webrtc_m130/pc/peer_connection_factory.cc
Vojin Ilic 504fc192d0 Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies.
This way we can have custom implementation of RtpTransportControllerSendInterface and pass it properly to Call.
Call relies on RtpTransportControllerSendInterface already so this is natural way to customize RTP related classes.

If there is custom factory present in dependencies it will be used, otherwise default factory will be used.

Intention behind this change is to have ability to have custom QoS with custom parameters.

Bug: webrtc:12778
Change-Id: I5b88957025621ef4bcd63eaa98c218ad213da9c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217769
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@nvidia.com>
Cr-Commit-Position: refs/heads/master@{#34181}
2021-06-01 06:57:31 +00:00

353 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/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/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/ref_counted_object.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
->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->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>()) {}
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_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;
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_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 channel_manager()->StartAecDump(FileWrapper(file), max_size_bytes);
}
void PeerConnectionFactory::StopAecDump() {
RTC_DCHECK_RUN_ON(worker_thread());
channel_manager()->StopAecDump();
}
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();
}
// 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, 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();
call_config.rtp_transport_controller_send_factory =
transport_controller_send_factory_.get();
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