blob: 8a998481c1ec74b1d00c671f25c6ba34f60302fb [file] [log] [blame]
/*
* 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 <type_traits>
#include <utility>
#include "absl/strings/match.h"
#include "api/async_resolver_factory.h"
#include "api/call/call_factory_interface.h"
#include "api/fec_controller.h"
#include "api/ice_transport_interface.h"
#include "api/media_stream_proxy.h"
#include "api/media_stream_track_proxy.h"
#include "api/network_state_predictor.h"
#include "api/packet_socket_factory.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/bitrate_settings.h"
#include "api/units/data_rate.h"
#include "call/audio_state.h"
#include "media/base/media_engine.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/session_description.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/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/synchronization/sequence_checker.h"
#include "rtc_base/system/file_wrapper.h"
namespace webrtc {
rtc::scoped_refptr<PeerConnectionFactoryInterface>
CreateModularPeerConnectionFactory(
PeerConnectionFactoryDependencies dependencies) {
// The PeerConnectionFactory must be created on the signaling thread.
if (dependencies.signaling_thread &&
!dependencies.signaling_thread->IsCurrent()) {
return dependencies.signaling_thread
->Invoke<rtc::scoped_refptr<PeerConnectionFactoryInterface>>(
RTC_FROM_HERE, [&dependencies] {
return CreateModularPeerConnectionFactory(
std::move(dependencies));
});
}
auto pc_factory = PeerConnectionFactory::Create(std::move(dependencies));
if (!pc_factory) {
return nullptr;
}
// Verify that the invocation and the initialization ended up agreeing on the
// thread.
RTC_DCHECK_RUN_ON(pc_factory->signaling_thread());
return PeerConnectionFactoryProxy::Create(pc_factory->signaling_thread(),
pc_factory);
}
// Static
rtc::scoped_refptr<PeerConnectionFactory> PeerConnectionFactory::Create(
PeerConnectionFactoryDependencies dependencies) {
auto context = ConnectionContext::Create(&dependencies);
if (!context) {
return nullptr;
}
return new rtc::RefCountedObject<PeerConnectionFactory>(context,
&dependencies);
}
PeerConnectionFactory::PeerConnectionFactory(
rtc::scoped_refptr<ConnectionContext> context,
PeerConnectionFactoryDependencies* dependencies)
: context_(context),
task_queue_factory_(std::move(dependencies->task_queue_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)) {}
PeerConnectionFactory::PeerConnectionFactory(
PeerConnectionFactoryDependencies dependencies)
: PeerConnectionFactory(ConnectionContext::Create(&dependencies),
&dependencies) {}
PeerConnectionFactory::~PeerConnectionFactory() {
RTC_DCHECK_RUN_ON(signaling_thread());
}
void PeerConnectionFactory::SetOptions(const Options& options) {
context_->SetOptions(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();
case cricket::MEDIA_TYPE_UNSUPPORTED:
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();
case cricket::MEDIA_TYPE_UNSUPPORTED:
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 = context_->default_socket_factory();
dependencies.allocator = std::make_unique<cricket::BasicPortAllocator>(
context_->default_network_manager(), 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>(context_, 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);
}
cricket::ChannelManager* PeerConnectionFactory::channel_manager() {
return context_->channel_manager();
}
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() || !context_->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();
return std::unique_ptr<Call>(
context_->call_factory()->CreateCall(call_config));
}
bool PeerConnectionFactory::IsTrialEnabled(absl::string_view key) const {
return absl::StartsWith(trials().Lookup(key), "Enabled");
}
} // namespace webrtc