webrtc_m130/pc/peer_connection_factory.cc
Henrik Boström cf2856b01c Add parameter to control the pacer's burst outside of field trials.
BurstyPacer is currently controlled via field trials. In order for
Chrome to be able to have burst without relying on a field trial, this
parameter is added.

When all burst experiments have concluded we may be able to have a
hardcoded constant instead, but for now the parameter is added to
RTCConfiguration.

NOTRY=True

Bug: chromium:1354491
Change-Id: I386c1651dbbcbf309c15ea3d3380cf8f632b5429
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283420
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38621}
2022-11-15 08:46:30 +00:00

361 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 <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/network_state_predictor.h"
#include "api/packet_socket_factory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/sequence_checker.h"
#include "api/transport/bitrate_settings.h"
#include "api/units/data_rate.h"
#include "call/audio_state.h"
#include "call/rtp_transport_controller_send_factory.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/base/port_allocator.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/media_stream_proxy.h"
#include "pc/media_stream_track_proxy.h"
#include "pc/peer_connection.h"
#include "pc/peer_connection_factory_proxy.h"
#include "pc/peer_connection_proxy.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/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/rtc_certificate_generator.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->BlockingCall([&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->worker_thread(), pc_factory);
}
// Static
rtc::scoped_refptr<PeerConnectionFactory> PeerConnectionFactory::Create(
PeerConnectionFactoryDependencies dependencies) {
auto context = ConnectionContext::Create(&dependencies);
if (!context) {
return nullptr;
}
return rtc::make_ref_counted<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)),
transport_controller_send_factory_(
(dependencies->transport_controller_send_factory)
? std::move(dependencies->transport_controller_send_factory)
: std::make_unique<RtpTransportControllerSendFactory>()),
metronome_(std::move(dependencies->metronome)) {}
PeerConnectionFactory::PeerConnectionFactory(
PeerConnectionFactoryDependencies dependencies)
: PeerConnectionFactory(ConnectionContext::Create(&dependencies),
&dependencies) {}
PeerConnectionFactory::~PeerConnectionFactory() {
RTC_DCHECK_RUN_ON(signaling_thread());
worker_thread()->BlockingCall([this] {
RTC_DCHECK_RUN_ON(worker_thread());
metronome_ = nullptr;
});
}
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;
cricket_codecs = media_engine()->voice().send_codecs();
auto extensions =
GetDefaultEnabledRtpHeaderExtensions(media_engine()->voice());
return ToRtpCapabilities(cricket_codecs, extensions);
}
case cricket::MEDIA_TYPE_VIDEO: {
cricket::VideoCodecs cricket_codecs;
cricket_codecs = media_engine()->video().send_codecs();
auto extensions =
GetDefaultEnabledRtpHeaderExtensions(media_engine()->video());
return ToRtpCapabilities(cricket_codecs, extensions);
}
case cricket::MEDIA_TYPE_DATA:
return RtpCapabilities();
case cricket::MEDIA_TYPE_UNSUPPORTED:
return RtpCapabilities();
}
RTC_DLOG(LS_ERROR) << "Got unexpected MediaType " << kind;
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;
cricket_codecs = media_engine()->voice().recv_codecs();
auto extensions =
GetDefaultEnabledRtpHeaderExtensions(media_engine()->voice());
return ToRtpCapabilities(cricket_codecs, extensions);
}
case cricket::MEDIA_TYPE_VIDEO: {
cricket::VideoCodecs cricket_codecs =
media_engine()->video().recv_codecs(context_->use_rtx());
auto extensions =
GetDefaultEnabledRtpHeaderExtensions(media_engine()->video());
return ToRtpCapabilities(cricket_codecs, extensions);
}
case cricket::MEDIA_TYPE_DATA:
return RtpCapabilities();
case cricket::MEDIA_TYPE_UNSUPPORTED:
return RtpCapabilities();
}
RTC_DLOG(LS_ERROR) << "Got unexpected MediaType " << kind;
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_RUN_ON(worker_thread());
return media_engine()->voice().StartAecDump(FileWrapper(file),
max_size_bytes);
}
void PeerConnectionFactory::StopAecDump() {
RTC_DCHECK_RUN_ON(worker_thread());
media_engine()->voice().StopAecDump();
}
cricket::MediaEngineInterface* PeerConnectionFactory::media_engine() const {
RTC_DCHECK(context_);
return context_->media_engine();
}
RTCErrorOr<rtc::scoped_refptr<PeerConnectionInterface>>
PeerConnectionFactory::CreatePeerConnectionOrError(
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies dependencies) {
RTC_DCHECK_RUN_ON(signaling_thread());
// 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) {
const FieldTrialsView* trials =
dependencies.trials ? dependencies.trials.get() : &field_trials();
dependencies.allocator = std::make_unique<cricket::BasicPortAllocator>(
context_->default_network_manager(), context_->default_socket_factory(),
configuration.turn_customizer, /*relay_port_factory=*/nullptr, trials);
dependencies.allocator->SetPortRange(
configuration.port_allocator_config.min_port,
configuration.port_allocator_config.max_port);
dependencies.allocator->set_flags(
configuration.port_allocator_config.flags);
}
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);
dependencies.allocator->SetVpnList(configuration.vpn_list);
std::unique_ptr<RtcEventLog> event_log =
worker_thread()->BlockingCall([this] { return CreateRtcEventLog_w(); });
const FieldTrialsView* trials =
dependencies.trials ? dependencies.trials.get() : &field_trials();
std::unique_ptr<Call> call = worker_thread()->BlockingCall(
[this, &event_log, trials,
pacer_burst_interval = configuration.pacer_burst_interval] {
return CreateCall_w(event_log.get(), *trials, pacer_burst_interval);
});
auto result = PeerConnection::Create(context_, options_, std::move(event_log),
std::move(call), configuration,
std::move(dependencies));
if (!result.ok()) {
return result.MoveError();
}
// We configure the proxy with a pointer to the network thread for methods
// that need to be invoked there rather than on the signaling thread.
// Internally, the proxy object has a member variable named `worker_thread_`
// which will point to the network thread (and not the factory's
// worker_thread()). All such methods have thread checks though, so the code
// should still be clear (outside of macro expansion).
rtc::scoped_refptr<PeerConnectionInterface> result_proxy =
PeerConnectionProxy::Create(signaling_thread(), network_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, rtc::scoped_refptr<VideoTrackSourceInterface>(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, rtc::scoped_refptr<AudioSourceInterface>(source));
return AudioTrackProxy::Create(signaling_thread(), track);
}
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_->Create(encoding_type)
: std::make_unique<RtcEventLogNull>();
}
std::unique_ptr<Call> PeerConnectionFactory::CreateCall_w(
RtcEventLog* event_log,
const FieldTrialsView& field_trials,
absl::optional<TimeDelta> pacer_burst_interval) {
RTC_DCHECK_RUN_ON(worker_thread());
webrtc::Call::Config call_config(event_log, network_thread());
if (!media_engine() || !context_->call_factory()) {
return nullptr;
}
call_config.audio_state = 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},
field_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 = &field_trials;
call_config.rtp_transport_controller_send_factory =
transport_controller_send_factory_.get();
call_config.metronome = metronome_.get();
call_config.pacer_burst_interval = pacer_burst_interval;
return std::unique_ptr<Call>(
context_->call_factory()->CreateCall(call_config));
}
bool PeerConnectionFactory::IsTrialEnabled(absl::string_view key) const {
return absl::StartsWith(field_trials().Lookup(key), "Enabled");
}
} // namespace webrtc