webrtc_m130/pc/peer_connection_factory.cc
Artem Titov 397cd82eaf Create port allocator on signaling thread and init on network
Port allocator can be created on one thread and then initialized and
used on another. So we can avoid sync invoke to network thread to create
port allocator.

Bug: webrtc:11799
Change-Id: I5020093a41acbf7e372f2e4970e016ce14a7f406
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180122
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31805}
2020-07-29 11:31:43 +00:00

401 lines
15 KiB
C++

/*
* Copyright 2004 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_factory.h"
#include <cstdio>
#include <memory>
#include <utility>
#include <vector>
#include "absl/strings/match.h"
#include "api/fec_controller.h"
#include "api/media_stream_proxy.h"
#include "api/media_stream_track_proxy.h"
#include "api/network_state_predictor.h"
#include "api/peer_connection_factory_proxy.h"
#include "api/peer_connection_proxy.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/transport/field_trial_based_config.h"
#include "api/turn_customizer.h"
#include "api/units/data_rate.h"
#include "api/video_track_source_proxy.h"
#include "media/base/rtp_data_engine.h"
#include "media/sctp/sctp_transport.h"
#include "p2p/base/basic_async_resolver_factory.h"
#include "p2p/base/basic_packet_socket_factory.h"
#include "p2p/base/default_ice_transport_factory.h"
#include "p2p/client/basic_port_allocator.h"
#include "pc/audio_track.h"
#include "pc/local_audio_source.h"
#include "pc/media_stream.h"
#include "pc/peer_connection.h"
#include "pc/rtp_parameters_conversion.h"
#include "pc/video_track.h"
#include "rtc_base/bind.h"
#include "rtc_base/checks.h"
#include "rtc_base/experiments/field_trial_parser.h"
#include "rtc_base/experiments/field_trial_units.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/system/file_wrapper.h"
namespace webrtc {
rtc::scoped_refptr<PeerConnectionFactoryInterface>
CreateModularPeerConnectionFactory(
PeerConnectionFactoryDependencies dependencies) {
rtc::scoped_refptr<PeerConnectionFactory> pc_factory(
new rtc::RefCountedObject<PeerConnectionFactory>(
std::move(dependencies)));
// Call Initialize synchronously but make sure it is executed on
// |signaling_thread|.
MethodCall<PeerConnectionFactory, bool> call(
pc_factory.get(), &PeerConnectionFactory::Initialize);
bool result = call.Marshal(RTC_FROM_HERE, pc_factory->signaling_thread());
if (!result) {
return nullptr;
}
return PeerConnectionFactoryProxy::Create(pc_factory->signaling_thread(),
pc_factory);
}
PeerConnectionFactory::PeerConnectionFactory(
PeerConnectionFactoryDependencies dependencies)
: wraps_current_thread_(false),
network_thread_(dependencies.network_thread),
worker_thread_(dependencies.worker_thread),
signaling_thread_(dependencies.signaling_thread),
task_queue_factory_(std::move(dependencies.task_queue_factory)),
media_engine_(std::move(dependencies.media_engine)),
call_factory_(std::move(dependencies.call_factory)),
event_log_factory_(std::move(dependencies.event_log_factory)),
fec_controller_factory_(std::move(dependencies.fec_controller_factory)),
network_state_predictor_factory_(
std::move(dependencies.network_state_predictor_factory)),
injected_network_controller_factory_(
std::move(dependencies.network_controller_factory)),
neteq_factory_(std::move(dependencies.neteq_factory)),
trials_(dependencies.trials ? std::move(dependencies.trials)
: std::make_unique<FieldTrialBasedConfig>()) {
if (!network_thread_) {
owned_network_thread_ = rtc::Thread::CreateWithSocketServer();
owned_network_thread_->SetName("pc_network_thread", nullptr);
owned_network_thread_->Start();
network_thread_ = owned_network_thread_.get();
}
if (!worker_thread_) {
owned_worker_thread_ = rtc::Thread::Create();
owned_worker_thread_->SetName("pc_worker_thread", nullptr);
owned_worker_thread_->Start();
worker_thread_ = owned_worker_thread_.get();
}
if (!signaling_thread_) {
signaling_thread_ = rtc::Thread::Current();
if (!signaling_thread_) {
// If this thread isn't already wrapped by an rtc::Thread, create a
// wrapper and own it in this class.
signaling_thread_ = rtc::ThreadManager::Instance()->WrapCurrentThread();
wraps_current_thread_ = true;
}
}
signaling_thread_->AllowInvokesToThread(worker_thread_);
signaling_thread_->AllowInvokesToThread(network_thread_);
worker_thread_->AllowInvokesToThread(network_thread_);
network_thread_->DisallowAllInvokes();
}
PeerConnectionFactory::~PeerConnectionFactory() {
RTC_DCHECK(signaling_thread_->IsCurrent());
channel_manager_.reset(nullptr);
// Make sure |worker_thread_| and |signaling_thread_| outlive
// |default_socket_factory_| and |default_network_manager_|.
default_socket_factory_ = nullptr;
default_network_manager_ = nullptr;
if (wraps_current_thread_)
rtc::ThreadManager::Instance()->UnwrapCurrentThread();
}
bool PeerConnectionFactory::Initialize() {
RTC_DCHECK(signaling_thread_->IsCurrent());
rtc::InitRandom(rtc::Time32());
default_network_manager_.reset(new rtc::BasicNetworkManager());
if (!default_network_manager_) {
return false;
}
default_socket_factory_.reset(
new rtc::BasicPacketSocketFactory(network_thread_));
if (!default_socket_factory_) {
return false;
}
channel_manager_ = std::make_unique<cricket::ChannelManager>(
std::move(media_engine_), std::make_unique<cricket::RtpDataEngine>(),
worker_thread_, network_thread_);
channel_manager_->SetVideoRtxEnabled(true);
if (!channel_manager_->Init()) {
return false;
}
return true;
}
void PeerConnectionFactory::SetOptions(const Options& options) {
options_ = options;
}
RtpCapabilities PeerConnectionFactory::GetRtpSenderCapabilities(
cricket::MediaType kind) const {
RTC_DCHECK_RUN_ON(signaling_thread_);
switch (kind) {
case cricket::MEDIA_TYPE_AUDIO: {
cricket::AudioCodecs cricket_codecs;
channel_manager_->GetSupportedAudioSendCodecs(&cricket_codecs);
return ToRtpCapabilities(
cricket_codecs,
channel_manager_->GetDefaultEnabledAudioRtpHeaderExtensions());
}
case cricket::MEDIA_TYPE_VIDEO: {
cricket::VideoCodecs cricket_codecs;
channel_manager_->GetSupportedVideoSendCodecs(&cricket_codecs);
return ToRtpCapabilities(
cricket_codecs,
channel_manager_->GetDefaultEnabledVideoRtpHeaderExtensions());
}
case cricket::MEDIA_TYPE_DATA:
return RtpCapabilities();
}
// Not reached; avoids compile warning.
FATAL();
}
RtpCapabilities PeerConnectionFactory::GetRtpReceiverCapabilities(
cricket::MediaType kind) const {
RTC_DCHECK_RUN_ON(signaling_thread_);
switch (kind) {
case cricket::MEDIA_TYPE_AUDIO: {
cricket::AudioCodecs cricket_codecs;
channel_manager_->GetSupportedAudioReceiveCodecs(&cricket_codecs);
return ToRtpCapabilities(
cricket_codecs,
channel_manager_->GetDefaultEnabledAudioRtpHeaderExtensions());
}
case cricket::MEDIA_TYPE_VIDEO: {
cricket::VideoCodecs cricket_codecs;
channel_manager_->GetSupportedVideoReceiveCodecs(&cricket_codecs);
return ToRtpCapabilities(
cricket_codecs,
channel_manager_->GetDefaultEnabledVideoRtpHeaderExtensions());
}
case cricket::MEDIA_TYPE_DATA:
return RtpCapabilities();
}
// Not reached; avoids compile warning.
FATAL();
}
rtc::scoped_refptr<AudioSourceInterface>
PeerConnectionFactory::CreateAudioSource(const cricket::AudioOptions& options) {
RTC_DCHECK(signaling_thread_->IsCurrent());
rtc::scoped_refptr<LocalAudioSource> source(
LocalAudioSource::Create(&options));
return source;
}
bool PeerConnectionFactory::StartAecDump(FILE* file, int64_t max_size_bytes) {
RTC_DCHECK(signaling_thread_->IsCurrent());
return channel_manager_->StartAecDump(FileWrapper(file), max_size_bytes);
}
void PeerConnectionFactory::StopAecDump() {
RTC_DCHECK(signaling_thread_->IsCurrent());
channel_manager_->StopAecDump();
}
rtc::scoped_refptr<PeerConnectionInterface>
PeerConnectionFactory::CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
std::unique_ptr<cricket::PortAllocator> allocator,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
PeerConnectionObserver* observer) {
// Convert the legacy API into the new dependency structure.
PeerConnectionDependencies dependencies(observer);
dependencies.allocator = std::move(allocator);
dependencies.cert_generator = std::move(cert_generator);
// Pass that into the new API.
return CreatePeerConnection(configuration, std::move(dependencies));
}
rtc::scoped_refptr<PeerConnectionInterface>
PeerConnectionFactory::CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies dependencies) {
RTC_DCHECK(signaling_thread_->IsCurrent());
RTC_DCHECK(!(dependencies.allocator && dependencies.packet_socket_factory))
<< "You can't set both allocator and packet_socket_factory; "
"the former is going away (see bugs.webrtc.org/7447";
// Set internal defaults if optional dependencies are not set.
if (!dependencies.cert_generator) {
dependencies.cert_generator =
std::make_unique<rtc::RTCCertificateGenerator>(signaling_thread_,
network_thread_);
}
if (!dependencies.allocator) {
rtc::PacketSocketFactory* packet_socket_factory;
if (dependencies.packet_socket_factory)
packet_socket_factory = dependencies.packet_socket_factory.get();
else
packet_socket_factory = default_socket_factory_.get();
dependencies.allocator = std::make_unique<cricket::BasicPortAllocator>(
default_network_manager_.get(), packet_socket_factory,
configuration.turn_customizer);
}
if (!dependencies.async_resolver_factory) {
dependencies.async_resolver_factory =
std::make_unique<webrtc::BasicAsyncResolverFactory>();
}
if (!dependencies.ice_transport_factory) {
dependencies.ice_transport_factory =
std::make_unique<DefaultIceTransportFactory>();
}
dependencies.allocator->SetNetworkIgnoreMask(options_.network_ignore_mask);
std::unique_ptr<RtcEventLog> event_log =
worker_thread_->Invoke<std::unique_ptr<RtcEventLog>>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnectionFactory::CreateRtcEventLog_w, this));
std::unique_ptr<Call> call = worker_thread_->Invoke<std::unique_ptr<Call>>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnectionFactory::CreateCall_w, this, event_log.get()));
rtc::scoped_refptr<PeerConnection> pc(
new rtc::RefCountedObject<PeerConnection>(this, std::move(event_log),
std::move(call)));
ActionsBeforeInitializeForTesting(pc);
if (!pc->Initialize(configuration, std::move(dependencies))) {
return nullptr;
}
return PeerConnectionProxy::Create(signaling_thread(), pc);
}
rtc::scoped_refptr<MediaStreamInterface>
PeerConnectionFactory::CreateLocalMediaStream(const std::string& stream_id) {
RTC_DCHECK(signaling_thread_->IsCurrent());
return MediaStreamProxy::Create(signaling_thread_,
MediaStream::Create(stream_id));
}
rtc::scoped_refptr<VideoTrackInterface> PeerConnectionFactory::CreateVideoTrack(
const std::string& id,
VideoTrackSourceInterface* source) {
RTC_DCHECK(signaling_thread_->IsCurrent());
rtc::scoped_refptr<VideoTrackInterface> track(
VideoTrack::Create(id, source, worker_thread_));
return VideoTrackProxy::Create(signaling_thread_, worker_thread_, track);
}
rtc::scoped_refptr<AudioTrackInterface> PeerConnectionFactory::CreateAudioTrack(
const std::string& id,
AudioSourceInterface* source) {
RTC_DCHECK(signaling_thread_->IsCurrent());
rtc::scoped_refptr<AudioTrackInterface> track(AudioTrack::Create(id, source));
return AudioTrackProxy::Create(signaling_thread_, track);
}
std::unique_ptr<cricket::SctpTransportInternalFactory>
PeerConnectionFactory::CreateSctpTransportInternalFactory() {
#ifdef HAVE_SCTP
return std::make_unique<cricket::SctpTransportFactory>(network_thread());
#else
return nullptr;
#endif
}
cricket::ChannelManager* PeerConnectionFactory::channel_manager() {
return channel_manager_.get();
}
std::unique_ptr<RtcEventLog> PeerConnectionFactory::CreateRtcEventLog_w() {
RTC_DCHECK_RUN_ON(worker_thread_);
auto encoding_type = RtcEventLog::EncodingType::Legacy;
if (IsTrialEnabled("WebRTC-RtcEventLogNewFormat"))
encoding_type = RtcEventLog::EncodingType::NewFormat;
return event_log_factory_
? event_log_factory_->CreateRtcEventLog(encoding_type)
: std::make_unique<RtcEventLogNull>();
}
std::unique_ptr<Call> PeerConnectionFactory::CreateCall_w(
RtcEventLog* event_log) {
RTC_DCHECK_RUN_ON(worker_thread_);
webrtc::Call::Config call_config(event_log);
if (!channel_manager_->media_engine() || !call_factory_) {
return nullptr;
}
call_config.audio_state =
channel_manager_->media_engine()->voice().GetAudioState();
FieldTrialParameter<DataRate> min_bandwidth("min",
DataRate::KilobitsPerSec(30));
FieldTrialParameter<DataRate> start_bandwidth("start",
DataRate::KilobitsPerSec(300));
FieldTrialParameter<DataRate> max_bandwidth("max",
DataRate::KilobitsPerSec(2000));
ParseFieldTrial({&min_bandwidth, &start_bandwidth, &max_bandwidth},
trials_->Lookup("WebRTC-PcFactoryDefaultBitrates"));
call_config.bitrate_config.min_bitrate_bps =
rtc::saturated_cast<int>(min_bandwidth->bps());
call_config.bitrate_config.start_bitrate_bps =
rtc::saturated_cast<int>(start_bandwidth->bps());
call_config.bitrate_config.max_bitrate_bps =
rtc::saturated_cast<int>(max_bandwidth->bps());
call_config.fec_controller_factory = fec_controller_factory_.get();
call_config.task_queue_factory = task_queue_factory_.get();
call_config.network_state_predictor_factory =
network_state_predictor_factory_.get();
call_config.neteq_factory = neteq_factory_.get();
if (IsTrialEnabled("WebRTC-Bwe-InjectedCongestionController")) {
RTC_LOG(LS_INFO) << "Using injected network controller factory";
call_config.network_controller_factory =
injected_network_controller_factory_.get();
} else {
RTC_LOG(LS_INFO) << "Using default network controller factory";
}
call_config.trials = trials_.get();
return std::unique_ptr<Call>(call_factory_->CreateCall(call_config));
}
bool PeerConnectionFactory::IsTrialEnabled(absl::string_view key) const {
RTC_DCHECK(trials_);
return absl::StartsWith(trials_->Lookup(key), "Enabled");
}
} // namespace webrtc