webrtc_m130/pc/peer_connection_factory.cc

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/*
* 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"
Revert "Revert "Enables PeerConnectionFactory using external fec controller"" This reverts commit 00733015fafbbc61ddc12dfdc88b21a9fcd9d122. Reason for revert: The reason for a downstream test failure on the original commit and a workaround has been found. Solution is to keep a PeerConnectionFactory constructor implementation as the same as before. Original change's description: > Revert "Enables PeerConnectionFactory using external fec controller" > > This reverts commit 4f07bdb25567d8ef528311e0b50a62c61d543fc3. > > Reason for revert: Speculatively reverting, because downstream test is now hitting "PeerConnectionFactory.initialize was not called before creating a PeerConnectionFactory" error, even though it did call initialize. I don't see how any change in this CL could cause that, but it's the only CL on the blamelist, and it does modify PeerConnectionFactory.java > > Original change's description: > > Enables PeerConnectionFactory using external fec controller > > > > Bug: webrtc:8799 > > Change-Id: Ieb2cf6163b9a83844ab9ed4822b4a7f1db4c24b8 > > Reviewed-on: https://webrtc-review.googlesource.com/43961 > > Commit-Queue: Ying Wang <yinwa@webrtc.org> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22038} > > TBR=sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org > > Change-Id: I95868c35d6f9973e0ebf563814cd71d0fcbd433d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8799 > Reviewed-on: https://webrtc-review.googlesource.com/54080 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22040} TBR=deadbeef@webrtc.org,sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org Bug: webrtc:8799 Change-Id: If9f3292bfcc739782967530c49f006d0abbc38a8 Reviewed-on: https://webrtc-review.googlesource.com/55400 Commit-Queue: Ying Wang <yinwa@webrtc.org> Reviewed-by: Ying Wang <yinwa@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22100}
2018-02-20 12:50:27 +01:00
#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(); });
Revert of Add logging of host lookups made by TurnPort to the RtcEventLog. (patchset #11 id:200001 of https://codereview.webrtc.org/2996933003/ ) Reason for revert: Breaks Chromium build due to the changed constructor in webrtc/p2p/client/basicportallocator.h. Build (example): https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/19739. Log: FAILED: obj/remoting/protocol/protocol/port_allocator.o /b/c/goma_client/gomacc ../../third_party/llvm-build/Release+Asserts/bin/clang++ -MMD -MF obj/remoting/protocol/protocol/port_allocator.o.d -DV8_DEPRECATION_WARNINGS -DUSE_UDEV -DUSE_AURA=1 -DUSE_PANGO=1 -DUSE_CAIRO=1 -DUSE_GLIB=1 -DUSE_NSS_CERTS=1 -DUSE_X11=1 -DFULL_SAFE_BROWSING -DSAFE_BROWSING_CSD -DSAFE_BROWSING_DB_LOCAL -DCHROMIUM_BUILD -DFIELDTRIAL_TESTING_ENABLED -DCR_CLANG_REVISION=\"310694-2\" -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_LARGEFILE64_SOURCE -D__STDC_CONSTANT_MACROS -D__STDC_FORMAT_MACROS -DCOMPONENT_BUILD -D_DEBUG -DDYNAMIC_ANNOTATIONS_ENABLED=1 -DWTF_USE_DYNAMIC_ANNOTATIONS=1 -D_GLIBCXX_DEBUG=1 -DGLIB_VERSION_MAX_ALLOWED=GLIB_VERSION_2_32 -DGLIB_VERSION_MIN_REQUIRED=GLIB_VERSION_2_26 -DEXPAT_RELATIVE_PATH -DGL_GLEXT_PROTOTYPES -DUSE_GLX -DUSE_EGL -DGOOGLE_PROTOBUF_NO_RTTI -DGOOGLE_PROTOBUF_NO_STATIC_INITIALIZER -DHAVE_PTHREAD -DPROTOBUF_USE_DLLS -DWEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=0 -DFEATURE_ENABLE_VOICEMAIL -DGTEST_RELATIVE_PATH -DWEBRTC_CHROMIUM_BUILD -DWEBRTC_POSIX -DWEBRTC_LINUX -DBORINGSSL_SHARED_LIBRARY -I../.. -Igen -I../../build/linux/debian_jessie_amd64-sysroot/usr/include/glib-2.0 -I../../build/linux/debian_jessie_amd64-sysroot/usr/lib/x86_64-linux-gnu/glib-2.0/include -I../../third_party/libwebp/src -I../../third_party/khronos -I../../gpu -I../../third_party/protobuf/src -Igen/protoc_out -I../../third_party/protobuf/src -I../../third_party/webrtc_overrides -I../../testing/gtest/include -I../../third_party -I../../third_party/webrtc_overrides -I../../third_party -I../../third_party/boringssl/src/include -I../../build/linux/debian_jessie_amd64-sysroot/usr/include/nss -I../../build/linux/debian_jessie_amd64-sysroot/usr/include/nspr -I../../third_party/libyuv/include -fno-strict-aliasing --param=ssp-buffer-size=4 -fstack-protector -Wno-builtin-macro-redefined -D__DATE__= -D__TIME__= -D__TIMESTAMP__= -funwind-tables -fPIC -pipe -B../../third_party/binutils/Linux_x64/Release/bin -pthread -fcolor-diagnostics -fdebug-prefix-map=/b/c/b/Linux_Builder__dbg_/src=. -m64 -march=x86-64 -Wall -Werror -Wextra -Wno-missing-field-initializers -Wno-unused-parameter -Wno-c++11-narrowing -Wno-covered-switch-default -Wno-unneeded-internal-declaration -Wno-inconsistent-missing-override -Wno-undefined-var-template -Wno-nonportable-include-path -Wno-address-of-packed-member -Wno-unused-lambda-capture -Wno-user-defined-warnings -Wno-enum-compare-switch -O0 -fno-omit-frame-pointer -g2 -gsplit-dwarf -fvisibility=hidden -Xclang -load -Xclang ../../third_party/llvm-build/Release+Asserts/lib/libFindBadConstructs.so -Xclang -add-plugin -Xclang find-bad-constructs -Xclang -plugin-arg-find-bad-constructs -Xclang check-auto-raw-pointer -Xclang -plugin-arg-find-bad-constructs -Xclang check-ipc -Wheader-hygiene -Wstring-conversion -Wtautological-overlap-compare -Wexit-time-destructors -Wno-header-guard -Wno-undefined-bool-conversion -Wno-tautological-undefined-compare -std=gnu++14 -fno-rtti -nostdinc++ -isystem../../buildtools/third_party/libc++/trunk/include -isystem../../buildtools/third_party/libc++abi/trunk/include --sysroot=../../build/linux/debian_jessie_amd64-sysroot -fno-exceptions -fvisibility-inlines-hidden -c ../../remoting/protocol/port_allocator.cc -o obj/remoting/protocol/protocol/port_allocator.o ../../remoting/protocol/port_allocator.cc:48:7: error: no matching constructor for initialization of 'cricket::BasicPortAllocator' : BasicPortAllocator(network_manager.get(), socket_factory.get()), ^ ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ../../third_party/webrtc/p2p/client/basicportallocator.h:35:12: note: candidate constructor not viable: requires single argument 'network_manager', but 2 arguments were provided explicit BasicPortAllocator(rtc::NetworkManager* network_manager); ^ ../../third_party/webrtc/p2p/client/basicportallocator.h:30:7: note: candidate constructor (the implicit copy constructor) not viable: requires 1 argument, but 2 were provided class BasicPortAllocator : public PortAllocator { ^ ../../third_party/webrtc/p2p/client/basicportallocator.h:32:3: note: candidate constructor not viable: requires 3 arguments, but 2 were provided BasicPortAllocator(rtc::NetworkManager* network_manager, ^ ../../third_party/webrtc/p2p/client/basicportallocator.h:36:3: note: candidate constructor not viable: requires 3 arguments, but 2 were provided BasicPortAllocator(rtc::NetworkManager* network_manager, ^ ../../third_party/webrtc/p2p/client/basicportallocator.h:39:3: note: candidate constructor not viable: requires 5 arguments, but 2 were provided BasicPortAllocator(rtc::NetworkManager* network_manager, ^ 1 error generated. Original issue's description: > Add logging host lookups made by TurnPort to the RtcEventLog. > > The following fields are logged: > - error, if there was an error. > - elapsed time in milliseconds > > BUG=webrtc:8100 > > Review-Url: https://codereview.webrtc.org/2996933003 > Cr-Commit-Position: refs/heads/master@{#19574} > Committed: https://chromium.googlesource.com/external/webrtc/+/c251cb13c08aba710ba3a12588beb4aa172c7323 TBR=terelius@webrtc.org,pthatcher@webrtc.org,jonaso@google.com,pthatcher@google.com,solenberg@webrtc.org,deadbeef@webrtc.org,jonaso@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:8100 Review-Url: https://codereview.webrtc.org/3012473002 Cr-Commit-Position: refs/heads/master@{#19578}
2017-08-29 04:49:00 -07:00
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();
}
Reland "Fix unsynchronized access to mid_to_transport_ in JsepTransportController" This reverts commit 6b143c1c0686519bc9d73223c1350cee286c8d78. Reason for revert: Relanding with updated expectations for SctpTransport::Information based on TransceiverStateSurfacer in Chromium. Original change's description: > Revert "Fix unsynchronized access to mid_to_transport_ in JsepTransportController" > > This reverts commit 6cd405850467683cf10d05028ea0f644a68a91a4. > > Reason for revert: Breaks WebRTC Chromium FYI Bots > > First failure: > https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/1925 > > Failed tests: > WebRtcDataBrowserTest.CallWithSctpDataAndMedia > WebRtcDataBrowserTest.CallWithSctpDataOnly > > Original change's description: > > Fix unsynchronized access to mid_to_transport_ in JsepTransportController > > > > * Added several thread checks to JTC to help with programmer errors. > > * Avoid a few Invokes() to the network thread here and there such > > as for fetching sctp transport name for getStats(). The transport > > name is now cached when it changes on the network thread. > > * JsepTransportController instances now get deleted on the network > > thread rather than on the signaling thread + issuing an Invoke() > > in the dtor. > > * Moved some thread hops from JTC over to PC which is where the problem > > exists and also (imho) makes it easier to see where hops happen in > > the PC code. > > * The sctp transport is now started asynchronously when we push down the > > media description. > > * PeerConnection proxy calls GetSctpTransport directly on the network > > thread instead of to the signaling thread + blocking on the network > > thread. > > * The above changes simplified things for webrtc::SctpTransport which > > allowed for removing locking from that class and delete some code. > > > > Bug: webrtc:9987, webrtc:12445 > > Change-Id: Ic89a9426e314e1b93c81751d4f732f05fa448fbc > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205620 > > Commit-Queue: Tommi <tommi@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33191} > > TBR=tommi@webrtc.org,hta@webrtc.org > > Change-Id: I7b2913d5133807589461105cf07eff3e9bb7157e > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9987 > Bug: webrtc:12445 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206466 > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Commit-Queue: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33204} TBR=tommi@webrtc.org,hta@webrtc.org,guidou@webrtc.org # Not skipping CQ checks because this is a reland. Bug: webrtc:9987 Bug: webrtc:12445 Change-Id: Icb205cbac493ed3b881d71ea3af4fb9018701bf4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206560 Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33219}
2021-02-10 13:05:44 +01:00
// 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_|
Reland "Fix unsynchronized access to mid_to_transport_ in JsepTransportController" This reverts commit 6b143c1c0686519bc9d73223c1350cee286c8d78. Reason for revert: Relanding with updated expectations for SctpTransport::Information based on TransceiverStateSurfacer in Chromium. Original change's description: > Revert "Fix unsynchronized access to mid_to_transport_ in JsepTransportController" > > This reverts commit 6cd405850467683cf10d05028ea0f644a68a91a4. > > Reason for revert: Breaks WebRTC Chromium FYI Bots > > First failure: > https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/1925 > > Failed tests: > WebRtcDataBrowserTest.CallWithSctpDataAndMedia > WebRtcDataBrowserTest.CallWithSctpDataOnly > > Original change's description: > > Fix unsynchronized access to mid_to_transport_ in JsepTransportController > > > > * Added several thread checks to JTC to help with programmer errors. > > * Avoid a few Invokes() to the network thread here and there such > > as for fetching sctp transport name for getStats(). The transport > > name is now cached when it changes on the network thread. > > * JsepTransportController instances now get deleted on the network > > thread rather than on the signaling thread + issuing an Invoke() > > in the dtor. > > * Moved some thread hops from JTC over to PC which is where the problem > > exists and also (imho) makes it easier to see where hops happen in > > the PC code. > > * The sctp transport is now started asynchronously when we push down the > > media description. > > * PeerConnection proxy calls GetSctpTransport directly on the network > > thread instead of to the signaling thread + blocking on the network > > thread. > > * The above changes simplified things for webrtc::SctpTransport which > > allowed for removing locking from that class and delete some code. > > > > Bug: webrtc:9987, webrtc:12445 > > Change-Id: Ic89a9426e314e1b93c81751d4f732f05fa448fbc > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205620 > > Commit-Queue: Tommi <tommi@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33191} > > TBR=tommi@webrtc.org,hta@webrtc.org > > Change-Id: I7b2913d5133807589461105cf07eff3e9bb7157e > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9987 > Bug: webrtc:12445 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206466 > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Commit-Queue: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33204} TBR=tommi@webrtc.org,hta@webrtc.org,guidou@webrtc.org # Not skipping CQ checks because this is a reland. Bug: webrtc:9987 Bug: webrtc:12445 Change-Id: Icb205cbac493ed3b881d71ea3af4fb9018701bf4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206560 Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33219}
2021-02-10 13:05:44 +01:00
// 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 =
Reland "Fix unsynchronized access to mid_to_transport_ in JsepTransportController" This reverts commit 6b143c1c0686519bc9d73223c1350cee286c8d78. Reason for revert: Relanding with updated expectations for SctpTransport::Information based on TransceiverStateSurfacer in Chromium. Original change's description: > Revert "Fix unsynchronized access to mid_to_transport_ in JsepTransportController" > > This reverts commit 6cd405850467683cf10d05028ea0f644a68a91a4. > > Reason for revert: Breaks WebRTC Chromium FYI Bots > > First failure: > https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/1925 > > Failed tests: > WebRtcDataBrowserTest.CallWithSctpDataAndMedia > WebRtcDataBrowserTest.CallWithSctpDataOnly > > Original change's description: > > Fix unsynchronized access to mid_to_transport_ in JsepTransportController > > > > * Added several thread checks to JTC to help with programmer errors. > > * Avoid a few Invokes() to the network thread here and there such > > as for fetching sctp transport name for getStats(). The transport > > name is now cached when it changes on the network thread. > > * JsepTransportController instances now get deleted on the network > > thread rather than on the signaling thread + issuing an Invoke() > > in the dtor. > > * Moved some thread hops from JTC over to PC which is where the problem > > exists and also (imho) makes it easier to see where hops happen in > > the PC code. > > * The sctp transport is now started asynchronously when we push down the > > media description. > > * PeerConnection proxy calls GetSctpTransport directly on the network > > thread instead of to the signaling thread + blocking on the network > > thread. > > * The above changes simplified things for webrtc::SctpTransport which > > allowed for removing locking from that class and delete some code. > > > > Bug: webrtc:9987, webrtc:12445 > > Change-Id: Ic89a9426e314e1b93c81751d4f732f05fa448fbc > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205620 > > Commit-Queue: Tommi <tommi@webrtc.org> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33191} > > TBR=tommi@webrtc.org,hta@webrtc.org > > Change-Id: I7b2913d5133807589461105cf07eff3e9bb7157e > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:9987 > Bug: webrtc:12445 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206466 > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Commit-Queue: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33204} TBR=tommi@webrtc.org,hta@webrtc.org,guidou@webrtc.org # Not skipping CQ checks because this is a reland. Bug: webrtc:9987 Bug: webrtc:12445 Change-Id: Icb205cbac493ed3b881d71ea3af4fb9018701bf4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206560 Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33219}
2021-02-10 13:05:44 +01:00
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)
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 17:06:18 +02:00
: 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());
Revert "Revert "Enables PeerConnectionFactory using external fec controller"" This reverts commit 00733015fafbbc61ddc12dfdc88b21a9fcd9d122. Reason for revert: The reason for a downstream test failure on the original commit and a workaround has been found. Solution is to keep a PeerConnectionFactory constructor implementation as the same as before. Original change's description: > Revert "Enables PeerConnectionFactory using external fec controller" > > This reverts commit 4f07bdb25567d8ef528311e0b50a62c61d543fc3. > > Reason for revert: Speculatively reverting, because downstream test is now hitting "PeerConnectionFactory.initialize was not called before creating a PeerConnectionFactory" error, even though it did call initialize. I don't see how any change in this CL could cause that, but it's the only CL on the blamelist, and it does modify PeerConnectionFactory.java > > Original change's description: > > Enables PeerConnectionFactory using external fec controller > > > > Bug: webrtc:8799 > > Change-Id: Ieb2cf6163b9a83844ab9ed4822b4a7f1db4c24b8 > > Reviewed-on: https://webrtc-review.googlesource.com/43961 > > Commit-Queue: Ying Wang <yinwa@webrtc.org> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22038} > > TBR=sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org > > Change-Id: I95868c35d6f9973e0ebf563814cd71d0fcbd433d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8799 > Reviewed-on: https://webrtc-review.googlesource.com/54080 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22040} TBR=deadbeef@webrtc.org,sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org Bug: webrtc:8799 Change-Id: If9f3292bfcc739782967530c49f006d0abbc38a8 Reviewed-on: https://webrtc-review.googlesource.com/55400 Commit-Queue: Ying Wang <yinwa@webrtc.org> Reviewed-by: Ying Wang <yinwa@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22100}
2018-02-20 12:50:27 +01:00
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();
Revert "Revert "Enables PeerConnectionFactory using external fec controller"" This reverts commit 00733015fafbbc61ddc12dfdc88b21a9fcd9d122. Reason for revert: The reason for a downstream test failure on the original commit and a workaround has been found. Solution is to keep a PeerConnectionFactory constructor implementation as the same as before. Original change's description: > Revert "Enables PeerConnectionFactory using external fec controller" > > This reverts commit 4f07bdb25567d8ef528311e0b50a62c61d543fc3. > > Reason for revert: Speculatively reverting, because downstream test is now hitting "PeerConnectionFactory.initialize was not called before creating a PeerConnectionFactory" error, even though it did call initialize. I don't see how any change in this CL could cause that, but it's the only CL on the blamelist, and it does modify PeerConnectionFactory.java > > Original change's description: > > Enables PeerConnectionFactory using external fec controller > > > > Bug: webrtc:8799 > > Change-Id: Ieb2cf6163b9a83844ab9ed4822b4a7f1db4c24b8 > > Reviewed-on: https://webrtc-review.googlesource.com/43961 > > Commit-Queue: Ying Wang <yinwa@webrtc.org> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22038} > > TBR=sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org > > Change-Id: I95868c35d6f9973e0ebf563814cd71d0fcbd433d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8799 > Reviewed-on: https://webrtc-review.googlesource.com/54080 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22040} TBR=deadbeef@webrtc.org,sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org Bug: webrtc:8799 Change-Id: If9f3292bfcc739782967530c49f006d0abbc38a8 Reviewed-on: https://webrtc-review.googlesource.com/55400 Commit-Queue: Ying Wang <yinwa@webrtc.org> Reviewed-by: Ying Wang <yinwa@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22100}
2018-02-20 12:50:27 +01:00
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