webrtc_m130/pc/peer_connection.h

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/*
* Copyright 2012 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.
*/
#ifndef PC_PEER_CONNECTION_H_
#define PC_PEER_CONNECTION_H_
#include <stdint.h>
#include <functional>
#include <map>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "absl/types/optional.h"
#include "api/adaptation/resource.h"
Reland "Use the new DNS resolver API in PeerConnection" This reverts commit 5a40b3710545edfd8a634341df3de26f57d79281. Reason for revert: Fixed the bug and ran layout tests. Original change's description: > Revert "Use the new DNS resolver API in PeerConnection" > > This reverts commit acf8ccb3c9f001b0ed749aca52b2d436d66f9586. > > Reason for revert: Speculative revert for https://ci.chromium.org/ui/p/chromium/builders/try/win10_chromium_x64_rel_ng/b8851745102358680592/overview. > > Original change's description: > > Use the new DNS resolver API in PeerConnection > > > > Bug: webrtc:12598 > > Change-Id: I5a14058e7f28c993ed927749df7357c715ba83fb > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212961 > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33561} > > # Not skipping CQ checks because original CL landed > 1 day ago. > > TBR=hta@webrtc.org > > Bug: webrtc:12598 > Change-Id: Idc9853cb569849c49052f9cbd865614710fff979 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213188 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33591} # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:12598 Change-Id: Ief7867f2f23de66504877cdab1b23a11df2d5de4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214120 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33647}
2021-04-08 07:25:04 +00:00
#include "api/async_dns_resolver.h"
#include "api/async_resolver_factory.h"
#include "api/audio_options.h"
#include "api/candidate.h"
#include "api/crypto/crypto_options.h"
#include "api/data_channel_interface.h"
#include "api/dtls_transport_interface.h"
#include "api/ice_transport_interface.h"
#include "api/jsep.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/packet_socket_factory.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_error.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtc_event_log_output.h"
#include "api/rtp_parameters.h"
#include "api/rtp_receiver_interface.h"
#include "api/rtp_sender_interface.h"
#include "api/rtp_transceiver_interface.h"
#include "api/scoped_refptr.h"
#include "api/sctp_transport_interface.h"
#include "api/sequence_checker.h"
#include "api/set_local_description_observer_interface.h"
#include "api/set_remote_description_observer_interface.h"
#include "api/stats/rtc_stats_collector_callback.h"
#include "api/transport/bitrate_settings.h"
#include "api/transport/data_channel_transport_interface.h"
#include "api/transport/enums.h"
#include "api/turn_customizer.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "call/call.h"
#include "media/base/media_channel.h"
#include "media/base/media_engine.h"
#include "p2p/base/ice_transport_internal.h"
#include "p2p/base/port.h"
#include "p2p/base/port_allocator.h"
#include "p2p/base/transport_description.h"
#include "pc/channel.h"
#include "pc/channel_interface.h"
#include "pc/channel_manager.h"
#include "pc/connection_context.h"
#include "pc/data_channel_controller.h"
#include "pc/data_channel_utils.h"
#include "pc/dtls_transport.h"
#include "pc/jsep_transport_controller.h"
#include "pc/peer_connection_internal.h"
#include "pc/peer_connection_message_handler.h"
#include "pc/rtc_stats_collector.h"
#include "pc/rtp_receiver.h"
Add RtpSenderInterface.SetStreams This is a reland of df5731e44d510e9f23a35b77e9e102eb41919bf4 with fixes to avoid existing chromium tests to fail. Instead of replacing the existing RtpSender::set_stream_ids() to also fire OnRenegotiationNeeded(), this CL keeps the old set_stream_ids() and adds the new RtpSender::SetStreams() which sets the stream IDs and fires the callback. This allows existing callsites to maintain behavior, and reserve SetStreams() for the cases when we want OnRenegotiationNeeded() to fire. Using the SetStreams() name instead of SetStreamIDs() to match the W3C spec and to make it more different that the existing set_stream_ids(). Original change's description: > Improve spec compliance of SetStreamIDs in RtpSenderInterface > > This CL makes RtpSender::SetStreamIDs fire fire negotiationneeded > event if needed and exposes the method on RtpSenderInterface. > > This is a spec-compliance change. > > Bug: webrtc:10129 > Change-Id: I2b98b92665c847102838b094516a79b24de0e47e > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135121 > Commit-Queue: Guido Urdaneta <guidou@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27974} Bug: webrtc:10129 Change-Id: Ic0b322bfa25c157e3a39465ef8b486f898eaf6bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137439 Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27992}
2019-05-20 19:31:53 +02:00
#include "pc/rtp_sender.h"
#include "pc/rtp_transceiver.h"
#include "pc/rtp_transmission_manager.h"
#include "pc/rtp_transport_internal.h"
#include "pc/sctp_data_channel.h"
#include "pc/sctp_transport.h"
#include "pc/sdp_offer_answer.h"
#include "pc/session_description.h"
#include "pc/stats_collector.h"
#include "pc/stream_collection.h"
#include "pc/transceiver_list.h"
#include "pc/transport_stats.h"
#include "pc/usage_pattern.h"
#include "rtc_base/checks.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/rtc_certificate.h"
#include "rtc_base/ssl_certificate.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/task_utils/pending_task_safety_flag.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/unique_id_generator.h"
#include "rtc_base/weak_ptr.h"
namespace webrtc {
// PeerConnection is the implementation of the PeerConnection object as defined
// by the PeerConnectionInterface API surface.
// The class currently is solely responsible for the following:
// - Managing the session state machine (signaling state).
// - Creating and initializing lower-level objects, like PortAllocator and
// BaseChannels.
// - Owning and managing the life cycle of the RtpSender/RtpReceiver and track
// objects.
// - Tracking the current and pending local/remote session descriptions.
// The class currently is jointly responsible for the following:
// - Parsing and interpreting SDP.
// - Generating offers and answers based on the current state.
// - The ICE state machine.
// - Generating stats.
class PeerConnection : public PeerConnectionInternal,
public JsepTransportController::Observer,
public sigslot::has_slots<> {
public:
// Creates a PeerConnection and initializes it with the given values.
// If the initialization fails, the function releases the PeerConnection
// and returns nullptr.
//
// Note that the function takes ownership of dependencies, and will
// either use them or release them, whether it succeeds or fails.
static RTCErrorOr<rtc::scoped_refptr<PeerConnection>> Create(
rtc::scoped_refptr<ConnectionContext> context,
const PeerConnectionFactoryInterface::Options& options,
std::unique_ptr<RtcEventLog> event_log,
std::unique_ptr<Call> call,
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies dependencies);
rtc::scoped_refptr<StreamCollectionInterface> local_streams() override;
rtc::scoped_refptr<StreamCollectionInterface> remote_streams() override;
bool AddStream(MediaStreamInterface* local_stream) override;
void RemoveStream(MediaStreamInterface* local_stream) override;
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrack(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids) override;
bool RemoveTrack(RtpSenderInterface* sender) override;
RTCError RemoveTrackNew(
rtc::scoped_refptr<RtpSenderInterface> sender) override;
Reland "Add AddTransceiver and GetTransceivers to PeerConnection" This reverts commit 8b13f96e2d4b0449e54a3665121a4302ceb56e80. Original change's description: > Revert "Add AddTransceiver and GetTransceivers to PeerConnection" > > This reverts commit f93d2800d9b0d5818a5a383def0aaef3d441df3a. > > Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout > > Original change's description: > > Add AddTransceiver and GetTransceivers to PeerConnection > > > > WebRTC 1.0 has added the transceiver API to PeerConnection. This > > is the first step towards exposing this to WebRTC consumers. For > > now, transceivers can be added and fetched but there is not yet > > support for creating offers/answers or setting local/remote > > descriptions. That support ("Unified Plan") will be added in > > follow-up CLs. > > > > The transceiver API is currently only available if the application > > opts in by specifying the kUnifiedPlan SDP semantics when creating > > the PeerConnection. > > > > Bug: webrtc:7600 > > Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e > > Reviewed-on: https://webrtc-review.googlesource.com/23880 > > Commit-Queue: Steve Anton <steveanton@webrtc.org> > > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20896} > > TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org > > Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7600 > Reviewed-on: https://webrtc-review.googlesource.com/26400 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20897} TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org Change-Id: I19fdf08c54f09302794e998a0ffddb82ae0d7b41 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7600 Reviewed-on: https://webrtc-review.googlesource.com/26401 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20898}
2017-11-27 13:01:52 -08:00
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track) override;
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init) override;
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
cricket::MediaType media_type) override;
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
cricket::MediaType media_type,
const RtpTransceiverInit& init) override;
rtc::scoped_refptr<RtpSenderInterface> CreateSender(
const std::string& kind,
const std::string& stream_id) override;
std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
const override;
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
const override;
Reland "Add AddTransceiver and GetTransceivers to PeerConnection" This reverts commit 8b13f96e2d4b0449e54a3665121a4302ceb56e80. Original change's description: > Revert "Add AddTransceiver and GetTransceivers to PeerConnection" > > This reverts commit f93d2800d9b0d5818a5a383def0aaef3d441df3a. > > Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout > > Original change's description: > > Add AddTransceiver and GetTransceivers to PeerConnection > > > > WebRTC 1.0 has added the transceiver API to PeerConnection. This > > is the first step towards exposing this to WebRTC consumers. For > > now, transceivers can be added and fetched but there is not yet > > support for creating offers/answers or setting local/remote > > descriptions. That support ("Unified Plan") will be added in > > follow-up CLs. > > > > The transceiver API is currently only available if the application > > opts in by specifying the kUnifiedPlan SDP semantics when creating > > the PeerConnection. > > > > Bug: webrtc:7600 > > Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e > > Reviewed-on: https://webrtc-review.googlesource.com/23880 > > Commit-Queue: Steve Anton <steveanton@webrtc.org> > > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20896} > > TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org > > Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7600 > Reviewed-on: https://webrtc-review.googlesource.com/26400 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20897} TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org Change-Id: I19fdf08c54f09302794e998a0ffddb82ae0d7b41 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7600 Reviewed-on: https://webrtc-review.googlesource.com/26401 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20898}
2017-11-27 13:01:52 -08:00
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> GetTransceivers()
const override;
RTCErrorOr<rtc::scoped_refptr<DataChannelInterface>> CreateDataChannelOrError(
const std::string& label,
const DataChannelInit* config) override;
2018-03-20 13:24:20 +01:00
// WARNING: LEGACY. See peerconnectioninterface.h
bool GetStats(StatsObserver* observer,
webrtc::MediaStreamTrackInterface* track,
StatsOutputLevel level) override;
2018-03-20 13:24:20 +01:00
// Spec-complaint GetStats(). See peerconnectioninterface.h
void GetStats(RTCStatsCollectorCallback* callback) override;
2018-03-20 13:24:20 +01:00
void GetStats(
rtc::scoped_refptr<RtpSenderInterface> selector,
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) override;
void GetStats(
rtc::scoped_refptr<RtpReceiverInterface> selector,
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) override;
void ClearStatsCache() override;
SignalingState signaling_state() override;
IceConnectionState ice_connection_state() override;
IceConnectionState standardized_ice_connection_state() override;
PeerConnectionState peer_connection_state() override;
IceGatheringState ice_gathering_state() override;
Reland "Expose can_trickle_ice_candidates on PeerConnection" This reverts commit cb8c40138ca170f841bc45fa6771cdfc4b966e5f. Reason for revert: Added missing default. Original change's description: > Revert "Expose can_trickle_ice_candidates on PeerConnection" > > This reverts commit c6a65c8866487c6adc0a7bb472d3bad9389501f9. > > Reason for revert: Breaks downstream due to missing default > > Original change's description: > > Expose can_trickle_ice_candidates on PeerConnection > > > > Bug: chromium:708484 > > Change-Id: I9a40e75066341f0d9f965bd3718bfcb3f0459533 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169450 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Taylor <deadbeef@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#30653} > > TBR=deadbeef@webrtc.org,hta@webrtc.org > > Change-Id: Iaa5b977c4237715a8a5127cf167cf6512a3f7059 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: chromium:708484 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169540 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30655} TBR=deadbeef@webrtc.org,hta@webrtc.org Change-Id: I608da7781f158b4b02dd226d4dcd5615c4935fa8 Bug: chromium:708484 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169541 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30656}
2020-03-02 11:20:00 +01:00
absl::optional<bool> can_trickle_ice_candidates() override;
const SessionDescriptionInterface* local_description() const override;
const SessionDescriptionInterface* remote_description() const override;
const SessionDescriptionInterface* current_local_description() const override;
const SessionDescriptionInterface* current_remote_description()
const override;
const SessionDescriptionInterface* pending_local_description() const override;
const SessionDescriptionInterface* pending_remote_description()
const override;
void RestartIce() override;
// JSEP01
void CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) override;
void CreateAnswer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) override;
Reland "[Perfect Negotiation] Implement non-racy version of SetLocalDescription." This is a reland of d4089cae47334a4228b69d6bb23f2e49ebb7496e with the following fix: Invoke MaybeStartGathering as the last step of DoSetLocalDescription. This ensures that candidates and onicegatheringstatechange does not happen before SLD is resolved. This is important for passing external/wpt/webrtc/RTCPeerConnection-iceGatheringState.html. Original change's description: > [Perfect Negotiation] Implement non-racy version of SetLocalDescription. > > BACKGROUND > > When SLD is invoked with SetSessionDescriptionObserver, the observer is > called by posting a message back to the execution thread, delaying the > call. This delay is "artificial" - it's not necessary; the operation is > already complete. It's a post from the signaling thread to the signaling > thread. The rationale for the post was to avoid the observer making > recursive calls back into the PeerConnection. The problem with this is > that by the time the observer is called, the PeerConnection could > already have executed other operations and modified its states. > > This causes the referenced bug: one can have a race where SLD is > resolved "too late" (after a pending SRD is executed) and the signaling > state observed when SLD resolves doesn't make sense. > > When implementing Unified Plan, we fixed similar issues for SRD by > adding a version that takes SetRemoteDescriptionObserverInterface as > argument instead of SetSessionDescriptionObserver. The new version did > not have the delay. The old version had to be kept around not to break > downstream projects that had dependencies both on he delay and on > allowing the PC to be destroyed midst-operation without informing its > observers. > > THIS CL > > This does the old SRD fix for SLD as well: A new observer interface is > added, SetLocalDescriptionObserverInterface, and > PeerConnection::SetLocalDescription() is overloaded. If you call it with > the old observer, you get the delay, but if you call it with the new > observer, you don't get a delay. > > - SetLocalDescriptionObserverInterface is added. > - SetLocalDescription is overloaded. > - The adapter for SetSessionDescriptionObserver that causes the delay > previously only used for SRD is updated to handle both SLD and SRD. > - FakeSetLocalDescriptionObserver is added and > MockSetRemoteDescriptionObserver is renamed "Fake...". > > Bug: chromium:1071733 > Change-Id: I920368e648bede481058ac22f5b8794752a220b3 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179100 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31798} TBR=hta@webrtc.org Bug: chromium:1071733 Change-Id: Ic6e8d96afa1c19604762f373716c08dbfa9d178c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180481 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31804}
2020-07-29 12:04:00 +02:00
void SetLocalDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer)
override;
void SetLocalDescription(
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer)
override;
// TODO(https://crbug.com/webrtc/11798): Delete these methods in favor of the
// ones taking SetLocalDescriptionObserverInterface as argument.
void SetLocalDescription(SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) override;
void SetLocalDescription(SetSessionDescriptionObserver* observer) override;
Reland "[Perfect Negotiation] Implement non-racy version of SetLocalDescription." This is a reland of d4089cae47334a4228b69d6bb23f2e49ebb7496e with the following fix: Invoke MaybeStartGathering as the last step of DoSetLocalDescription. This ensures that candidates and onicegatheringstatechange does not happen before SLD is resolved. This is important for passing external/wpt/webrtc/RTCPeerConnection-iceGatheringState.html. Original change's description: > [Perfect Negotiation] Implement non-racy version of SetLocalDescription. > > BACKGROUND > > When SLD is invoked with SetSessionDescriptionObserver, the observer is > called by posting a message back to the execution thread, delaying the > call. This delay is "artificial" - it's not necessary; the operation is > already complete. It's a post from the signaling thread to the signaling > thread. The rationale for the post was to avoid the observer making > recursive calls back into the PeerConnection. The problem with this is > that by the time the observer is called, the PeerConnection could > already have executed other operations and modified its states. > > This causes the referenced bug: one can have a race where SLD is > resolved "too late" (after a pending SRD is executed) and the signaling > state observed when SLD resolves doesn't make sense. > > When implementing Unified Plan, we fixed similar issues for SRD by > adding a version that takes SetRemoteDescriptionObserverInterface as > argument instead of SetSessionDescriptionObserver. The new version did > not have the delay. The old version had to be kept around not to break > downstream projects that had dependencies both on he delay and on > allowing the PC to be destroyed midst-operation without informing its > observers. > > THIS CL > > This does the old SRD fix for SLD as well: A new observer interface is > added, SetLocalDescriptionObserverInterface, and > PeerConnection::SetLocalDescription() is overloaded. If you call it with > the old observer, you get the delay, but if you call it with the new > observer, you don't get a delay. > > - SetLocalDescriptionObserverInterface is added. > - SetLocalDescription is overloaded. > - The adapter for SetSessionDescriptionObserver that causes the delay > previously only used for SRD is updated to handle both SLD and SRD. > - FakeSetLocalDescriptionObserver is added and > MockSetRemoteDescriptionObserver is renamed "Fake...". > > Bug: chromium:1071733 > Change-Id: I920368e648bede481058ac22f5b8794752a220b3 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179100 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31798} TBR=hta@webrtc.org Bug: chromium:1071733 Change-Id: Ic6e8d96afa1c19604762f373716c08dbfa9d178c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180481 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31804}
2020-07-29 12:04:00 +02:00
Reland "SetRemoteDescriptionObserverInterface added." Description for changes from the original CL: Calling legacy SRD, implemented using SetRemoteDescriptionObserverAdapter wrapping the old observer, was meant to have the exact same behavior as the legacy SRD implementation which invokes the callbacks in a Post. However, in the CL that landed and got reverted (PS1), the Adapter had its own message handler, and callbacks would be invoked even if the PC was destroyed. In PS2 I've changed the Adapter to use the PeerConnection's message handler. If the PC is destroyed, the callback will not be invoked. This gives identical behavior to before this CL, and the legacy behavior is covered by a new unittest. I changed the adapter to be an implementation detail of peerconnection.cc, therefor some stuff was moved, and the only tests covering this is now in peerconnection_rtp_unittest.cc. This is a reland of 6c7ec32bd63ab2b45d4d83ae1de817ee946b4d72 Original change's description: > SetRemoteDescriptionObserverInterface added. > > The new observer replaced SetSessionDescriptionObserver for > SetRemoteDescription. Unlike SetSessionDescriptionObserver, > SetRemoteDescriptionObserverInterface is invoked synchronously so > that the you can rely on the state of the PeerConnection to represent > the result of the SetRemoteDescription call in the callback. > > The new observer succeeds or fails with an RTCError. > > This deprecates the need for PeerConnectionObserver::OnAdd/RemoveTrack > and SetSessionDescriptionObserver, with the benefit that all media > object changes can be processed in a single callback by the application > in a synchronous callback. This will help Chromium keep objects in-sync > across layers and threads in a non-racy and straight-forward way, see > design doc (Proposal 2): > https://docs.google.com/a/google.com/document/d/1-cDDC82mgU5zrHacfFz720p3xwRtuBkOPSRchh07Ho0/edit?usp=sharing > > An adapter for SetSessionDescriptionObserver is added to allow calling > the old SetRemoteDescription signature and get the old behavior > (OnSuccess/OnFailure callback in a Post) until third parties switch. > > Bug: webrtc:8473 > Change-Id: I3d4eb60da6dd34615f2c9f384aeaf4634e648c99 > Reviewed-on: https://webrtc-review.googlesource.com/17523 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20841} TBR=pthatcher@webrtc.org Bug: webrtc:8473 Change-Id: If2b1a1929663b0e77fcc9c2ebeef043e6f73adf5 Reviewed-on: https://webrtc-review.googlesource.com/25640 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20854}
2017-11-23 17:48:32 +01:00
void SetRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer)
override;
Reland "[Perfect Negotiation] Implement non-racy version of SetLocalDescription." This is a reland of d4089cae47334a4228b69d6bb23f2e49ebb7496e with the following fix: Invoke MaybeStartGathering as the last step of DoSetLocalDescription. This ensures that candidates and onicegatheringstatechange does not happen before SLD is resolved. This is important for passing external/wpt/webrtc/RTCPeerConnection-iceGatheringState.html. Original change's description: > [Perfect Negotiation] Implement non-racy version of SetLocalDescription. > > BACKGROUND > > When SLD is invoked with SetSessionDescriptionObserver, the observer is > called by posting a message back to the execution thread, delaying the > call. This delay is "artificial" - it's not necessary; the operation is > already complete. It's a post from the signaling thread to the signaling > thread. The rationale for the post was to avoid the observer making > recursive calls back into the PeerConnection. The problem with this is > that by the time the observer is called, the PeerConnection could > already have executed other operations and modified its states. > > This causes the referenced bug: one can have a race where SLD is > resolved "too late" (after a pending SRD is executed) and the signaling > state observed when SLD resolves doesn't make sense. > > When implementing Unified Plan, we fixed similar issues for SRD by > adding a version that takes SetRemoteDescriptionObserverInterface as > argument instead of SetSessionDescriptionObserver. The new version did > not have the delay. The old version had to be kept around not to break > downstream projects that had dependencies both on he delay and on > allowing the PC to be destroyed midst-operation without informing its > observers. > > THIS CL > > This does the old SRD fix for SLD as well: A new observer interface is > added, SetLocalDescriptionObserverInterface, and > PeerConnection::SetLocalDescription() is overloaded. If you call it with > the old observer, you get the delay, but if you call it with the new > observer, you don't get a delay. > > - SetLocalDescriptionObserverInterface is added. > - SetLocalDescription is overloaded. > - The adapter for SetSessionDescriptionObserver that causes the delay > previously only used for SRD is updated to handle both SLD and SRD. > - FakeSetLocalDescriptionObserver is added and > MockSetRemoteDescriptionObserver is renamed "Fake...". > > Bug: chromium:1071733 > Change-Id: I920368e648bede481058ac22f5b8794752a220b3 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179100 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31798} TBR=hta@webrtc.org Bug: chromium:1071733 Change-Id: Ic6e8d96afa1c19604762f373716c08dbfa9d178c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180481 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31804}
2020-07-29 12:04:00 +02:00
// TODO(https://crbug.com/webrtc/11798): Delete this methods in favor of the
// ones taking SetRemoteDescriptionObserverInterface as argument.
void SetRemoteDescription(SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) override;
PeerConnectionInterface::RTCConfiguration GetConfiguration() override;
RTCError SetConfiguration(
const PeerConnectionInterface::RTCConfiguration& configuration) override;
bool AddIceCandidate(const IceCandidateInterface* candidate) override;
void AddIceCandidate(std::unique_ptr<IceCandidateInterface> candidate,
std::function<void(RTCError)> callback) override;
bool RemoveIceCandidates(
const std::vector<cricket::Candidate>& candidates) override;
RTCError SetBitrate(const BitrateSettings& bitrate) override;
void SetAudioPlayout(bool playout) override;
void SetAudioRecording(bool recording) override;
rtc::scoped_refptr<DtlsTransportInterface> LookupDtlsTransportByMid(
const std::string& mid) override;
rtc::scoped_refptr<DtlsTransport> LookupDtlsTransportByMidInternal(
const std::string& mid);
rtc::scoped_refptr<SctpTransportInterface> GetSctpTransport() const override;
void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) override;
Revert "Revert "Encode log events periodically instead of for every event."" This reverts commit 33c5c7f5e4f018a5103770021328fc530d451d75. Reason for revert: Fix broken API change. TBR=sprang@webrtc.org,solenberg@webrtc.org TBRing because only pc/ and api/ have changed since last LGTMed version. Original change's description: > Revert "Encode log events periodically instead of for every event." > > This reverts commit b154c27e72fddb6c0d7cac69a9c68fff22154519. > > Reason for revert: Broke the internal project. > > Original change's description: > > Encode log events periodically instead of for every event. > > > > Updated unit test to take output_period and random seed as parameters. > > Updated the peerconnection interface to allow passing in an output_period. > > > > This is in preparation of some upcoming CLs that will change the format > > to store batches of delta-encoded values. > > > > > > Bug: webrtc:8111 > > Change-Id: Id5d9844dfad8d8edad346cd7cbebc7eadaaa5416 > > Reviewed-on: https://webrtc-review.googlesource.com/22600 > > Commit-Queue: Björn Terelius <terelius@webrtc.org> > > Reviewed-by: Elad Alon <eladalon@webrtc.org> > > Reviewed-by: Tommi <tommi@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20736} > > Change-Id: I2257c46c014adb8c7c4fb28538635cabed1f2229 > Bug: webrtc:8111 > Reviewed-on: https://webrtc-review.googlesource.com/24160 > Reviewed-by: Zhi Huang <zhihuang@webrtc.org> > Commit-Queue: Zhi Huang <zhihuang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20738} Bug: webrtc:8111 Change-Id: Ie69862cd52d11c1e15adeb6e2caacafe16863c80 Reviewed-on: https://webrtc-review.googlesource.com/24620 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Elad Alon <eladalon@webrtc.org> Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20811}
2017-11-20 17:38:14 +01:00
bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms) override;
bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output) override;
void StopRtcEventLog() override;
void Close() override;
rtc::Thread* signaling_thread() const final {
return context_->signaling_thread();
}
// PeerConnectionInternal implementation.
rtc::Thread* network_thread() const final {
return context_->network_thread();
}
rtc::Thread* worker_thread() const final { return context_->worker_thread(); }
std::string session_id() const override {
return session_id_;
}
bool initial_offerer() const override {
RTC_DCHECK_RUN_ON(signaling_thread());
return transport_controller_ && transport_controller_->initial_offerer();
}
std::vector<
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
GetTransceiversInternal() const override {
RTC_DCHECK_RUN_ON(signaling_thread());
return rtp_manager()->transceivers()->List();
}
sigslot::signal1<SctpDataChannel*>& SignalSctpDataChannelCreated() override {
return data_channel_controller_.SignalSctpDataChannelCreated();
}
std::vector<DataChannelStats> GetDataChannelStats() const override;
absl::optional<std::string> sctp_transport_name() const override;
absl::optional<std::string> sctp_mid() const override;
cricket::CandidateStatsList GetPooledCandidateStats() const override;
std::map<std::string, cricket::TransportStats> GetTransportStatsByNames(
const std::set<std::string>& transport_names) override;
Call::Stats GetCallStats() override;
bool GetLocalCertificate(
const std::string& transport_name,
rtc::scoped_refptr<rtc::RTCCertificate>* certificate) override;
std::unique_ptr<rtc::SSLCertChain> GetRemoteSSLCertChain(
const std::string& transport_name) override;
bool IceRestartPending(const std::string& content_name) const override;
bool NeedsIceRestart(const std::string& content_name) const override;
bool GetSslRole(const std::string& content_name, rtc::SSLRole* role) override;
// Functions needed by DataChannelController
void NoteDataAddedEvent() { NoteUsageEvent(UsageEvent::DATA_ADDED); }
// Returns the observer. Will crash on CHECK if the observer is removed.
PeerConnectionObserver* Observer() const;
bool IsClosed() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return !sdp_handler_ ||
sdp_handler_->signaling_state() == PeerConnectionInterface::kClosed;
}
// Get current SSL role used by SCTP's underlying transport.
bool GetSctpSslRole(rtc::SSLRole* role);
// Handler for the "channel closed" signal
void OnSctpDataChannelClosed(DataChannelInterface* channel);
bool ShouldFireNegotiationNeededEvent(uint32_t event_id) override;
// Functions needed by SdpOfferAnswerHandler
StatsCollector* stats() {
RTC_DCHECK_RUN_ON(signaling_thread());
return stats_.get();
}
DataChannelController* data_channel_controller() {
RTC_DCHECK_RUN_ON(signaling_thread());
return &data_channel_controller_;
}
bool dtls_enabled() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return dtls_enabled_;
}
const PeerConnectionInterface::RTCConfiguration* configuration() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return &configuration_;
}
PeerConnectionMessageHandler* message_handler() {
RTC_DCHECK_RUN_ON(signaling_thread());
return &message_handler_;
}
RtpTransmissionManager* rtp_manager() { return rtp_manager_.get(); }
const RtpTransmissionManager* rtp_manager() const {
return rtp_manager_.get();
}
cricket::ChannelManager* channel_manager() const;
JsepTransportController* transport_controller() {
return transport_controller_.get();
}
cricket::PortAllocator* port_allocator() { return port_allocator_.get(); }
Call* call_ptr() { return call_ptr_; }
ConnectionContext* context() { return context_.get(); }
const PeerConnectionFactoryInterface::Options* options() const {
return &options_;
}
void SetIceConnectionState(IceConnectionState new_state);
void NoteUsageEvent(UsageEvent event);
// Asynchronously adds a remote candidate on the network thread.
void AddRemoteCandidate(const std::string& mid,
const cricket::Candidate& candidate);
// Report the UMA metric SdpFormatReceived for the given remote description.
void ReportSdpFormatReceived(
const SessionDescriptionInterface& remote_description);
// Report the UMA metric BundleUsage for the given remote description.
void ReportSdpBundleUsage(
const SessionDescriptionInterface& remote_description);
// Returns true if the PeerConnection is configured to use Unified Plan
// semantics for creating offers/answers and setting local/remote
// descriptions. If this is true the RtpTransceiver API will also be available
// to the user. If this is false, Plan B semantics are assumed.
// TODO(bugs.webrtc.org/8530): Flip the default to be Unified Plan once
// sufficient time has passed.
bool IsUnifiedPlan() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return is_unified_plan_;
}
[Unified Plan] Support multiple BUNDLE groups. In this CL, JsepTransportController and MediaSessionDescriptionFactory are updated not to assume that there only exists at most a single BUNDLE group but a list of N groups. This makes it possible to create multiple BUNDLE groups by having multiple "a=group:BUNDLE" lines in the SDP. This makes it possible to have some m= sections in one group and some other m= sections in another group. For example, you could group all audio m= sections in one group and all video m= sections in another group. This enables "send all audio tracks on one transport and all video tracks on another transport" in Unified Plan. This is something that was possible in Plan B because all ssrcs in the same m= section were implicitly bundled together forming a group of audio m= section and video m= section (even without use of the BUNDLE tag). PeerConnection will never create multiple BUNDLE groups by default, but upon setting SDP with multiple BUNDLE groups the PeerConnection will accept them if configured to accept BUNDLE. This makes it possible to accept an SFU's BUNDLE offer without having to SDP munge the answer. C++ unit tests are added. This fix has also been verified manually on: https://jsfiddle.net/henbos/to89L6ce/43/ Without fix: 0+2 get bundled, 1+3 don't get bundled. With fix: 0+2 get bundled in first group, 1+3 get bundled in second group. Bug: webrtc:10208 Change-Id: Iaf451fa5459c484730c8018274166ef154b19af8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214487 Reviewed-by: Taylor <deadbeef@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33838}
2021-04-26 21:04:26 +02:00
bool ValidateBundleSettings(
const cricket::SessionDescription* desc,
const std::map<std::string, const cricket::ContentGroup*>&
bundle_groups_by_mid);
// Returns the MID for the data section associated with the
// SCTP data channel, if it has been set. If no data
// channels are configured this will return nullopt.
absl::optional<std::string> GetDataMid() const;
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
void SetSctpDataMid(const std::string& mid);
void ResetSctpDataMid();
// Asynchronously calls SctpTransport::Start() on the network thread for
// |sctp_mid()| if set. Called as part of setting the local description.
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
void StartSctpTransport(int local_port,
int remote_port,
int max_message_size);
// Returns the CryptoOptions for this PeerConnection. This will always
// return the RTCConfiguration.crypto_options if set and will only default
// back to the PeerConnectionFactory settings if nothing was set.
CryptoOptions GetCryptoOptions();
// Internal implementation for AddTransceiver family of methods. If
// |fire_callback| is set, fires OnRenegotiationNeeded callback if successful.
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
cricket::MediaType media_type,
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init,
bool fire_callback = true);
// Returns rtp transport, result can not be nullptr.
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
RtpTransportInternal* GetRtpTransport(const std::string& mid);
// Returns true if SRTP (either using DTLS-SRTP or SDES) is required by
// this session.
bool SrtpRequired() const;
bool SetupDataChannelTransport_n(const std::string& mid)
RTC_RUN_ON(network_thread());
void TeardownDataChannelTransport_n() RTC_RUN_ON(network_thread());
cricket::ChannelInterface* GetChannel(const std::string& content_name)
RTC_RUN_ON(network_thread());
// Functions made public for testing.
void ReturnHistogramVeryQuicklyForTesting() {
RTC_DCHECK_RUN_ON(signaling_thread());
return_histogram_very_quickly_ = true;
}
void RequestUsagePatternReportForTesting();
protected:
// Available for rtc::scoped_refptr creation
PeerConnection(rtc::scoped_refptr<ConnectionContext> context,
const PeerConnectionFactoryInterface::Options& options,
bool is_unified_plan,
std::unique_ptr<RtcEventLog> event_log,
std::unique_ptr<Call> call,
Reland "Remove thread hops from events provided by JsepTransportController." This reverts commit 6e4fcac31312f2dda5b60d33874ff0cd62f94321. Reason for revert: Parent CL issue has been resolved. Original change's description: > Revert "Remove thread hops from events provided by JsepTransportController." > > This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76. > > Reason for revert: Parent CL breaks FYI bots. > See https://webrtc-review.googlesource.com/c/src/+/206466 > > Original change's description: > > Remove thread hops from events provided by JsepTransportController. > > > > Events associated with Subscribe* methods in JTC had trampolines that > > would use an async invoker to fire the events on the signaling thread. > > This was being done for the purposes of PeerConnection but the concept > > of a signaling thread is otherwise not applicable to JTC and use of > > JTC from PC is inconsistent across threads (as has been flagged in > > webrtc:9987). > > > > This change makes all CallbackList members only accessible from the > > network thread and moves the signaling thread related work over to > > PeerConnection, which makes hops there more visible as well as making > > that class easier to refactor for thread efficiency. > > > > This CL removes the AsyncInvoker from JTC (webrtc:12339) > > > > The signaling_thread_ variable is also removed from JTC and more thread > > checks added to catch errors. > > > > Bug: webrtc:12427, webrtc:11988, webrtc:12339 > > Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062 > > Commit-Queue: Tommi <tommi@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33195} > > TBR=nisse@webrtc.org,tommi@webrtc.org > > Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:12427 > Bug: webrtc:11988 > Bug: webrtc:12339 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467 > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Commit-Queue: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33203} TBR=nisse@webrtc.org,tommi@webrtc.org,guidou@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:12427 Bug: webrtc:11988 Bug: webrtc:12339 Change-Id: I4e2e1490e1f9a87ed6ac4d722fd3c442e3059ae0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206809 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33225}
2021-02-10 17:40:08 +00:00
PeerConnectionDependencies& dependencies,
bool dtls_enabled);
~PeerConnection() override;
private:
RTCError Initialize(
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies dependencies);
Reland "Remove thread hops from events provided by JsepTransportController." This reverts commit 6e4fcac31312f2dda5b60d33874ff0cd62f94321. Reason for revert: Parent CL issue has been resolved. Original change's description: > Revert "Remove thread hops from events provided by JsepTransportController." > > This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76. > > Reason for revert: Parent CL breaks FYI bots. > See https://webrtc-review.googlesource.com/c/src/+/206466 > > Original change's description: > > Remove thread hops from events provided by JsepTransportController. > > > > Events associated with Subscribe* methods in JTC had trampolines that > > would use an async invoker to fire the events on the signaling thread. > > This was being done for the purposes of PeerConnection but the concept > > of a signaling thread is otherwise not applicable to JTC and use of > > JTC from PC is inconsistent across threads (as has been flagged in > > webrtc:9987). > > > > This change makes all CallbackList members only accessible from the > > network thread and moves the signaling thread related work over to > > PeerConnection, which makes hops there more visible as well as making > > that class easier to refactor for thread efficiency. > > > > This CL removes the AsyncInvoker from JTC (webrtc:12339) > > > > The signaling_thread_ variable is also removed from JTC and more thread > > checks added to catch errors. > > > > Bug: webrtc:12427, webrtc:11988, webrtc:12339 > > Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062 > > Commit-Queue: Tommi <tommi@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33195} > > TBR=nisse@webrtc.org,tommi@webrtc.org > > Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:12427 > Bug: webrtc:11988 > Bug: webrtc:12339 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467 > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Commit-Queue: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33203} TBR=nisse@webrtc.org,tommi@webrtc.org,guidou@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:12427 Bug: webrtc:11988 Bug: webrtc:12339 Change-Id: I4e2e1490e1f9a87ed6ac4d722fd3c442e3059ae0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206809 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33225}
2021-02-10 17:40:08 +00:00
void InitializeTransportController_n(
const RTCConfiguration& configuration,
const PeerConnectionDependencies& dependencies)
RTC_RUN_ON(network_thread());
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
FindTransceiverBySender(rtc::scoped_refptr<RtpSenderInterface> sender)
RTC_RUN_ON(signaling_thread());
void SetStandardizedIceConnectionState(
PeerConnectionInterface::IceConnectionState new_state)
RTC_RUN_ON(signaling_thread());
void SetConnectionState(
PeerConnectionInterface::PeerConnectionState new_state)
RTC_RUN_ON(signaling_thread());
// Called any time the IceGatheringState changes.
void OnIceGatheringChange(IceGatheringState new_state)
RTC_RUN_ON(signaling_thread());
// New ICE candidate has been gathered.
void OnIceCandidate(std::unique_ptr<IceCandidateInterface> candidate)
RTC_RUN_ON(signaling_thread());
// Gathering of an ICE candidate failed.
void OnIceCandidateError(const std::string& address,
int port,
const std::string& url,
int error_code,
const std::string& error_text)
RTC_RUN_ON(signaling_thread());
// Some local ICE candidates have been removed.
void OnIceCandidatesRemoved(const std::vector<cricket::Candidate>& candidates)
RTC_RUN_ON(signaling_thread());
void OnSelectedCandidatePairChanged(
const cricket::CandidatePairChangeEvent& event)
RTC_RUN_ON(signaling_thread());
void OnNegotiationNeeded();
// Returns the specified SCTP DataChannel in sctp_data_channels_,
// or nullptr if not found.
SctpDataChannel* FindDataChannelBySid(int sid) const
RTC_RUN_ON(signaling_thread());
// Called when first configuring the port allocator.
struct InitializePortAllocatorResult {
bool enable_ipv6;
};
InitializePortAllocatorResult InitializePortAllocator_n(
const cricket::ServerAddresses& stun_servers,
const std::vector<cricket::RelayServerConfig>& turn_servers,
const RTCConfiguration& configuration);
// Called when SetConfiguration is called to apply the supported subset
// of the configuration on the network thread.
bool ReconfigurePortAllocator_n(
const cricket::ServerAddresses& stun_servers,
const std::vector<cricket::RelayServerConfig>& turn_servers,
IceTransportsType type,
int candidate_pool_size,
PortPrunePolicy turn_port_prune_policy,
webrtc::TurnCustomizer* turn_customizer,
absl::optional<int> stun_candidate_keepalive_interval,
bool have_local_description);
// Starts output of an RTC event log to the given output object.
// This function should only be called from the worker thread.
Revert "Revert "Encode log events periodically instead of for every event."" This reverts commit 33c5c7f5e4f018a5103770021328fc530d451d75. Reason for revert: Fix broken API change. TBR=sprang@webrtc.org,solenberg@webrtc.org TBRing because only pc/ and api/ have changed since last LGTMed version. Original change's description: > Revert "Encode log events periodically instead of for every event." > > This reverts commit b154c27e72fddb6c0d7cac69a9c68fff22154519. > > Reason for revert: Broke the internal project. > > Original change's description: > > Encode log events periodically instead of for every event. > > > > Updated unit test to take output_period and random seed as parameters. > > Updated the peerconnection interface to allow passing in an output_period. > > > > This is in preparation of some upcoming CLs that will change the format > > to store batches of delta-encoded values. > > > > > > Bug: webrtc:8111 > > Change-Id: Id5d9844dfad8d8edad346cd7cbebc7eadaaa5416 > > Reviewed-on: https://webrtc-review.googlesource.com/22600 > > Commit-Queue: Björn Terelius <terelius@webrtc.org> > > Reviewed-by: Elad Alon <eladalon@webrtc.org> > > Reviewed-by: Tommi <tommi@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20736} > > Change-Id: I2257c46c014adb8c7c4fb28538635cabed1f2229 > Bug: webrtc:8111 > Reviewed-on: https://webrtc-review.googlesource.com/24160 > Reviewed-by: Zhi Huang <zhihuang@webrtc.org> > Commit-Queue: Zhi Huang <zhihuang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20738} Bug: webrtc:8111 Change-Id: Ie69862cd52d11c1e15adeb6e2caacafe16863c80 Reviewed-on: https://webrtc-review.googlesource.com/24620 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Elad Alon <eladalon@webrtc.org> Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20811}
2017-11-20 17:38:14 +01:00
bool StartRtcEventLog_w(std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms);
// Stops recording an RTC event log.
// This function should only be called from the worker thread.
void StopRtcEventLog_w();
// Returns true and the TransportInfo of the given |content_name|
// from |description|. Returns false if it's not available.
static bool GetTransportDescription(
const cricket::SessionDescription* description,
const std::string& content_name,
cricket::TransportDescription* info);
// Returns the media index for a local ice candidate given the content name.
// Returns false if the local session description does not have a media
// content called |content_name|.
bool GetLocalCandidateMediaIndex(const std::string& content_name,
int* sdp_mline_index)
RTC_RUN_ON(signaling_thread());
// JsepTransportController signal handlers.
void OnTransportControllerConnectionState(cricket::IceConnectionState state)
RTC_RUN_ON(signaling_thread());
void OnTransportControllerGatheringState(cricket::IceGatheringState state)
RTC_RUN_ON(signaling_thread());
void OnTransportControllerCandidatesGathered(
const std::string& transport_name,
const std::vector<cricket::Candidate>& candidates)
RTC_RUN_ON(signaling_thread());
void OnTransportControllerCandidateError(
const cricket::IceCandidateErrorEvent& event)
RTC_RUN_ON(signaling_thread());
void OnTransportControllerCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates)
RTC_RUN_ON(signaling_thread());
void OnTransportControllerCandidateChanged(
const cricket::CandidatePairChangeEvent& event)
RTC_RUN_ON(signaling_thread());
void OnTransportControllerDtlsHandshakeError(rtc::SSLHandshakeError error);
// Invoked when TransportController connection completion is signaled.
// Reports stats for all transports in use.
void ReportTransportStats() RTC_RUN_ON(network_thread());
// Gather the usage of IPv4/IPv6 as best connection.
Reland "Remove thread hops from events provided by JsepTransportController." This reverts commit 6e4fcac31312f2dda5b60d33874ff0cd62f94321. Reason for revert: Parent CL issue has been resolved. Original change's description: > Revert "Remove thread hops from events provided by JsepTransportController." > > This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76. > > Reason for revert: Parent CL breaks FYI bots. > See https://webrtc-review.googlesource.com/c/src/+/206466 > > Original change's description: > > Remove thread hops from events provided by JsepTransportController. > > > > Events associated with Subscribe* methods in JTC had trampolines that > > would use an async invoker to fire the events on the signaling thread. > > This was being done for the purposes of PeerConnection but the concept > > of a signaling thread is otherwise not applicable to JTC and use of > > JTC from PC is inconsistent across threads (as has been flagged in > > webrtc:9987). > > > > This change makes all CallbackList members only accessible from the > > network thread and moves the signaling thread related work over to > > PeerConnection, which makes hops there more visible as well as making > > that class easier to refactor for thread efficiency. > > > > This CL removes the AsyncInvoker from JTC (webrtc:12339) > > > > The signaling_thread_ variable is also removed from JTC and more thread > > checks added to catch errors. > > > > Bug: webrtc:12427, webrtc:11988, webrtc:12339 > > Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062 > > Commit-Queue: Tommi <tommi@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33195} > > TBR=nisse@webrtc.org,tommi@webrtc.org > > Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:12427 > Bug: webrtc:11988 > Bug: webrtc:12339 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467 > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Commit-Queue: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33203} TBR=nisse@webrtc.org,tommi@webrtc.org,guidou@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:12427 Bug: webrtc:11988 Bug: webrtc:12339 Change-Id: I4e2e1490e1f9a87ed6ac4d722fd3c442e3059ae0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206809 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33225}
2021-02-10 17:40:08 +00:00
static void ReportBestConnectionState(const cricket::TransportStats& stats);
Reland "Remove thread hops from events provided by JsepTransportController." This reverts commit 6e4fcac31312f2dda5b60d33874ff0cd62f94321. Reason for revert: Parent CL issue has been resolved. Original change's description: > Revert "Remove thread hops from events provided by JsepTransportController." > > This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76. > > Reason for revert: Parent CL breaks FYI bots. > See https://webrtc-review.googlesource.com/c/src/+/206466 > > Original change's description: > > Remove thread hops from events provided by JsepTransportController. > > > > Events associated with Subscribe* methods in JTC had trampolines that > > would use an async invoker to fire the events on the signaling thread. > > This was being done for the purposes of PeerConnection but the concept > > of a signaling thread is otherwise not applicable to JTC and use of > > JTC from PC is inconsistent across threads (as has been flagged in > > webrtc:9987). > > > > This change makes all CallbackList members only accessible from the > > network thread and moves the signaling thread related work over to > > PeerConnection, which makes hops there more visible as well as making > > that class easier to refactor for thread efficiency. > > > > This CL removes the AsyncInvoker from JTC (webrtc:12339) > > > > The signaling_thread_ variable is also removed from JTC and more thread > > checks added to catch errors. > > > > Bug: webrtc:12427, webrtc:11988, webrtc:12339 > > Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062 > > Commit-Queue: Tommi <tommi@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33195} > > TBR=nisse@webrtc.org,tommi@webrtc.org > > Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:12427 > Bug: webrtc:11988 > Bug: webrtc:12339 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467 > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Commit-Queue: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33203} TBR=nisse@webrtc.org,tommi@webrtc.org,guidou@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:12427 Bug: webrtc:11988 Bug: webrtc:12339 Change-Id: I4e2e1490e1f9a87ed6ac4d722fd3c442e3059ae0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206809 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33225}
2021-02-10 17:40:08 +00:00
static void ReportNegotiatedCiphers(
bool dtls_enabled,
const cricket::TransportStats& stats,
const std::set<cricket::MediaType>& media_types);
void ReportIceCandidateCollected(const cricket::Candidate& candidate)
RTC_RUN_ON(signaling_thread());
void ReportUsagePattern() const RTC_RUN_ON(signaling_thread());
void ReportRemoteIceCandidateAdded(const cricket::Candidate& candidate);
// JsepTransportController::Observer override.
//
// Called by |transport_controller_| when processing transport information
// from a session description, and the mapping from m= sections to transports
// changed (as a result of BUNDLE negotiation, or m= sections being
// rejected).
Changes to enable use of DatagramTransport as a data channel transport. PeerConnection now has a new setting in RTCConfiguration to enable use of datagram transport for data channels. There is also a corresponding field trial, which has both a kill-switch and a way to change the default value. PeerConnection's interaction with MediaTransport for data channels has been refactored to work with DataChannelTransportInterface instead. Adds a DataChannelState and OnStateChanged() to the DataChannelSink callbacks. This allows PeerConnection to listen to the data channel's state directly, instead of indirectly by monitoring media transport state. This is necessary to enable use of non-media-transport (eg. datagram transport) data channel transports. For now, PeerConnection watches the state through MediaTransport as well. This will persist until MediaTransport implements the new callback. Datagram transport use is negotiated. As such, an offer that requests to use datagram transport for data channels may be rejected by the answerer. If the offer includes DTLS, the data channels will be negotiated as SCTP/DTLS data channels with an extra x-opaque parameter for datagram transport. If the opaque parameter is rejected (by an answerer without datagram support), the offerer may fall back to SCTP. If DTLS is not enabled, there is no viable fallback. In this case, the data channels are negotiated as media transport data channels. If the receiver does not understand the x-opaque line, it will reject these data channels, and the offerer's data channels will be closed. Bug: webrtc:9719 Change-Id: Ic1bf3664c4bcf9d754482df59897f5f72fe68fcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147702 Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28932}
2019-08-21 10:44:59 -07:00
bool OnTransportChanged(
const std::string& mid,
RtpTransportInternal* rtp_transport,
rtc::scoped_refptr<DtlsTransport> dtls_transport,
Reland "Reland "Refactor SCTP data channels to use DataChannelTransportInterface."" This is a reland of 487f9a17e426fd14bb06b13e861071b3f15d119b Original change's description: > Reland "Refactor SCTP data channels to use DataChannelTransportInterface." > > Also clears SctpTransport before deleting JsepTransport. > > SctpTransport is ref-counted, but the underlying transport is deleted when > JsepTransport clears the rtp_dtls_transport. This results in crashes when > usrsctp attempts to send outgoing packets through a dangling pointer to the > underlying transport. > > Clearing SctpTransport before DtlsTransport removes the pointer to the > underlying transport before it becomes invalid. > > This fixes a crash in chromium's web platform tests (see > https://chromium-review.googlesource.com/c/chromium/src/+/1776711). > > Original change's description: > > Refactor SCTP data channels to use DataChannelTransportInterface. > > > > This change moves SctpTransport to be owned by JsepTransport, which now > > holds a DataChannelTransport implementation for SCTP when it is used for > > data channels. > > > > This simplifies negotiation and fallback to SCTP. Negotiation can now > > use a composite DataChannelTransport, just as negotiation for RTP uses a > > composite RTP transport. > > > > PeerConnection also has one fewer way it needs to manage data channels. > > It now handles SCTP and datagram- or media-transport-based data channels > > the same way. > > > > There are a few leaky abstractions left. For example, PeerConnection > > calls Start() on the SctpTransport at a particular point in negotiation, > > but does not need to call this for other transports. Similarly, PC > > exposes an interface to the SCTP transport directly to the user; there > > is no equivalent for other transports. > > Bug: webrtc:9719 > Change-Id: I64e94b88afb119fdbf5f22750f88c8a084d53937 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151981 > Reviewed-by: Benjamin Wright <benwright@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Benjamin Wright <benwright@webrtc.org> > Commit-Queue: Bjorn Mellem <mellem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29120} Bug: webrtc:9719 Change-Id: I28481a3de64a3506bc57748106383eeba4ef205c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152740 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29290}
2019-09-23 14:53:54 -07:00
DataChannelTransportInterface* data_channel_transport) override;
std::function<void(const rtc::CopyOnWriteBuffer& packet,
int64_t packet_time_us)>
InitializeRtcpCallback();
const rtc::scoped_refptr<ConnectionContext> context_;
const PeerConnectionFactoryInterface::Options options_;
PeerConnectionObserver* observer_ RTC_GUARDED_BY(signaling_thread()) =
nullptr;
const bool is_unified_plan_;
// The EventLog needs to outlive |call_| (and any other object that uses it).
std::unique_ptr<RtcEventLog> event_log_ RTC_GUARDED_BY(worker_thread());
// Points to the same thing as `event_log_`. Since it's const, we may read the
// pointer (but not touch the object) from any thread.
RtcEventLog* const event_log_ptr_ RTC_PT_GUARDED_BY(worker_thread());
IceConnectionState ice_connection_state_ RTC_GUARDED_BY(signaling_thread()) =
kIceConnectionNew;
PeerConnectionInterface::IceConnectionState standardized_ice_connection_state_
RTC_GUARDED_BY(signaling_thread()) = kIceConnectionNew;
PeerConnectionInterface::PeerConnectionState connection_state_
RTC_GUARDED_BY(signaling_thread()) = PeerConnectionState::kNew;
IceGatheringState ice_gathering_state_ RTC_GUARDED_BY(signaling_thread()) =
kIceGatheringNew;
PeerConnectionInterface::RTCConfiguration configuration_
RTC_GUARDED_BY(signaling_thread());
Reland "Use the new DNS resolver API in PeerConnection" This reverts commit 5a40b3710545edfd8a634341df3de26f57d79281. Reason for revert: Fixed the bug and ran layout tests. Original change's description: > Revert "Use the new DNS resolver API in PeerConnection" > > This reverts commit acf8ccb3c9f001b0ed749aca52b2d436d66f9586. > > Reason for revert: Speculative revert for https://ci.chromium.org/ui/p/chromium/builders/try/win10_chromium_x64_rel_ng/b8851745102358680592/overview. > > Original change's description: > > Use the new DNS resolver API in PeerConnection > > > > Bug: webrtc:12598 > > Change-Id: I5a14058e7f28c993ed927749df7357c715ba83fb > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212961 > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33561} > > # Not skipping CQ checks because original CL landed > 1 day ago. > > TBR=hta@webrtc.org > > Bug: webrtc:12598 > Change-Id: Idc9853cb569849c49052f9cbd865614710fff979 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213188 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33591} # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:12598 Change-Id: Ief7867f2f23de66504877cdab1b23a11df2d5de4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214120 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33647}
2021-04-08 07:25:04 +00:00
const std::unique_ptr<AsyncDnsResolverFactoryInterface>
async_dns_resolver_factory_;
std::unique_ptr<cricket::PortAllocator>
port_allocator_; // TODO(bugs.webrtc.org/9987): Accessed on both
// signaling and network thread.
const std::unique_ptr<webrtc::IceTransportFactory>
ice_transport_factory_; // TODO(bugs.webrtc.org/9987): Accessed on the
// signaling thread but the underlying raw
// pointer is given to
// |jsep_transport_controller_| and used on the
// network thread.
const std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier_
RTC_GUARDED_BY(network_thread());
// The unique_ptr belongs to the worker thread, but the Call object manages
// its own thread safety.
std::unique_ptr<Call> call_ RTC_GUARDED_BY(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
ScopedTaskSafety signaling_thread_safety_;
rtc::scoped_refptr<PendingTaskSafetyFlag> network_thread_safety_;
Reland "Remove thread hops from events provided by JsepTransportController." This reverts commit 6e4fcac31312f2dda5b60d33874ff0cd62f94321. Reason for revert: Parent CL issue has been resolved. Original change's description: > Revert "Remove thread hops from events provided by JsepTransportController." > > This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76. > > Reason for revert: Parent CL breaks FYI bots. > See https://webrtc-review.googlesource.com/c/src/+/206466 > > Original change's description: > > Remove thread hops from events provided by JsepTransportController. > > > > Events associated with Subscribe* methods in JTC had trampolines that > > would use an async invoker to fire the events on the signaling thread. > > This was being done for the purposes of PeerConnection but the concept > > of a signaling thread is otherwise not applicable to JTC and use of > > JTC from PC is inconsistent across threads (as has been flagged in > > webrtc:9987). > > > > This change makes all CallbackList members only accessible from the > > network thread and moves the signaling thread related work over to > > PeerConnection, which makes hops there more visible as well as making > > that class easier to refactor for thread efficiency. > > > > This CL removes the AsyncInvoker from JTC (webrtc:12339) > > > > The signaling_thread_ variable is also removed from JTC and more thread > > checks added to catch errors. > > > > Bug: webrtc:12427, webrtc:11988, webrtc:12339 > > Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062 > > Commit-Queue: Tommi <tommi@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33195} > > TBR=nisse@webrtc.org,tommi@webrtc.org > > Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:12427 > Bug: webrtc:11988 > Bug: webrtc:12339 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467 > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Commit-Queue: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33203} TBR=nisse@webrtc.org,tommi@webrtc.org,guidou@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:12427 Bug: webrtc:11988 Bug: webrtc:12339 Change-Id: I4e2e1490e1f9a87ed6ac4d722fd3c442e3059ae0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206809 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33225}
2021-02-10 17:40:08 +00:00
rtc::scoped_refptr<PendingTaskSafetyFlag> worker_thread_safety_;
// Points to the same thing as `call_`. Since it's const, we may read the
// pointer from any thread.
// TODO(bugs.webrtc.org/11992): Remove this workaround (and potential dangling
// pointer).
Call* const call_ptr_;
std::unique_ptr<StatsCollector> stats_
RTC_GUARDED_BY(signaling_thread()); // A pointer is passed to senders_
rtc::scoped_refptr<RTCStatsCollector> stats_collector_
RTC_GUARDED_BY(signaling_thread());
const std::string session_id_;
std::unique_ptr<JsepTransportController>
transport_controller_; // TODO(bugs.webrtc.org/9987): Accessed on both
// signaling and network thread.
// |sctp_mid_| is the content name (MID) in SDP.
Changes to enable use of DatagramTransport as a data channel transport. PeerConnection now has a new setting in RTCConfiguration to enable use of datagram transport for data channels. There is also a corresponding field trial, which has both a kill-switch and a way to change the default value. PeerConnection's interaction with MediaTransport for data channels has been refactored to work with DataChannelTransportInterface instead. Adds a DataChannelState and OnStateChanged() to the DataChannelSink callbacks. This allows PeerConnection to listen to the data channel's state directly, instead of indirectly by monitoring media transport state. This is necessary to enable use of non-media-transport (eg. datagram transport) data channel transports. For now, PeerConnection watches the state through MediaTransport as well. This will persist until MediaTransport implements the new callback. Datagram transport use is negotiated. As such, an offer that requests to use datagram transport for data channels may be rejected by the answerer. If the offer includes DTLS, the data channels will be negotiated as SCTP/DTLS data channels with an extra x-opaque parameter for datagram transport. If the opaque parameter is rejected (by an answerer without datagram support), the offerer may fall back to SCTP. If DTLS is not enabled, there is no viable fallback. In this case, the data channels are negotiated as media transport data channels. If the receiver does not understand the x-opaque line, it will reject these data channels, and the offerer's data channels will be closed. Bug: webrtc:9719 Change-Id: Ic1bf3664c4bcf9d754482df59897f5f72fe68fcc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147702 Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28932}
2019-08-21 10:44:59 -07:00
// Note: this is used as the data channel MID by both SCTP and data channel
// transports. It is set when either transport is initialized and unset when
// both transports are deleted.
// There is one copy on the signaling thread and another copy on the
// networking thread. Changes are always initiated from the signaling
// thread, but applied first on the networking thread via an invoke().
absl::optional<std::string> sctp_mid_s_ RTC_GUARDED_BY(signaling_thread());
absl::optional<std::string> sctp_mid_n_ RTC_GUARDED_BY(network_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
std::string sctp_transport_name_s_ RTC_GUARDED_BY(signaling_thread());
// The machinery for handling offers and answers. Const after initialization.
std::unique_ptr<SdpOfferAnswerHandler> sdp_handler_
RTC_GUARDED_BY(signaling_thread());
Reland "Remove thread hops from events provided by JsepTransportController." This reverts commit 6e4fcac31312f2dda5b60d33874ff0cd62f94321. Reason for revert: Parent CL issue has been resolved. Original change's description: > Revert "Remove thread hops from events provided by JsepTransportController." > > This reverts commit f554b3c577f69fa9ffad5c07155898c2d985ac76. > > Reason for revert: Parent CL breaks FYI bots. > See https://webrtc-review.googlesource.com/c/src/+/206466 > > Original change's description: > > Remove thread hops from events provided by JsepTransportController. > > > > Events associated with Subscribe* methods in JTC had trampolines that > > would use an async invoker to fire the events on the signaling thread. > > This was being done for the purposes of PeerConnection but the concept > > of a signaling thread is otherwise not applicable to JTC and use of > > JTC from PC is inconsistent across threads (as has been flagged in > > webrtc:9987). > > > > This change makes all CallbackList members only accessible from the > > network thread and moves the signaling thread related work over to > > PeerConnection, which makes hops there more visible as well as making > > that class easier to refactor for thread efficiency. > > > > This CL removes the AsyncInvoker from JTC (webrtc:12339) > > > > The signaling_thread_ variable is also removed from JTC and more thread > > checks added to catch errors. > > > > Bug: webrtc:12427, webrtc:11988, webrtc:12339 > > Change-Id: Id232aedd00dfd5403b2ba0ca147d3eca7c12c7c5 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206062 > > Commit-Queue: Tommi <tommi@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33195} > > TBR=nisse@webrtc.org,tommi@webrtc.org > > Change-Id: I6134b71b74a9408854b79d44506d513519e9cf4d > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:12427 > Bug: webrtc:11988 > Bug: webrtc:12339 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206467 > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > Commit-Queue: Guido Urdaneta <guidou@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33203} TBR=nisse@webrtc.org,tommi@webrtc.org,guidou@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:12427 Bug: webrtc:11988 Bug: webrtc:12339 Change-Id: I4e2e1490e1f9a87ed6ac4d722fd3c442e3059ae0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206809 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33225}
2021-02-10 17:40:08 +00:00
const bool dtls_enabled_;
UsagePattern usage_pattern_ RTC_GUARDED_BY(signaling_thread());
bool return_histogram_very_quickly_ RTC_GUARDED_BY(signaling_thread()) =
false;
DataChannelController data_channel_controller_;
// Machinery for handling messages posted to oneself
PeerConnectionMessageHandler message_handler_;
// Administration of senders, receivers and transceivers
// Accessed on both signaling and network thread. Const after Initialize().
std::unique_ptr<RtpTransmissionManager> rtp_manager_;
rtc::WeakPtrFactory<PeerConnection> weak_factory_;
// Did the connectionState ever change to `connected`?
// Used to gather metrics only the first such state change.
bool was_ever_connected_ RTC_GUARDED_BY(signaling_thread()) = false;
};
} // namespace webrtc
#endif // PC_PEER_CONNECTION_H_