannel();
} else {
audio_receiver->SetupMediaChannel(remote_sender_info.first_ssrc);
}
auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(), audio_receiver);
GetAudioTransceiver()->internal()->AddReceiver(receiver);
Observer()->OnAddTrack(receiver, streams);
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
}
void PeerConnection::CreateVideoReceiver(
MediaStreamInterface* stream,
const RtpSenderInfo& remote_sender_info) {
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams;
streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
// TODO(https://crbug.com/webrtc/9480): When we remove remote_streams(), use
// the constructor taking stream IDs instead.
auto* video_receiver = new VideoRtpReceiver(
worker_thread(), remote_sender_info.sender_id, streams);
video_receiver->SetMediaChannel(video_media_channel());
if (remote_sender_info.sender_id == kDefaultVideoSenderId) {
video_receiver->SetupUnsignaledMediaChannel();
} else {
video_receiver->SetupMediaChannel(remote_sender_info.first_ssrc);
}
auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(), video_receiver);
GetVideoTransceiver()->internal()->AddReceiver(receiver);
Observer()->OnAddTrack(receiver, streams);
NoteUsageEvent(UsageEvent::VIDEO_ADDED);
}
// TODO(deadbeef): Keep RtpReceivers around even if track goes away in remote
// description.
rtc::scoped_refptr<RtpReceiverInterface> PeerConnection::RemoveAndStopReceiver(
const RtpSenderInfo& remote_sender_info) {
auto receiver = FindReceiverById(remote_sender_info.sender_id);
if (!receiver) {
RTC_LOG(LS_WARNING) << "RtpReceiver for track with id "
<< remote_sender_info.sender_id << " doesn't exist.";
return nullptr;
}
if (receiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
GetAudioTransceiver()->internal()->RemoveReceiver(receiver);
} else {
GetVideoTransceiver()->internal()->RemoveReceiver(receiver);
}
return receiver;
}
void PeerConnection::AddAudioTrack(AudioTrackInterface* track,
MediaStreamInterface* stream) {
RTC_DCHECK(!IsClosed());
RTC_DCHECK(track);
RTC_DCHECK(stream);
auto sender = FindSenderForTrack(track);
if (sender) {
// 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_ids({stream->id()});
return;
}
// Normal case; we've never seen this track before.
auto new_sender = CreateSender(cricket::MEDIA_TYPE_AUDIO, track->id(), track,
{stream->id()}, {});
new_sender->internal()->SetMediaChannel(voice_media_channel());
GetAudioTransceiver()->internal()->AddSender(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 RtpSenderInfo* sender_info =
FindSenderInfo(local_audio_sender_infos_, stream->id(), track->id());
if (sender_info) {
new_sender->internal()->SetSsrc(sender_info->first_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) {
RTC_LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
<< " doesn't exist.";
return;
}
GetAudioTransceiver()->internal()->RemoveSender(sender);
}
void PeerConnection::AddVideoTrack(VideoTrackInterface* track,
MediaStreamInterface* stream) {
RTC_DCHECK(!IsClosed());
RTC_DCHECK(track);
RTC_DCHECK(stream);
auto sender = FindSenderForTrack(track);
if (sender) {
// 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_ids({stream->id()});
return;
}
// Normal case; we've never seen this track before.
auto new_sender = CreateSender(cricket::MEDIA_TYPE_VIDEO, track->id(), track,
{stream->id()}, {});
new_sender->internal()->SetMediaChannel(video_media_channel());
GetVideoTransceiver()->internal()->AddSender(new_sender);
const RtpSenderInfo* sender_info =
FindSenderInfo(local_video_sender_infos_, stream->id(), track->id());
if (sender_info) {
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
}
}
void PeerConnection::RemoveVideoTrack(VideoTrackInterface* track,
MediaStreamInterface* stream) {
RTC_DCHECK(!IsClosed());
auto sender = FindSenderForTrack(track);
if (!sender) {
RTC_LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
<< " doesn't exist.";
return;
}
GetVideoTransceiver()->internal()->RemoveSender(sender);
}
void PeerConnection::SetIceConnectionState(IceConnectionState new_state) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (ice_connection_state_ == new_state) {
return;
}
// After transitioning to "closed", ignore any additional states from
// TransportController (such as "disconnected").
if (IsClosed()) {
return;
}
RTC_LOG(LS_INFO) << "Changing IceConnectionState " << ice_connection_state_
<< " => " << new_state;
RTC_DCHECK(ice_connection_state_ !=
PeerConnectionInterface::kIceConnectionClosed);
ice_connection_state_ = new_state;
Observer()->OnIceConnectionChange(ice_connection_state_);
}
void PeerConnection::SetStandardizedIceConnectionState(
PeerConnectionInterface::IceConnectionState new_state) {
if (standardized_ice_connection_state_ == new_state) {
return;
}
if (IsClosed()) {
return;
}
RTC_LOG(LS_INFO) << "Changing standardized IceConnectionState "
<< standardized_ice_connection_state_ << " => " << new_state;
standardized_ice_connection_state_ = new_state;
Observer()->OnStandardizedIceConnectionChange(new_state);
}
void PeerConnection::SetConnectionState(
PeerConnectionInterface::PeerConnectionState new_state) {
if (connection_state_ == new_state)
return;
if (IsClosed())
return;
connection_state_ = new_state;
Observer()->OnConnectionChange(new_state);
}
void PeerConnection::OnIceGatheringChange(
PeerConnectionInterface::IceGatheringState new_state) {
if (IsClosed()) {
return;
}
ice_gathering_state_ = new_state;
Observer()->OnIceGatheringChange(ice_gathering_state_);
}
void PeerConnection::OnIceCandidate(
std::unique_ptr<IceCandidateInterface> candidate) {
if (IsClosed()) {
return;
}
ReportIceCandidateCollected(candidate->candidate());
Observer()->OnIceCandidate(candidate.get());
}
void PeerConnection::OnIceCandidateError(const std::string& address,
int port,
const std::string& url,
int error_code,
const std::string& error_text) {
if (IsClosed()) {
return;
}
Observer()->OnIceCandidateError(address, port, url, error_code, error_text);
// Leftover not to break wpt test during migration to the new API.
Observer()->OnIceCandidateError(address + ":", url, error_code, error_text);
}
void PeerConnection::OnIceCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates) {
if (IsClosed()) {
return;
}
Observer()->OnIceCandidatesRemoved(candidates);
}
void PeerConnection::OnSelectedCandidatePairChanged(
const cricket::CandidatePairChangeEvent& event) {
if (IsClosed()) {
return;
}
if (event.selected_candidate_pair.local_candidate().type() ==
LOCAL_PORT_TYPE &&
event.selected_candidate_pair.remote_candidate().type() ==
LOCAL_PORT_TYPE) {
NoteUsageEvent(UsageEvent::DIRECT_CONNECTION_SELECTED);
}
Observer()->OnIceSelectedCandidatePairChanged(event);
}
void PeerConnection::OnAudioTrackAdded(AudioTrackInterface* track,
MediaStreamInterface* stream) {
if (IsClosed()) {
return;
}
AddAudioTrack(track, stream);
sdp_handler_.UpdateNegotiationNeeded();
}
void PeerConnection::OnAudioTrackRemoved(AudioTrackInterface* track,
MediaStreamInterface* stream) {
if (IsClosed()) {
return;
}
RemoveAudioTrack(track, stream);
sdp_handler_.UpdateNegotiationNeeded();
}
void PeerConnection::OnVideoTrackAdded(VideoTrackInterface* track,
MediaStreamInterface* stream) {
if (IsClosed()) {
return;
}
AddVideoTrack(track, stream);
sdp_handler_.UpdateNegotiationNeeded();
}
void PeerConnection::OnVideoTrackRemoved(VideoTrackInterface* track,
MediaStreamInterface* stream) {
if (IsClosed()) {
return;
}
RemoveVideoTrack(track, stream);
sdp_handler_.UpdateNegotiationNeeded();
}
void PeerConnection::PostSetSessionDescriptionSuccess(
SetSessionDescriptionObserver* observer) {
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
signaling_thread()->Post(RTC_FROM_HERE, this,
MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
}
void PeerConnection::PostSetSessionDescriptionFailure(
SetSessionDescriptionObserver* observer,
RTCError&& error) {
RTC_DCHECK(!error.ok());
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
msg->error = std::move(error);
signaling_thread()->Post(RTC_FROM_HERE, this,
MSG_SET_SESSIONDESCRIPTION_FAILED, msg);
}
void PeerConnection::PostCreateSessionDescriptionFailure(
CreateSessionDescriptionObserver* observer,
RTCError error) {
RTC_DCHECK(!error.ok());
CreateSessionDescriptionMsg* msg = new CreateSessionDescriptionMsg(observer);
msg->error = std::move(error);
signaling_thread()->Post(RTC_FROM_HERE, this,
MSG_CREATE_SESSIONDESCRIPTION_FAILED, msg);
}
void PeerConnection::GenerateMediaDescriptionOptions(
const SessionDescriptionInterface* session_desc,
RtpTransceiverDirection audio_direction,
RtpTransceiverDirection video_direction,
absl::optional<size_t>* audio_index,
absl::optional<size_t>* video_index,
absl::optional<size_t>* data_index,
cricket::MediaSessionOptions* session_options) {
RTC_DCHECK_RUN_ON(signaling_thread());
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,
RtpTransceiverDirection::kInactive, /*stopped=*/true));
} else {
bool stopped = (audio_direction == RtpTransceiverDirection::kInactive);
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO,
content.name, audio_direction,
stopped));
*audio_index = session_options->media_description_options.size() - 1;
}
session_options->media_description_options.back().header_extensions =
channel_manager()->GetSupportedAudioRtpHeaderExtensions();
} 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,
RtpTransceiverDirection::kInactive, /*stopped=*/true));
} else {
bool stopped = (video_direction == RtpTransceiverDirection::kInactive);
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO,
content.name, video_direction,
stopped));
*video_index = session_options->media_description_options.size() - 1;
}
session_options->media_description_options.back().header_extensions =
channel_manager()->GetSupportedVideoRtpHeaderExtensions();
} 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(
GetMediaDescriptionOptionsForRejectedData(content.name));
} else {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForActiveData(content.name));
*data_index = session_options->media_description_options.size() - 1;
}
}
}
}
cricket::MediaDescriptionOptions
PeerConnection::GetMediaDescriptionOptionsForActiveData(
const std::string& mid) const {
RTC_DCHECK_RUN_ON(signaling_thread());
// Direction for data sections is meaningless, but legacy endpoints might
// expect sendrecv.
cricket::MediaDescriptionOptions options(cricket::MEDIA_TYPE_DATA, mid,
RtpTransceiverDirection::kSendRecv,
/*stopped=*/false);
AddRtpDataChannelOptions(*data_channel_controller_.rtp_data_channels(),
&options);
return options;
}
cricket::MediaDescriptionOptions
PeerConnection::GetMediaDescriptionOptionsForRejectedData(
const std::string& mid) const {
RTC_DCHECK_RUN_ON(signaling_thread());
cricket::MediaDescriptionOptions options(cricket::MEDIA_TYPE_DATA, mid,
RtpTransceiverDirection::kInactive,
/*stopped=*/true);
AddRtpDataChannelOptions(*data_channel_controller_.rtp_data_channels(),
&options);
return options;
}
absl::optional<std::string> PeerConnection::GetDataMid() const {
RTC_DCHECK_RUN_ON(signaling_thread());
switch (data_channel_type()) {
case cricket::DCT_RTP:
if (!data_channel_controller_.rtp_data_channel()) {
return absl::nullopt;
}
return data_channel_controller_.rtp_data_channel()->content_name();
case cricket::DCT_SCTP:
return sctp_mid_s_;
default:
return absl::nullopt;
}
}
void PeerConnection::RemoveSenders(cricket::MediaType media_type) {
RTC_DCHECK_RUN_ON(signaling_thread());
UpdateLocalSenders(std::vector<cricket::StreamParams>(), media_type);
UpdateRemoteSendersList(std::vector<cricket::StreamParams>(), false,
media_type, nullptr);
}
void PeerConnection::UpdateRemoteSendersList(
const cricket::StreamParamsVec& streams,
bool default_sender_needed,
cricket::MediaType media_type,
StreamCollection* new_streams) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(!IsUnifiedPlan());
std::vector<RtpSenderInfo>* current_senders =
GetRemoteSenderInfos(media_type);
// Find removed senders. I.e., senders where the sender id or ssrc don't match
// the new StreamParam.
for (auto sender_it = current_senders->begin();
sender_it != current_senders->end();
/* incremented manually */) {
const RtpSenderInfo& info = *sender_it;
const cricket::StreamParams* params =
cricket::GetStreamBySsrc(streams, info.first_ssrc);
std::string params_stream_id;
if (params) {
params_stream_id =
(!params->first_stream_id().empty() ? params->first_stream_id()
: kDefaultStreamId);
}
bool sender_exists = params && params->id == info.sender_id &&
params_stream_id == info.stream_id;
// If this is a default track, and we still need it, don't remove it.
if ((info.stream_id == kDefaultStreamId && default_sender_needed) ||
sender_exists) {
++sender_it;
} else {
OnRemoteSenderRemoved(info, media_type);
sender_it = current_senders->erase(sender_it);
}
}
// Find new and active senders.
for (const cricket::StreamParams& params : streams) {
if (!params.has_ssrcs()) {
// The remote endpoint has streams, but didn't signal ssrcs. For an active
// sender, this means it is coming from a Unified Plan endpoint,so we just
// create a default.
default_sender_needed = true;
break;
}
// |params.id| is the sender id and the stream id uses the first of
// |params.stream_ids|. The remote description could come from a Unified
// Plan endpoint, with multiple or no stream_ids() signaled. Since this is
// not supported in Plan B, we just take the first here and create the
// default stream ID if none is specified.
const std::string& stream_id =
(!params.first_stream_id().empty() ? params.first_stream_id()
: kDefaultStreamId);
const std::string& sender_id = params.id;
uint32_t ssrc = params.first_ssrc();
rtc::scoped_refptr<MediaStreamInterface> stream =
remote_streams_->find(stream_id);
if (!stream) {
// This is a new MediaStream. Create a new remote MediaStream.
stream = MediaStreamProxy::Create(rtc::Thread::Current(),
MediaStream::Create(stream_id));
remote_streams_->AddStream(stream);
new_streams->AddStream(stream);
}
const RtpSenderInfo* sender_info =
FindSenderInfo(*current_senders, stream_id, sender_id);
if (!sender_info) {
current_senders->push_back(RtpSenderInfo(stream_id, sender_id, ssrc));
OnRemoteSenderAdded(current_senders->back(), media_type);
}
}
// Add default sender if necessary.
if (default_sender_needed) {
rtc::scoped_refptr<MediaStreamInterface> default_stream =
remote_streams_->find(kDefaultStreamId);
if (!default_stream) {
// Create the new default MediaStream.
default_stream = MediaStreamProxy::Create(
rtc::Thread::Current(), MediaStream::Create(kDefaultStreamId));
remote_streams_->AddStream(default_stream);
new_streams->AddStream(default_stream);
}
std::string default_sender_id = (media_type == cricket::MEDIA_TYPE_AUDIO)
? kDefaultAudioSenderId
: kDefaultVideoSenderId;
const RtpSenderInfo* default_sender_info =
FindSenderInfo(*current_senders, kDefaultStreamId, default_sender_id);
if (!default_sender_info) {
current_senders->push_back(
RtpSenderInfo(kDefaultStreamId, default_sender_id, /*ssrc=*/0));
OnRemoteSenderAdded(current_senders->back(), media_type);
}
}
}
void PeerConnection::OnRemoteSenderAdded(const RtpSenderInfo& sender_info,
cricket::MediaType media_type) {
RTC_LOG(LS_INFO) << "Creating " << cricket::MediaTypeToString(media_type)
<< " receiver for track_id=" << sender_info.sender_id
<< " and stream_id=" << sender_info.stream_id;
MediaStreamInterface* stream = remote_streams_->find(sender_info.stream_id);
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
CreateAudioReceiver(stream, sender_info);
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
CreateVideoReceiver(stream, sender_info);
} else {
RTC_NOTREACHED() << "Invalid media type";
}
}
void PeerConnection::OnRemoteSenderRemoved(const RtpSenderInfo& sender_info,
cricket::MediaType media_type) {
RTC_LOG(LS_INFO) << "Removing " << cricket::MediaTypeToString(media_type)
<< " receiver for track_id=" << sender_info.sender_id
<< " and stream_id=" << sender_info.stream_id;
MediaStreamInterface* stream = remote_streams_->find(sender_info.stream_id);
rtc::scoped_refptr<RtpReceiverInterface> receiver;
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().
receiver = RemoveAndStopReceiver(sender_info);
rtc::scoped_refptr<AudioTrackInterface> audio_track =
stream->FindAudioTrack(sender_info.sender_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().
receiver = RemoveAndStopReceiver(sender_info);
rtc::scoped_refptr<VideoTrackInterface> video_track =
stream->FindVideoTrack(sender_info.sender_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";
}
if (receiver) {
Observer()->OnRemoveTrack(receiver);
}
}
void PeerConnection::UpdateEndedRemoteMediaStreams() {
RTC_DCHECK_RUN_ON(signaling_thread());
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::UpdateLocalSenders(
const std::vector<cricket::StreamParams>& streams,
cricket::MediaType media_type) {
RTC_DCHECK_RUN_ON(signaling_thread());
std::vector<RtpSenderInfo>* current_senders = GetLocalSenderInfos(media_type);
// Find removed tracks. I.e., tracks where the track id, stream id or ssrc
// don't match the new StreamParam.
for (auto sender_it = current_senders->begin();
sender_it != current_senders->end();
/* incremented manually */) {
const RtpSenderInfo& info = *sender_it;
const cricket::StreamParams* params =
cricket::GetStreamBySsrc(streams, info.first_ssrc);
if (!params || params->id != info.sender_id ||
params->first_stream_id() != info.stream_id) {
OnLocalSenderRemoved(info, media_type);
sender_it = current_senders->erase(sender_it);
} else {
++sender_it;
}
}
// Find new and active senders.
for (const cricket::StreamParams& params : streams) {
// The sync_label is the MediaStream label and the |stream.id| is the
// sender id.
const std::string& stream_id = params.first_stream_id();
const std::string& sender_id = params.id;
uint32_t ssrc = params.first_ssrc();
const RtpSenderInfo* sender_info =
FindSenderInfo(*current_senders, stream_id, sender_id);
if (!sender_info) {
current_senders->push_back(RtpSenderInfo(stream_id, sender_id, ssrc));
OnLocalSenderAdded(current_senders->back(), media_type);
}
}
}
void PeerConnection::OnLocalSenderAdded(const RtpSenderInfo& sender_info,
cricket::MediaType media_type) {
RTC_DCHECK(!IsUnifiedPlan());
auto sender = FindSenderById(sender_info.sender_id);
if (!sender) {
RTC_LOG(LS_WARNING) << "An unknown RtpSender with id "
<< sender_info.sender_id
<< " has been configured in the local description.";
return;
}
if (sender->media_type() != media_type) {
RTC_LOG(LS_WARNING) << "An RtpSender has been configured in the local"
" description with an unexpected media type.";
return;
}
sender->internal()->set_stream_ids({sender_info.stream_id});
sender->internal()->SetSsrc(sender_info.first_ssrc);
}
void PeerConnection::OnLocalSenderRemoved(const RtpSenderInfo& sender_info,
cricket::MediaType media_type) {
auto sender = FindSenderById(sender_info.sender_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) {
RTC_LOG(LS_WARNING) << "An RtpSender has been configured in the local"
" description with an unexpected media type.";
return;
}
sender->internal()->SetSsrc(0);
}
void PeerConnection::OnSctpDataChannelClosed(DataChannelInterface* channel) {
// Since data_channel_controller doesn't do signals, this
// signal is relayed here.
data_channel_controller_.OnSctpDataChannelClosed(
static_cast<SctpDataChannel*>(channel));
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::GetAudioTransceiver() const {
RTC_DCHECK_RUN_ON(signaling_thread());
// This method only works with Plan B SDP, where there is a single
// audio/video transceiver.
RTC_DCHECK(!IsUnifiedPlan());
for (auto transceiver : transceivers_.List()) {
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
return transceiver;
}
}
RTC_NOTREACHED();
return nullptr;
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::GetVideoTransceiver() const {
RTC_DCHECK_RUN_ON(signaling_thread());
// This method only works with Plan B SDP, where there is a single
// audio/video transceiver.
RTC_DCHECK(!IsUnifiedPlan());
for (auto transceiver : transceivers_.List()) {
if (transceiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
return transceiver;
}
}
RTC_NOTREACHED();
return nullptr;
}
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) const {
for (const auto& transceiver : transceivers_.List()) {
for (auto sender : transceiver->internal()->senders()) {
if (sender->track() == track) {
return sender;
}
}
}
return nullptr;
}
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
PeerConnection::FindSenderById(const std::string& sender_id) const {
for (const auto& transceiver : transceivers_.List()) {
for (auto sender : transceiver->internal()->senders()) {
if (sender->id() == sender_id) {
return sender;
}
}
}
return nullptr;
}
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
PeerConnection::FindReceiverById(const std::string& receiver_id) const {
for (const auto& transceiver : transceivers_.List()) {
for (auto receiver : transceiver->internal()->receivers()) {
if (receiver->id() == receiver_id) {
return receiver;
}
}
}
return nullptr;
}
std::vector<PeerConnection::RtpSenderInfo>*
PeerConnection::GetRemoteSenderInfos(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_sender_infos_
: &remote_video_sender_infos_;
}
std::vector<PeerConnection::RtpSenderInfo>* PeerConnection::GetLocalSenderInfos(
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_sender_infos_
: &local_video_sender_infos_;
}
const PeerConnection::RtpSenderInfo* PeerConnection::FindSenderInfo(
const std::vector<PeerConnection::RtpSenderInfo>& infos,
const std::string& stream_id,
const std::string sender_id) const {
for (const RtpSenderInfo& sender_info : infos) {
if (sender_info.stream_id == stream_id &&
sender_info.sender_id == sender_id) {
return &sender_info;
}
}
return nullptr;
}
SctpDataChannel* PeerConnection::FindDataChannelBySid(int sid) const {
return data_channel_controller_.FindDataChannelBySid(sid);
}
PeerConnection::InitializePortAllocatorResult
PeerConnection::InitializePortAllocator_n(
const cricket::ServerAddresses& stun_servers,
const std::vector<cricket::RelayServerConfig>& turn_servers,
const RTCConfiguration& configuration) {
RTC_DCHECK_RUN_ON(network_thread());
port_allocator_->Initialize();
// To handle both internal and externally created port allocator, we will
// enable BUNDLE here.
int port_allocator_flags = port_allocator_->flags();
port_allocator_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) {
port_allocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
} else if (absl::StartsWith(factory_->trials().Lookup("WebRTC-IPv6Default"),
"Disabled")) {
port_allocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
}
if (configuration.disable_ipv6_on_wifi) {
port_allocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI);
RTC_LOG(LS_INFO) << "IPv6 candidates on Wi-Fi are disabled.";
}
if (configuration.tcp_candidate_policy == kTcpCandidatePolicyDisabled) {
port_allocator_flags |= cricket::PORTALLOCATOR_DISABLE_TCP;
RTC_LOG(LS_INFO) << "TCP candidates are disabled.";
}
if (configuration.candidate_network_policy ==
kCandidateNetworkPolicyLowCost) {
port_allocator_flags |= cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS;
RTC_LOG(LS_INFO) << "Do not gather candidates on high-cost networks";
}
if (configuration.disable_link_local_networks) {
port_allocator_flags |= cricket::PORTALLOCATOR_DISABLE_LINK_LOCAL_NETWORKS;
RTC_LOG(LS_INFO) << "Disable candidates on link-local network interfaces.";
}
port_allocator_->set_flags(port_allocator_flags);
// No step delay is used while allocating ports.
port_allocator_->set_step_delay(cricket::kMinimumStepDelay);
port_allocator_->SetCandidateFilter(
ConvertIceTransportTypeToCandidateFilter(configuration.type));
port_allocator_->set_max_ipv6_networks(configuration.max_ipv6_networks);
auto turn_servers_copy = turn_servers;
for (auto& turn_server : turn_servers_copy) {
turn_server.tls_cert_verifier = tls_cert_verifier_.get();
}
// Call this last since it may create pooled allocator sessions using the
// properties set above.
port_allocator_->SetConfiguration(
stun_servers, std::move(turn_servers_copy),
configuration.ice_candidate_pool_size,
configuration.GetTurnPortPrunePolicy(), configuration.turn_customizer,
configuration.stun_candidate_keepalive_interval);
InitializePortAllocatorResult res;
res.enable_ipv6 = port_allocator_flags & cricket::PORTALLOCATOR_ENABLE_IPV6;
return res;
}
bool PeerConnection::ReconfigurePortAllocator_n(
const cricket::ServerAddresses& stun_servers,
const std::vector<cricket::RelayServerConfig>& turn_servers,
IceTransportsType type,
int candidate_pool_size,
PortPrunePolicy turn_port_prune_policy,
webrtc::TurnCustomizer* turn_customizer,
absl::optional<int> stun_candidate_keepalive_interval,
bool have_local_description) {
port_allocator_->SetCandidateFilter(
ConvertIceTransportTypeToCandidateFilter(type));
// 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.
if (have_local_description) {
port_allocator_->FreezeCandidatePool();
}
// Add the custom tls turn servers if they exist.
auto turn_servers_copy = turn_servers;
for (auto& turn_server : turn_servers_copy) {
turn_server.tls_cert_verifier = tls_cert_verifier_.get();
}
// Call this last since it may create pooled allocator sessions using the
// candidate filter set above.
return port_allocator_->SetConfiguration(
stun_servers, std::move(turn_servers_copy), candidate_pool_size,
turn_port_prune_policy, turn_customizer,
stun_candidate_keepalive_interval);
}
cricket::ChannelManager* PeerConnection::channel_manager() const {
return factory_->channel_manager();
}
bool PeerConnection::StartRtcEventLog_w(
std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms) {
RTC_DCHECK_RUN_ON(worker_thread());
if (!event_log_) {
return false;
}
return event_log_->StartLogging(std::move(output), output_period_ms);
}
void PeerConnection::StopRtcEventLog_w() {
RTC_DCHECK_RUN_ON(worker_thread());
if (event_log_) {
event_log_->StopLogging();
}
}
cricket::ChannelInterface* PeerConnection::GetChannel(
const std::string& content_name) {
for (const auto& transceiver : transceivers_.List()) {
cricket::ChannelInterface* channel = transceiver->internal()->channel();
if (channel && channel->content_name() == content_name) {
return channel;
}
}
if (rtp_data_channel() &&
rtp_data_channel()->content_name() == content_name) {
return rtp_data_channel();
}
return nullptr;
}
bool PeerConnection::GetSctpSslRole(rtc::SSLRole* role) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!local_description() || !remote_description()) {
RTC_LOG(LS_VERBOSE)
<< "Local and Remote descriptions must be applied to get the "
"SSL Role of the SCTP transport.";
return false;
}
if (!data_channel_controller_.data_channel_transport()) {
RTC_LOG(LS_INFO) << "Non-rejected SCTP m= section is needed to get the "
"SSL Role of the SCTP transport.";
return false;
}
absl::optional<rtc::SSLRole> dtls_role;
if (sctp_mid_s_) {
dtls_role = transport_controller_->GetDtlsRole(*sctp_mid_s_);
if (!dtls_role && sdp_handler_.is_caller().has_value()) {
dtls_role = *sdp_handler_.is_caller() ? rtc::SSL_SERVER : rtc::SSL_CLIENT;
}
*role = *dtls_role;
return true;
}
return false;
}
bool PeerConnection::GetSslRole(const std::string& content_name,
rtc::SSLRole* role) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!local_description() || !remote_description()) {
RTC_LOG(LS_INFO)
<< "Local and Remote descriptions must be applied to get the "
"SSL Role of the session.";
return false;
}
auto dtls_role = transport_controller_->GetDtlsRole(content_name);
if (dtls_role) {
*role = *dtls_role;
return true;
}
return false;
}
void PeerConnection::SetSessionError(SessionError error,
const std::string& error_desc) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (error != session_error_) {
session_error_ = error;
session_error_desc_ = error_desc;
}
}
void PeerConnection::UpdatePayloadTypeDemuxingState(
cricket::ContentSource source) {
// We may need to delete any created default streams and disable creation of
// new ones on the basis of payload type. This is needed to avoid SSRC
// collisions in Call's RtpDemuxer, in the case that a transceiver has
// created a default stream, and then some other channel gets the SSRC
// signaled in the corresponding Unified Plan "m=" section. For more context
// see https://bugs.chromium.org/p/webrtc/issues/detail?id=11477
const SessionDescriptionInterface* sdesc =
(source == cricket::CS_LOCAL ? local_description()
: remote_description());
size_t num_receiving_video_transceivers = 0;
size_t num_receiving_audio_transceivers = 0;
for (auto& content_info : sdesc->description()->contents()) {
if (content_info.rejected ||
(source == cricket::ContentSource::CS_LOCAL &&
!RtpTransceiverDirectionHasRecv(
content_info.media_description()->direction())) ||
(source == cricket::ContentSource::CS_REMOTE &&
!RtpTransceiverDirectionHasSend(
content_info.media_description()->direction()))) {
// Ignore transceivers that are not receiving.
continue;
}
switch (content_info.media_description()->type()) {
case cricket::MediaType::MEDIA_TYPE_AUDIO:
++num_receiving_audio_transceivers;
break;
case cricket::MediaType::MEDIA_TYPE_VIDEO:
++num_receiving_video_transceivers;
break;
default:
// Ignore data channels.
continue;
}
}
bool pt_demuxing_enabled_video = num_receiving_video_transceivers <= 1;
bool pt_demuxing_enabled_audio = num_receiving_audio_transceivers <= 1;
// Gather all updates ahead of time so that all channels can be updated in a
// single Invoke; necessary due to thread guards.
std::vector<std::pair<RtpTransceiverDirection, cricket::ChannelInterface*>>
channels_to_update;
for (const auto& transceiver : transceivers_.List()) {
cricket::ChannelInterface* channel = transceiver->internal()->channel();
const ContentInfo* content =
sdp_handler_.FindMediaSectionForTransceiver(transceiver, sdesc);
if (!channel || !content) {
continue;
}
RtpTransceiverDirection local_direction =
content->media_description()->direction();
if (source == cricket::CS_REMOTE) {
local_direction = RtpTransceiverDirectionReversed(local_direction);
}
channels_to_update.emplace_back(local_direction,
transceiver->internal()->channel());
}
if (!channels_to_update.empty()) {
worker_thread()->Invoke<void>(
RTC_FROM_HERE, [&channels_to_update, pt_demuxing_enabled_audio,
pt_demuxing_enabled_video]() {
for (const auto& it : channels_to_update) {
RtpTransceiverDirection local_direction = it.first;
cricket::ChannelInterface* channel = it.second;
cricket::MediaType media_type = channel->media_type();
if (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO) {
channel->SetPayloadTypeDemuxingEnabled(
pt_demuxing_enabled_audio &&
RtpTransceiverDirectionHasRecv(local_direction));
} else if (media_type == cricket::MediaType::MEDIA_TYPE_VIDEO) {
channel->SetPayloadTypeDemuxingEnabled(
pt_demuxing_enabled_video &&
RtpTransceiverDirectionHasRecv(local_direction));
}
}
});
}
}
RTCError PeerConnection::PushdownMediaDescription(
SdpType type,
cricket::ContentSource source) {
const SessionDescriptionInterface* sdesc =
(source == cricket::CS_LOCAL ? local_description()
: remote_description());
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(sdesc);
UpdatePayloadTypeDemuxingState(source);
// Push down the new SDP media section for each audio/video transceiver.
for (const auto& transceiver : transceivers_.List()) {
const ContentInfo* content_info =
sdp_handler_.FindMediaSectionForTransceiver(transceiver, sdesc);
cricket::ChannelInterface* channel = transceiver->internal()->channel();
if (!channel || !content_info || content_info->rejected) {
continue;
}
const MediaContentDescription* content_desc =
content_info->media_description();
if (!content_desc) {
continue;
}
std::string error;
bool success = (source == cricket::CS_LOCAL)
? channel->SetLocalContent(content_desc, type, &error)
: channel->SetRemoteContent(content_desc, type, &error);
if (!success) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, error);
}
}
// If using the RtpDataChannel, push down the new SDP section for it too.
if (data_channel_controller_.rtp_data_channel()) {
const ContentInfo* data_content =
cricket::GetFirstDataContent(sdesc->description());
if (data_content && !data_content->rejected) {
const MediaContentDescription* data_desc =
data_content->media_description();
if (data_desc) {
std::string error;
bool success =
(source == cricket::CS_LOCAL)
? data_channel_controller_.rtp_data_channel()->SetLocalContent(
data_desc, type, &error)
: data_channel_controller_.rtp_data_channel()->SetRemoteContent(
data_desc, type, &error);
if (!success) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, error);
}
}
}
}
// Need complete offer/answer with an SCTP m= section before starting SCTP,
// according to https://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-19
if (sctp_mid_s_ && local_description() && remote_description()) {
rtc::scoped_refptr<SctpTransport> sctp_transport =
transport_controller_->GetSctpTransport(*sctp_mid_s_);
auto local_sctp_description = cricket::GetFirstSctpDataContentDescription(
local_description()->description());
auto remote_sctp_description = cricket::GetFirstSctpDataContentDescription(
remote_description()->description());
if (sctp_transport && local_sctp_description && remote_sctp_description) {
int max_message_size;
// A remote max message size of zero means "any size supported".
// We configure the connection with our own max message size.
if (remote_sctp_description->max_message_size() == 0) {
max_message_size = local_sctp_description->max_message_size();
} else {
max_message_size =
std::min(local_sctp_description->max_message_size(),
remote_sctp_description->max_message_size());
}
sctp_transport->Start(local_sctp_description->port(),
remote_sctp_description->port(), max_message_size);
}
}
return RTCError::OK();
}
RTCError PeerConnection::PushdownTransportDescription(
cricket::ContentSource source,
SdpType type) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (source == cricket::CS_LOCAL) {
const SessionDescriptionInterface* sdesc = local_description();
RTC_DCHECK(sdesc);
return transport_controller_->SetLocalDescription(type,
sdesc->description());
} else {
const SessionDescriptionInterface* sdesc = remote_description();
RTC_DCHECK(sdesc);
return transport_controller_->SetRemoteDescription(type,
sdesc->description());
}
}
bool PeerConnection::GetTransportDescription(
const SessionDescription* description,
const std::string& content_name,
cricket::TransportDescription* tdesc) {
if (!description || !tdesc) {
return false;
}
const TransportInfo* transport_info =
description->GetTransportInfoByName(content_name);
if (!transport_info) {
return false;
}
*tdesc = transport_info->description;
return true;
}
cricket::IceConfig PeerConnection::ParseIceConfig(
const PeerConnectionInterface::RTCConfiguration& config) const {
cricket::ContinualGatheringPolicy gathering_policy;
switch (config.continual_gathering_policy) {
case PeerConnectionInterface::GATHER_ONCE:
gathering_policy = cricket::GATHER_ONCE;
break;
case PeerConnectionInterface::GATHER_CONTINUALLY:
gathering_policy = cricket::GATHER_CONTINUALLY;
break;
default:
RTC_NOTREACHED();
gathering_policy = cricket::GATHER_ONCE;
}
cricket::IceConfig ice_config;
ice_config.receiving_timeout = RTCConfigurationToIceConfigOptionalInt(
config.ice_connection_receiving_timeout);
ice_config.prioritize_most_likely_candidate_pairs =
config.prioritize_most_likely_ice_candidate_pairs;
ice_config.backup_connection_ping_interval =
RTCConfigurationToIceConfigOptionalInt(
config.ice_backup_candidate_pair_ping_interval);
ice_config.continual_gathering_policy = gathering_policy;
ice_config.presume_writable_when_fully_relayed =
config.presume_writable_when_fully_relayed;
ice_config.surface_ice_candidates_on_ice_transport_type_changed =
config.surface_ice_candidates_on_ice_transport_type_changed;
ice_config.ice_check_interval_strong_connectivity =
config.ice_check_interval_strong_connectivity;
ice_config.ice_check_interval_weak_connectivity =
config.ice_check_interval_weak_connectivity;
ice_config.ice_check_min_interval = config.ice_check_min_interval;
ice_config.ice_unwritable_timeout = config.ice_unwritable_timeout;
ice_config.ice_unwritable_min_checks = config.ice_unwritable_min_checks;
ice_config.ice_inactive_timeout = config.ice_inactive_timeout;
ice_config.stun_keepalive_interval = config.stun_candidate_keepalive_interval;
ice_config.network_preference = config.network_preference;
return ice_config;
}
std::vector<DataChannelStats> PeerConnection::GetDataChannelStats() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return data_channel_controller_.GetDataChannelStats();
}
absl::optional<std::string> PeerConnection::sctp_transport_name() const {
RTC_DCHECK_RUN_ON(signaling_thread());
if (sctp_mid_s_ && transport_controller_) {
auto dtls_transport = transport_controller_->GetDtlsTransport(*sctp_mid_s_);
if (dtls_transport) {
return dtls_transport->transport_name();
}
return absl::optional<std::string>();
}
return absl::optional<std::string>();
}
cricket::CandidateStatsList PeerConnection::GetPooledCandidateStats() const {
cricket::CandidateStatsList candidate_states_list;
network_thread()->Invoke<void>(
RTC_FROM_HERE,
rtc::Bind(&cricket::PortAllocator::GetCandidateStatsFromPooledSessions,
port_allocator_.get(), &candidate_states_list));
return candidate_states_list;
}
std::map<std::string, std::string> PeerConnection::GetTransportNamesByMid()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
std::map<std::string, std::string> transport_names_by_mid;
for (const auto& transceiver : transceivers_.List()) {
cricket::ChannelInterface* channel = transceiver->internal()->channel();
if (channel) {
transport_names_by_mid[channel->content_name()] =
channel->transport_name();
}
}
if (data_channel_controller_.rtp_data_channel()) {
transport_names_by_mid[data_channel_controller_.rtp_data_channel()
->content_name()] =
data_channel_controller_.rtp_data_channel()->transport_name();
}
if (data_channel_controller_.data_channel_transport()) {
absl::optional<std::string> transport_name = sctp_transport_name();
RTC_DCHECK(transport_name);
transport_names_by_mid[*sctp_mid_s_] = *transport_name;
}
return transport_names_by_mid;
}
std::map<std::string, cricket::TransportStats>
PeerConnection::GetTransportStatsByNames(
const std::set<std::string>& transport_names) {
if (!network_thread()->IsCurrent()) {
return network_thread()
->Invoke<std::map<std::string, cricket::TransportStats>>(
RTC_FROM_HERE,
[&] { return GetTransportStatsByNames(transport_names); });
}
RTC_DCHECK_RUN_ON(network_thread());
std::map<std::string, cricket::TransportStats> transport_stats_by_name;
for (const std::string& transport_name : transport_names) {
cricket::TransportStats transport_stats;
bool success =
transport_controller_->GetStats(transport_name, &transport_stats);
if (success) {
transport_stats_by_name[transport_name] = std::move(transport_stats);
} else {
RTC_LOG(LS_ERROR) << "Failed to get transport stats for transport_name="
<< transport_name;
}
}
return transport_stats_by_name;
}
bool PeerConnection::GetLocalCertificate(
const std::string& transport_name,
rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
if (!certificate) {
return false;
}
*certificate = transport_controller_->GetLocalCertificate(transport_name);
return *certificate != nullptr;
}
std::unique_ptr<rtc::SSLCertChain> PeerConnection::GetRemoteSSLCertChain(
const std::string& transport_name) {
return transport_controller_->GetRemoteSSLCertChain(transport_name);
}
cricket::DataChannelType PeerConnection::data_channel_type() const {
return data_channel_controller_.data_channel_type();
}
bool PeerConnection::IceRestartPending(const std::string& content_name) const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_.IceRestartPending(content_name);
}
bool PeerConnection::NeedsIceRestart(const std::string& content_name) const {
return transport_controller_->NeedsIceRestart(content_name);
}
void PeerConnection::OnCertificateReady(
const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
transport_controller_->SetLocalCertificate(certificate);
}
void PeerConnection::OnTransportControllerConnectionState(
cricket::IceConnectionState state) {
switch (state) {
case cricket::kIceConnectionConnecting:
// If the current state is Connected or Completed, then there were
// writable channels but now there are not, so the next state must
// be Disconnected.
// kIceConnectionConnecting is currently used as the default,
// un-connected state by the TransportController, so its only use is
// detecting disconnections.
if (ice_connection_state_ ==
PeerConnectionInterface::kIceConnectionConnected ||
ice_connection_state_ ==
PeerConnectionInterface::kIceConnectionCompleted) {
SetIceConnectionState(
PeerConnectionInterface::kIceConnectionDisconnected);
}
break;
case cricket::kIceConnectionFailed:
SetIceConnectionState(PeerConnectionInterface::kIceConnectionFailed);
break;
case cricket::kIceConnectionConnected:
RTC_LOG(LS_INFO) << "Changing to ICE connected state because "
"all transports are writable.";
SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
NoteUsageEvent(UsageEvent::ICE_STATE_CONNECTED);
break;
case cricket::kIceConnectionCompleted:
RTC_LOG(LS_INFO) << "Changing to ICE completed state because "
"all transports are complete.";
if (ice_connection_state_ !=
PeerConnectionInterface::kIceConnectionConnected) {
// If jumping directly from "checking" to "connected",
// signal "connected" first.
SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
}
SetIceConnectionState(PeerConnectionInterface::kIceConnectionCompleted);
NoteUsageEvent(UsageEvent::ICE_STATE_CONNECTED);
ReportTransportStats();
break;
default:
RTC_NOTREACHED();
}
}
void PeerConnection::OnTransportControllerCandidatesGathered(
const std::string& transport_name,
const cricket::Candidates& candidates) {
int sdp_mline_index;
if (!GetLocalCandidateMediaIndex(transport_name, &sdp_mline_index)) {
RTC_LOG(LS_ERROR)
<< "OnTransportControllerCandidatesGathered: content name "
<< transport_name << " not found";
return;
}
for (cricket::Candidates::const_iterator citer = candidates.begin();
citer != candidates.end(); ++citer) {
// Use transport_name as the candidate media id.
std::unique_ptr<JsepIceCandidate> candidate(
new JsepIceCandidate(transport_name, sdp_mline_index, *citer));
sdp_handler_.AddLocalIceCandidate(candidate.get());
OnIceCandidate(std::move(candidate));
}
}
void PeerConnection::OnTransportControllerCandidateError(
const cricket::IceCandidateErrorEvent& event) {
OnIceCandidateError(event.address, event.port, event.url, event.error_code,
event.error_text);
}
void PeerConnection::OnTransportControllerCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates) {
// Sanity check.
for (const cricket::Candidate& candidate : candidates) {
if (candidate.transport_name().empty()) {
RTC_LOG(LS_ERROR) << "OnTransportControllerCandidatesRemoved: "
"empty content name in candidate "
<< candidate.ToString();
return;
}
}
sdp_handler_.RemoveLocalIceCandidates(candidates);
OnIceCandidatesRemoved(candidates);
}
void PeerConnection::OnTransportControllerCandidateChanged(
const cricket::CandidatePairChangeEvent& event) {
OnSelectedCandidatePairChanged(event);
}
void PeerConnection::OnTransportControllerDtlsHandshakeError(
rtc::SSLHandshakeError error) {
RTC_HISTOGRAM_ENUMERATION(
"WebRTC.PeerConnection.DtlsHandshakeError", static_cast<int>(error),
static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE));
}
void PeerConnection::EnableSending() {
RTC_DCHECK_RUN_ON(signaling_thread());
for (const auto& transceiver : transceivers_.List()) {
cricket::ChannelInterface* channel = transceiver->internal()->channel();
if (channel && !channel->enabled()) {
channel->Enable(true);
}
}
if (data_channel_controller_.rtp_data_channel() &&
!data_channel_controller_.rtp_data_channel()->enabled()) {
data_channel_controller_.rtp_data_channel()->Enable(true);
}
}
// Returns the media index for a local ice candidate given the content name.
bool PeerConnection::GetLocalCandidateMediaIndex(
const std::string& content_name,
int* sdp_mline_index) {
if (!local_description() || !sdp_mline_index) {
return false;
}
bool content_found = false;
const ContentInfos& contents = local_description()->description()->contents();
for (size_t index = 0; index < contents.size(); ++index) {
if (contents[index].name == content_name) {
*sdp_mline_index = static_cast<int>(index);
content_found = true;
break;
}
}
return content_found;
}
bool PeerConnection::UseCandidatesInSessionDescription(
const SessionDescriptionInterface* remote_desc) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!remote_desc) {
return true;
}
bool ret = true;
for (size_t m = 0; m < remote_desc->number_of_mediasections(); ++m) {
const IceCandidateCollection* candidates = remote_desc->candidates(m);
for (size_t n = 0; n < candidates->count(); ++n) {
const IceCandidateInterface* candidate = candidates->at(n);
bool valid = false;
if (!ReadyToUseRemoteCandidate(candidate, remote_desc, &valid)) {
if (valid) {
RTC_LOG(LS_INFO)
<< "UseCandidatesInSessionDescription: Not ready to use "
"candidate.";
}
continue;
}
ret = UseCandidate(candidate);
if (!ret) {
break;
}
}
}
return ret;
}
bool PeerConnection::UseCandidate(const IceCandidateInterface* candidate) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTCErrorOr<const cricket::ContentInfo*> result =
FindContentInfo(remote_description(), candidate);
if (!result.ok()) {
RTC_LOG(LS_ERROR) << "UseCandidate: Invalid candidate. "
<< result.error().message();
return false;
}
std::vector<cricket::Candidate> candidates;
candidates.push_back(candidate->candidate());
// Invoking BaseSession method to handle remote candidates.
RTCError error = transport_controller_->AddRemoteCandidates(
result.value()->name, candidates);
if (error.ok()) {
ReportRemoteIceCandidateAdded(candidate->candidate());
// Candidates successfully submitted for checking.
if (ice_connection_state_ == PeerConnectionInterface::kIceConnectionNew ||
ice_connection_state_ ==
PeerConnectionInterface::kIceConnectionDisconnected) {
// If state is New, then the session has just gotten its first remote ICE
// candidates, so go to Checking.
// If state is Disconnected, the session is re-using old candidates or
// receiving additional ones, so go to Checking.
// If state is Connected, stay Connected.
// TODO(bemasc): If state is Connected, and the new candidates are for a
// newly added transport, then the state actually _should_ move to
// checking. Add a way to distinguish that case.
SetIceConnectionState(PeerConnectionInterface::kIceConnectionChecking);
}
// TODO(bemasc): If state is Completed, go back to Connected.
} else {
RTC_LOG(LS_WARNING) << error.message();
}
return true;
}
RTCErrorOr<const cricket::ContentInfo*> PeerConnection::FindContentInfo(
const SessionDescriptionInterface* description,
const IceCandidateInterface* candidate) {
if (candidate->sdp_mline_index() >= 0) {
size_t mediacontent_index =
static_cast<size_t>(candidate->sdp_mline_index());
size_t content_size = description->description()->contents().size();
if (mediacontent_index < content_size) {
return &description->description()->contents()[mediacontent_index];
} else {
return RTCError(RTCErrorType::INVALID_RANGE,
"Media line index (" +
rtc::ToString(candidate->sdp_mline_index()) +
") out of range (number of mlines: " +
rtc::ToString(content_size) + ").");
}
} else if (!candidate->sdp_mid().empty()) {
auto& contents = description->description()->contents();
auto it = absl::c_find_if(
contents, [candidate](const cricket::ContentInfo& content_info) {
return content_info.mid() == candidate->sdp_mid();
});
if (it == contents.end()) {
return RTCError(
RTCErrorType::INVALID_PARAMETER,
"Mid " + candidate->sdp_mid() +
" specified but no media section with that mid found.");
} else {
return &*it;
}
}
return RTCError(RTCErrorType::INVALID_PARAMETER,
"Neither sdp_mline_index nor sdp_mid specified.");
}
void PeerConnection::RemoveUnusedChannels(const SessionDescription* desc) {
RTC_DCHECK_RUN_ON(signaling_thread());
// Destroy video channel first since it may have a pointer to the
// voice channel.
const cricket::ContentInfo* video_info = cricket::GetFirstVideoContent(desc);
if (!video_info || video_info->rejected) {
DestroyTransceiverChannel(GetVideoTransceiver());
}
const cricket::ContentInfo* audio_info = cricket::GetFirstAudioContent(desc);
if (!audio_info || audio_info->rejected) {
DestroyTransceiverChannel(GetAudioTransceiver());
}
const cricket::ContentInfo* data_info = cricket::GetFirstDataContent(desc);
if (!data_info || data_info->rejected) {
DestroyDataChannelTransport();
}
}
RTCError PeerConnection::CreateChannels(const SessionDescription& desc) {
// Creating the media channels. Transports should already have been created
// at this point.
RTC_DCHECK_RUN_ON(signaling_thread());
const cricket::ContentInfo* voice = cricket::GetFirstAudioContent(&desc);
if (voice && !voice->rejected &&
!GetAudioTransceiver()->internal()->channel()) {
cricket::VoiceChannel* voice_channel = CreateVoiceChannel(voice->name);
if (!voice_channel) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to create voice channel.");
}
GetAudioTransceiver()->internal()->SetChannel(voice_channel);
}
const cricket::ContentInfo* video = cricket::GetFirstVideoContent(&desc);
if (video && !video->rejected &&
!GetVideoTransceiver()->internal()->channel()) {
cricket::VideoChannel* video_channel = CreateVideoChannel(video->name);
if (!video_channel) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to create video channel.");
}
GetVideoTransceiver()->internal()->SetChannel(video_channel);
}
const cricket::ContentInfo* data = cricket::GetFirstDataContent(&desc);
if (data_channel_type() != cricket::DCT_NONE && data && !data->rejected &&
!data_channel_controller_.rtp_data_channel() &&
!data_channel_controller_.data_channel_transport()) {
if (!CreateDataChannel(data->name)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to create data channel.");
}
}
return RTCError::OK();
}
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
cricket::VoiceChannel* PeerConnection::CreateVoiceChannel(
const std::string& mid) {
RTC_DCHECK_RUN_ON(signaling_thread());
RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
// TODO(bugs.webrtc.org/11992): CreateVoiceChannel internally switches to the
// worker thread. We shouldn't be using the |call_ptr_| hack here but simply
// be on the worker thread and use |call_| (update upstream code).
cricket::VoiceChannel* voice_channel = channel_manager()->CreateVoiceChannel(
call_ptr_, configuration_.media_config, rtp_transport, signaling_thread(),
mid, SrtpRequired(), GetCryptoOptions(), &ssrc_generator_,
audio_options_);
if (!voice_channel) {
return nullptr;
}
voice_channel->SignalSentPacket().connect(this,
&PeerConnection::OnSentPacket_w);
voice_channel->SetRtpTransport(rtp_transport);
return voice_channel;
}
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
cricket::VideoChannel* PeerConnection::CreateVideoChannel(
const std::string& mid) {
RTC_DCHECK_RUN_ON(signaling_thread());
RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
// TODO(bugs.webrtc.org/11992): CreateVideoChannel internally switches to the
// worker thread. We shouldn't be using the |call_ptr_| hack here but simply
// be on the worker thread and use |call_| (update upstream code).
cricket::VideoChannel* video_channel = channel_manager()->CreateVideoChannel(
call_ptr_, configuration_.media_config, rtp_transport, signaling_thread(),
mid, SrtpRequired(), GetCryptoOptions(), &ssrc_generator_, video_options_,
video_bitrate_allocator_factory_.get());
if (!video_channel) {
return nullptr;
}
video_channel->SignalSentPacket().connect(this,
&PeerConnection::OnSentPacket_w);
video_channel->SetRtpTransport(rtp_transport);
return video_channel;
}
bool PeerConnection::CreateDataChannel(const std::string& mid) {
RTC_DCHECK_RUN_ON(signaling_thread());
switch (data_channel_type()) {
case cricket::DCT_SCTP:
if (network_thread()->Invoke<bool>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnection::SetupDataChannelTransport_n, this,
mid))) {
sctp_mid_s_ = mid;
} else {
return false;
}
return true;
case cricket::DCT_RTP:
default:
RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
// TODO(bugs.webrtc.org/9987): set_rtp_data_channel() should be called on
// the network thread like set_data_channel_transport is.
data_channel_controller_.set_rtp_data_channel(
channel_manager()->CreateRtpDataChannel(
configuration_.media_config, rtp_transport, signaling_thread(),
mid, SrtpRequired(), GetCryptoOptions(), &ssrc_generator_));
if (!data_channel_controller_.rtp_data_channel()) {
return false;
}
data_channel_controller_.rtp_data_channel()->SignalSentPacket().connect(
this, &PeerConnection::OnSentPacket_w);
data_channel_controller_.rtp_data_channel()->SetRtpTransport(
rtp_transport);
have_pending_rtp_data_channel_ = true;
return true;
}
return false;
}
Call::Stats PeerConnection::GetCallStats() {
if (!worker_thread()->IsCurrent()) {
return worker_thread()->Invoke<Call::Stats>(
RTC_FROM_HERE, rtc::Bind(&PeerConnection::GetCallStats, this));
}
RTC_DCHECK_RUN_ON(worker_thread());
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
if (call_) {
return call_->GetStats();
} else {
return Call::Stats();
}
}
bool PeerConnection::SetupDataChannelTransport_n(const std::string& mid) {
DataChannelTransportInterface* transport =
transport_controller_->GetDataChannelTransport(mid);
if (!transport) {
RTC_LOG(LS_ERROR)
<< "Data channel transport is not available for data channels, mid="
<< mid;
return false;
}
RTC_LOG(LS_INFO) << "Setting up data channel transport for mid=" << mid;
data_channel_controller_.set_data_channel_transport(transport);
data_channel_controller_.SetupDataChannelTransport_n();
sctp_mid_n_ = mid;
// Note: setting the data sink and checking initial state must be done last,
// after setting up the data channel. Setting the data sink may trigger
// callbacks to PeerConnection which require the transport to be completely
// set up (eg. OnReadyToSend()).
transport->SetDataSink(&data_channel_controller_);
return true;
}
void PeerConnection::TeardownDataChannelTransport_n() {
if (!sctp_mid_n_ && !data_channel_controller_.data_channel_transport()) {
return;
}
RTC_LOG(LS_INFO) << "Tearing down data channel transport for mid="
<< *sctp_mid_n_;
// |sctp_mid_| may still be active through an SCTP transport. If not, unset
// it.
sctp_mid_n_.reset();
data_channel_controller_.TeardownDataChannelTransport_n();
}
// Returns false if bundle is enabled and rtcp_mux is disabled.
bool PeerConnection::ValidateBundleSettings(const SessionDescription* desc) {
bool bundle_enabled = desc->HasGroup(cricket::GROUP_TYPE_BUNDLE);
if (!bundle_enabled)
return true;
const cricket::ContentGroup* bundle_group =
desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
RTC_DCHECK(bundle_group != NULL);
const cricket::ContentInfos& contents = desc->contents();
for (cricket::ContentInfos::const_iterator citer = contents.begin();
citer != contents.end(); ++citer) {
const cricket::ContentInfo* content = (&*citer);
RTC_DCHECK(content != NULL);
if (bundle_group->HasContentName(content->name) && !content->rejected &&
content->type == MediaProtocolType::kRtp) {
if (!HasRtcpMuxEnabled(content))
return false;
}
}
// RTCP-MUX is enabled in all the contents.
return true;
}
bool PeerConnection::HasRtcpMuxEnabled(const cricket::ContentInfo* content) {
return content->media_description()->rtcp_mux();
}
const char* PeerConnection::SessionErrorToString(SessionError error) const {
switch (error) {
case SessionError::kNone:
return "ERROR_NONE";
case SessionError::kContent:
return "ERROR_CONTENT";
case SessionError::kTransport:
return "ERROR_TRANSPORT";
}
RTC_NOTREACHED();
return "";
}
std::string PeerConnection::GetSessionErrorMsg() {
RTC_DCHECK_RUN_ON(signaling_thread());
rtc::StringBuilder desc;
desc << kSessionError << SessionErrorToString(session_error()) << ". ";
desc << kSessionErrorDesc << session_error_desc() << ".";
return desc.Release();
}
void PeerConnection::ReportSdpFormatReceived(
const SessionDescriptionInterface& remote_offer) {
int num_audio_mlines = 0;
int num_video_mlines = 0;
int num_audio_tracks = 0;
int num_video_tracks = 0;
for (const ContentInfo& content : remote_offer.description()->contents()) {
cricket::MediaType media_type = content.media_description()->type();
int num_tracks = std::max(
1, static_cast<int>(content.media_description()->streams().size()));
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
num_audio_mlines += 1;
num_audio_tracks += num_tracks;
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
num_video_mlines += 1;
num_video_tracks += num_tracks;
}
}
SdpFormatReceived format = kSdpFormatReceivedNoTracks;
if (num_audio_mlines > 1 || num_video_mlines > 1) {
format = kSdpFormatReceivedComplexUnifiedPlan;
} else if (num_audio_tracks > 1 || num_video_tracks > 1) {
format = kSdpFormatReceivedComplexPlanB;
} else if (num_audio_tracks > 0 || num_video_tracks > 0) {
format = kSdpFormatReceivedSimple;
}
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SdpFormatReceived", format,
kSdpFormatReceivedMax);
}
void PeerConnection::ReportIceCandidateCollected(
const cricket::Candidate& candidate) {
NoteUsageEvent(UsageEvent::CANDIDATE_COLLECTED);
if (candidate.address().IsPrivateIP()) {
NoteUsageEvent(UsageEvent::PRIVATE_CANDIDATE_COLLECTED);
}
if (candidate.address().IsUnresolvedIP()) {
NoteUsageEvent(UsageEvent::MDNS_CANDIDATE_COLLECTED);
}
if (candidate.address().family() == AF_INET6) {
NoteUsageEvent(UsageEvent::IPV6_CANDIDATE_COLLECTED);
}
}
void PeerConnection::ReportRemoteIceCandidateAdded(
const cricket::Candidate& candidate) {
NoteUsageEvent(UsageEvent::REMOTE_CANDIDATE_ADDED);
if (candidate.address().IsPrivateIP()) {
NoteUsageEvent(UsageEvent::REMOTE_PRIVATE_CANDIDATE_ADDED);
}
if (candidate.address().IsUnresolvedIP()) {
NoteUsageEvent(UsageEvent::REMOTE_MDNS_CANDIDATE_ADDED);
}
if (candidate.address().family() == AF_INET6) {
NoteUsageEvent(UsageEvent::REMOTE_IPV6_CANDIDATE_ADDED);
}
}
void PeerConnection::NoteUsageEvent(UsageEvent event) {
RTC_DCHECK_RUN_ON(signaling_thread());
usage_event_accumulator_ |= static_cast<int>(event);
}
void PeerConnection::ReportUsagePattern() const {
RTC_DLOG(LS_INFO) << "Usage signature is " << usage_event_accumulator_;
RTC_HISTOGRAM_ENUMERATION_SPARSE("WebRTC.PeerConnection.UsagePattern",
usage_event_accumulator_,
static_cast<int>(UsageEvent::MAX_VALUE));
const int bad_bits =
static_cast<int>(UsageEvent::SET_LOCAL_DESCRIPTION_SUCCEEDED) |
static_cast<int>(UsageEvent::CANDIDATE_COLLECTED);
const int good_bits =
static_cast<int>(UsageEvent::SET_REMOTE_DESCRIPTION_SUCCEEDED) |
static_cast<int>(UsageEvent::REMOTE_CANDIDATE_ADDED) |
static_cast<int>(UsageEvent::ICE_STATE_CONNECTED);
if ((usage_event_accumulator_ & bad_bits) == bad_bits &&
(usage_event_accumulator_ & good_bits) == 0) {
// If called after close(), we can't report, because observer may have
// been deallocated, and therefore pointer is null. Write to log instead.
if (observer_) {
Observer()->OnInterestingUsage(usage_event_accumulator_);
} else {
RTC_LOG(LS_INFO) << "Interesting usage signature "
<< usage_event_accumulator_
<< " observed after observer shutdown";
}
}
}
void PeerConnection::ReportNegotiatedSdpSemantics(
const SessionDescriptionInterface& answer) {
SdpSemanticNegotiated semantics_negotiated;
switch (answer.description()->msid_signaling()) {
case 0:
semantics_negotiated = kSdpSemanticNegotiatedNone;
break;
case cricket::kMsidSignalingMediaSection:
semantics_negotiated = kSdpSemanticNegotiatedUnifiedPlan;
break;
case cricket::kMsidSignalingSsrcAttribute:
semantics_negotiated = kSdpSemanticNegotiatedPlanB;
break;
case cricket::kMsidSignalingMediaSection |
cricket::kMsidSignalingSsrcAttribute:
semantics_negotiated = kSdpSemanticNegotiatedMixed;
break;
default:
RTC_NOTREACHED();
}
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SdpSemanticNegotiated",
semantics_negotiated, kSdpSemanticNegotiatedMax);
}
// We need to check the local/remote description for the Transport instead of
// the session, because a new Transport added during renegotiation may have
// them unset while the session has them set from the previous negotiation.
// Not doing so may trigger the auto generation of transport description and
// mess up DTLS identity information, ICE credential, etc.
bool PeerConnection::ReadyToUseRemoteCandidate(
const IceCandidateInterface* candidate,
const SessionDescriptionInterface* remote_desc,
bool* valid) {
RTC_DCHECK_RUN_ON(signaling_thread());
*valid = true;
const SessionDescriptionInterface* current_remote_desc =
remote_desc ? remote_desc : remote_description();
if (!current_remote_desc) {
return false;
}
RTCErrorOr<const cricket::ContentInfo*> result =
FindContentInfo(current_remote_desc, candidate);
if (!result.ok()) {
RTC_LOG(LS_ERROR) << "ReadyToUseRemoteCandidate: Invalid candidate. "
<< result.error().message();
*valid = false;
return false;
}
std::string transport_name = GetTransportName(result.value()->name);
return !transport_name.empty();
}
bool PeerConnection::SrtpRequired() const {
return (dtls_enabled_ ||
sdp_handler_.webrtc_session_desc_factory()->SdesPolicy() ==
cricket::SEC_REQUIRED);
}
void PeerConnection::OnTransportControllerGatheringState(
cricket::IceGatheringState state) {
RTC_DCHECK(signaling_thread()->IsCurrent());
if (state == cricket::kIceGatheringGathering) {
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringGathering);
} else if (state == cricket::kIceGatheringComplete) {
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringComplete);
} else if (state == cricket::kIceGatheringNew) {
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringNew);
} else {
RTC_LOG(LS_ERROR) << "Unknown state received: " << state;
RTC_NOTREACHED();
}
}
void PeerConnection::ReportTransportStats() {
std::map<std::string, std::set<cricket::MediaType>>
media_types_by_transport_name;
for (const auto& transceiver : transceivers_.List()) {
if (transceiver->internal()->channel()) {
const std::string& transport_name =
transceiver->internal()->channel()->transport_name();
media_types_by_transport_name[transport_name].insert(
transceiver->media_type());
}
}
if (rtp_data_channel()) {
media_types_by_transport_name[rtp_data_channel()->transport_name()].insert(
cricket::MEDIA_TYPE_DATA);
}
absl::optional<std::string> transport_name = sctp_transport_name();
if (transport_name) {
media_types_by_transport_name[*transport_name].insert(
cricket::MEDIA_TYPE_DATA);
}
for (const auto& entry : media_types_by_transport_name) {
const std::string& transport_name = entry.first;
const std::set<cricket::MediaType> media_types = entry.second;
cricket::TransportStats stats;
if (transport_controller_->GetStats(transport_name, &stats)) {
ReportBestConnectionState(stats);
ReportNegotiatedCiphers(stats, media_types);
}
}
}
// Walk through the ConnectionInfos to gather best connection usage
// for IPv4 and IPv6.
void PeerConnection::ReportBestConnectionState(
const cricket::TransportStats& stats) {
for (const cricket::TransportChannelStats& channel_stats :
stats.channel_stats) {
for (const cricket::ConnectionInfo& connection_info :
channel_stats.ice_transport_stats.connection_infos) {
if (!connection_info.best_connection) {
continue;
}
const cricket::Candidate& local = connection_info.local_candidate;
const cricket::Candidate& remote = connection_info.remote_candidate;
// Increment the counter for IceCandidatePairType.
if (local.protocol() == cricket::TCP_PROTOCOL_NAME ||
(local.type() == RELAY_PORT_TYPE &&
local.relay_protocol() == cricket::TCP_PROTOCOL_NAME)) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_TCP",
GetIceCandidatePairCounter(local, remote),
kIceCandidatePairMax);
} else if (local.protocol() == cricket::UDP_PROTOCOL_NAME) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_UDP",
GetIceCandidatePairCounter(local, remote),
kIceCandidatePairMax);
} else {
RTC_CHECK(0);
}
// Increment the counter for IP type.
if (local.address().family() == AF_INET) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics",
kBestConnections_IPv4,
kPeerConnectionAddressFamilyCounter_Max);
} else if (local.address().family() == AF_INET6) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics",
kBestConnections_IPv6,
kPeerConnectionAddressFamilyCounter_Max);
} else {
RTC_CHECK(!local.address().hostname().empty() &&
local.address().IsUnresolvedIP());
}
return;
}
}
}
void PeerConnection::ReportNegotiatedCiphers(
const cricket::TransportStats& stats,
const std::set<cricket::MediaType>& media_types) {
if (!dtls_enabled_ || stats.channel_stats.empty()) {
return;
}
int srtp_crypto_suite = stats.channel_stats[0].srtp_crypto_suite;
int ssl_cipher_suite = stats.channel_stats[0].ssl_cipher_suite;
if (srtp_crypto_suite == rtc::SRTP_INVALID_CRYPTO_SUITE &&
ssl_cipher_suite == rtc::TLS_NULL_WITH_NULL_NULL) {
return;
}
if (srtp_crypto_suite != rtc::SRTP_INVALID_CRYPTO_SUITE) {
for (cricket::MediaType media_type : media_types) {
switch (media_type) {
case cricket::MEDIA_TYPE_AUDIO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio", srtp_crypto_suite,
rtc::SRTP_CRYPTO_SUITE_MAX_VALUE);
break;
case cricket::MEDIA_TYPE_VIDEO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SrtpCryptoSuite.Video", srtp_crypto_suite,
rtc::SRTP_CRYPTO_SUITE_MAX_VALUE);
break;
case cricket::MEDIA_TYPE_DATA:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SrtpCryptoSuite.Data", srtp_crypto_suite,
rtc::SRTP_CRYPTO_SUITE_MAX_VALUE);
break;
default:
RTC_NOTREACHED();
continue;
}
}
}
if (ssl_cipher_suite != rtc::TLS_NULL_WITH_NULL_NULL) {
for (cricket::MediaType media_type : media_types) {
switch (media_type) {
case cricket::MEDIA_TYPE_AUDIO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslCipherSuite.Audio", ssl_cipher_suite,
rtc::SSL_CIPHER_SUITE_MAX_VALUE);
break;
case cricket::MEDIA_TYPE_VIDEO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslCipherSuite.Video", ssl_cipher_suite,
rtc::SSL_CIPHER_SUITE_MAX_VALUE);
break;
case cricket::MEDIA_TYPE_DATA:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslCipherSuite.Data", ssl_cipher_suite,
rtc::SSL_CIPHER_SUITE_MAX_VALUE);
break;
default:
RTC_NOTREACHED();
continue;
}
}
}
}
void PeerConnection::OnSentPacket_w(const rtc::SentPacket& sent_packet) {
RTC_DCHECK_RUN_ON(worker_thread());
RTC_DCHECK(call_);
call_->OnSentPacket(sent_packet);
}
const std::string PeerConnection::GetTransportName(
const std::string& content_name) {
cricket::ChannelInterface* channel = GetChannel(content_name);
if (channel) {
return channel->transport_name();
}
if (data_channel_controller_.data_channel_transport()) {
RTC_DCHECK(sctp_mid_s_);
if (content_name == *sctp_mid_s_) {
return *sctp_transport_name();
}
}
// Return an empty string if failed to retrieve the transport name.
return "";
}
void PeerConnection::DestroyTransceiverChannel(
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
transceiver) {
RTC_DCHECK(transceiver);
cricket::ChannelInterface* channel = transceiver->internal()->channel();
if (channel) {
transceiver->internal()->SetChannel(nullptr);
DestroyChannelInterface(channel);
}
}
void PeerConnection::DestroyDataChannelTransport() {
RTC_DCHECK_RUN_ON(signaling_thread());
if (data_channel_controller_.rtp_data_channel()) {
data_channel_controller_.OnTransportChannelClosed();
DestroyChannelInterface(data_channel_controller_.rtp_data_channel());
data_channel_controller_.set_rtp_data_channel(nullptr);
}
// Note: Cannot use rtc::Bind to create a functor to invoke because it will
// grab a reference to this PeerConnection. If this is called from the
// PeerConnection destructor, the RefCountedObject vtable will have already
// been destroyed (since it is a subclass of PeerConnection) and using
// rtc::Bind will cause "Pure virtual function called" error to appear.
if (sctp_mid_s_) {
data_channel_controller_.OnTransportChannelClosed();
network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
RTC_DCHECK_RUN_ON(network_thread());
TeardownDataChannelTransport_n();
});
sctp_mid_s_.reset();
}
}
void PeerConnection::DestroyChannelInterface(
cricket::ChannelInterface* channel) {
// TODO(bugs.webrtc.org/11992): All the below methods should be called on the
// worker thread. (they switch internally anyway). Change
// DestroyChannelInterface to either be called on the worker thread, or do
// this asynchronously on the worker.
RTC_DCHECK(channel);
switch (channel->media_type()) {
case cricket::MEDIA_TYPE_AUDIO:
channel_manager()->DestroyVoiceChannel(
static_cast<cricket::VoiceChannel*>(channel));
break;
case cricket::MEDIA_TYPE_VIDEO:
channel_manager()->DestroyVideoChannel(
static_cast<cricket::VideoChannel*>(channel));
break;
case cricket::MEDIA_TYPE_DATA:
channel_manager()->DestroyRtpDataChannel(
static_cast<cricket::RtpDataChannel*>(channel));
break;
default:
RTC_NOTREACHED() << "Unknown media type: " << channel->media_type();
break;
}
}
bool PeerConnection::OnTransportChanged(
const std::string& mid,
RtpTransportInternal* rtp_transport,
rtc::scoped_refptr<DtlsTransport> dtls_transport,
DataChannelTransportInterface* data_channel_transport) {
RTC_DCHECK_RUN_ON(network_thread());
bool ret = true;
auto base_channel = GetChannel(mid);
if (base_channel) {
ret = base_channel->SetRtpTransport(rtp_transport);
}
if (mid == sctp_mid_n_) {
data_channel_controller_.OnTransportChanged(data_channel_transport);
}
return ret;
}
void PeerConnection::OnSetStreams() {
RTC_DCHECK_RUN_ON(signaling_thread());
if (IsUnifiedPlan())
sdp_handler_.UpdateNegotiationNeeded();
}
PeerConnectionObserver* PeerConnection::Observer() const {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(observer_);
return observer_;
}
CryptoOptions PeerConnection::GetCryptoOptions() {
RTC_DCHECK_RUN_ON(signaling_thread());
// TODO(bugs.webrtc.org/9891) - Remove PeerConnectionFactory::CryptoOptions
// after it has been removed.
return configuration_.crypto_options.has_value()
? *configuration_.crypto_options
: factory_->options().crypto_options;
}
void PeerConnection::ClearStatsCache() {
RTC_DCHECK_RUN_ON(signaling_thread());
if (stats_collector_) {
stats_collector_->ClearCachedStatsReport();
}
}
void PeerConnection::RequestUsagePatternReportForTesting() {
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_REPORT_USAGE_PATTERN,
nullptr);
}
bool PeerConnection::ShouldFireNegotiationNeededEvent(uint32_t event_id) {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_.ShouldFireNegotiationNeededEvent(event_id);
}
std::function<void(const rtc::CopyOnWriteBuffer& packet,
int64_t packet_time_us)>
PeerConnection::InitializeRtcpCallback() {
RTC_DCHECK_RUN_ON(signaling_thread());
auto flag =
worker_thread()->Invoke<rtc::scoped_refptr<PendingTaskSafetyFlag>>(
RTC_FROM_HERE, [this] {
RTC_DCHECK_RUN_ON(worker_thread());
if (!call_)
return rtc::scoped_refptr<PendingTaskSafetyFlag>();
if (!call_safety_)
call_safety_.reset(new ScopedTaskSafety());
return call_safety_->flag();
});
if (!flag)
return [](const rtc::CopyOnWriteBuffer&, int64_t) {};
return [this, flag = std::move(flag)](const rtc::CopyOnWriteBuffer& packet,
int64_t packet_time_us) {
RTC_DCHECK_RUN_ON(network_thread());
// TODO(bugs.webrtc.org/11993): We should actually be delivering this call
// directly to the Call class somehow directly on the network thread and not
// incur this hop here. The DeliverPacket() method will eventually just have
// to hop back over to the network thread.
worker_thread()->PostTask(ToQueuedTask(flag, [this, packet,
packet_time_us] {
RTC_DCHECK_RUN_ON(worker_thread());
call_->Receiver()->DeliverPacket(MediaType::ANY, packet, packet_time_us);
}));
};
}
} // namespace webrtc