webrtc_m130/pc/peer_connection_factory.cc
Tomas Gunnarsson 41bfcf4a63 Inject network thread to Call.
This will allow for transitioning PacketReceiver callbacks and
network related callbacks from being posted over to the worker thread
and instead can stay on the network thread along with related state.

Bug: webrtc:11993
Change-Id: I38df462d4dee064015c490f2b8f809cb47f23cf1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202039
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33116}
2021-01-31 10:56:14 +00:00

364 lines
14 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 <memory>
#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/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) {
RTC_DCHECK_RUN_ON(signaling_thread());
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();
case cricket::MEDIA_TYPE_UNSUPPORTED:
return RtpCapabilities();
}
RTC_CHECK_NOTREACHED();
}
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();
}
RTC_CHECK_NOTREACHED();
}
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) {
auto result =
CreatePeerConnectionOrError(configuration, std::move(dependencies));
if (result.ok()) {
return result.MoveValue();
} else {
return nullptr;
}
}
RTCErrorOr<rtc::scoped_refptr<PeerConnectionInterface>>
PeerConnectionFactory::CreatePeerConnectionOrError(
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies dependencies) {
RTC_DCHECK_RUN_ON(signaling_thread());
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, [this] { return CreateRtcEventLog_w(); });
std::unique_ptr<Call> call = worker_thread()->Invoke<std::unique_ptr<Call>>(
RTC_FROM_HERE,
[this, &event_log] { return CreateCall_w(event_log.get()); });
auto result = PeerConnection::Create(context_, options_, std::move(event_log),
std::move(call), configuration,
std::move(dependencies));
if (!result.ok()) {
return result.MoveError();
}
rtc::scoped_refptr<PeerConnectionInterface> result_proxy =
PeerConnectionProxy::Create(signaling_thread(), result.MoveValue());
return result_proxy;
}
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, network_thread());
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