Add thread safety annotations for PeerConnection::configuration_
Plus all the annotations that are necessary to make things compile
again.
Bug: webrtc:9987
Change-Id: I4be508284af573d93657c933a64e9f970b7e3adf
Reviewed-on: https://webrtc-review.googlesource.com/c/123190
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26805}diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index c2defe2..341a570 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -900,6 +900,7 @@
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies dependencies) {
RTC_DCHECK_RUN_ON(signaling_thread());
+ RTC_DCHECK_RUNS_SERIALIZED(&use_media_transport_race_checker_);
TRACE_EVENT0("webrtc", "PeerConnection::Initialize");
RTCError config_error = ValidateConfiguration(configuration);
@@ -1029,6 +1030,7 @@
stats_collector_ = RTCStatsCollector::Create(this);
configuration_ = configuration;
+ use_media_transport_ = configuration.use_media_transport;
// Obtain a certificate from RTCConfiguration if any were provided (optional).
rtc::scoped_refptr<rtc::RTCCertificate> certificate;
@@ -1144,6 +1146,7 @@
}
rtc::scoped_refptr<StreamCollectionInterface> PeerConnection::local_streams() {
+ RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(!IsUnifiedPlan()) << "local_streams is not available with Unified "
"Plan SdpSemantics. Please use GetSenders "
"instead.";
@@ -1151,6 +1154,7 @@
}
rtc::scoped_refptr<StreamCollectionInterface> PeerConnection::remote_streams() {
+ RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(!IsUnifiedPlan()) << "remote_streams is not available with Unified "
"Plan SdpSemantics. Please use GetReceivers "
"instead.";
@@ -1624,6 +1628,7 @@
rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
const std::string& kind,
const std::string& stream_id) {
+ RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(!IsUnifiedPlan()) << "CreateSender is not available with Unified "
"Plan SdpSemantics. Please use AddTransceiver "
"instead.";
@@ -1714,6 +1719,7 @@
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::GetTransceivers() const {
+ RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(IsUnifiedPlan())
<< "GetTransceivers is only supported with Unified Plan SdpSemantics.";
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> all_transceivers;
@@ -1867,6 +1873,7 @@
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {
+ RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
if (!observer) {
@@ -2023,6 +2030,7 @@
void PeerConnection::SetLocalDescription(
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc_ptr) {
+ RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::SetLocalDescription");
// The SetLocalDescription contract is that we take ownership of the session
@@ -2382,6 +2390,7 @@
void PeerConnection::SetRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {
+ RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::SetRemoteDescription");
if (!observer) {
@@ -3234,11 +3243,14 @@
}
PeerConnectionInterface::RTCConfiguration PeerConnection::GetConfiguration() {
+ RTC_DCHECK_RUN_ON(signaling_thread());
return configuration_;
}
bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration,
RTCError* error) {
+ RTC_DCHECK_RUN_ON(signaling_thread());
+ RTC_DCHECK_RUNS_SERIALIZED(&use_media_transport_race_checker_);
TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
if (IsClosed()) {
RTC_LOG(LS_ERROR) << "SetConfiguration: PeerConnection is closed.";
@@ -3394,6 +3406,7 @@
}
configuration_ = modified_config;
+ use_media_transport_ = configuration.use_media_transport;
return SafeSetError(RTCErrorType::NONE, error);
}
@@ -6937,6 +6950,7 @@
RtpTransportInternal* rtp_transport,
cricket::DtlsTransportInternal* dtls_transport,
MediaTransportInterface* media_transport) {
+ RTC_DCHECK_RUNS_SERIALIZED(&use_media_transport_race_checker_);
bool ret = true;
auto base_channel = GetChannel(mid);
if (base_channel) {
@@ -6946,7 +6960,7 @@
sctp_transport_->SetDtlsTransport(dtls_transport);
}
- if (configuration_.use_media_transport) {
+ if (use_media_transport_) {
// Only pass media transport to call object if media transport is used
// for media (and not data channel).
call_->MediaTransportChange(media_transport);
diff --git a/pc/peer_connection.h b/pc/peer_connection.h
index 0640458..3d22bf4 100644
--- a/pc/peer_connection.h
+++ b/pc/peer_connection.h
@@ -29,6 +29,7 @@
#include "pc/stats_collector.h"
#include "pc/stream_collection.h"
#include "pc/webrtc_session_description_factory.h"
+#include "rtc_base/race_checker.h"
#include "rtc_base/unique_id_generator.h"
namespace webrtc {
@@ -303,8 +304,10 @@
// Plan B helpers for getting the voice/video media channels for the single
// audio/video transceiver, if it exists.
- cricket::VoiceMediaChannel* voice_media_channel() const;
- cricket::VideoMediaChannel* video_media_channel() const;
+ cricket::VoiceMediaChannel* voice_media_channel() const
+ RTC_RUN_ON(signaling_thread());
+ cricket::VideoMediaChannel* video_media_channel() const
+ RTC_RUN_ON(signaling_thread());
std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
GetSendersInternal() const;
@@ -313,9 +316,9 @@
GetReceiversInternal() const;
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
- GetAudioTransceiver() const;
+ GetAudioTransceiver() const RTC_RUN_ON(signaling_thread());
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
- GetVideoTransceiver() const;
+ GetVideoTransceiver() const RTC_RUN_ON(signaling_thread());
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
GetFirstAudioTransceiver() const;
@@ -328,16 +331,20 @@
const RtpSenderInfo& remote_sender_info)
RTC_RUN_ON(signaling_thread());
rtc::scoped_refptr<RtpReceiverInterface> RemoveAndStopReceiver(
- const RtpSenderInfo& remote_sender_info);
+ const RtpSenderInfo& remote_sender_info) RTC_RUN_ON(signaling_thread());
// May be called either by AddStream/RemoveStream, or when a track is
// added/removed from a stream previously added via AddStream.
- void AddAudioTrack(AudioTrackInterface* track, MediaStreamInterface* stream);
+ void AddAudioTrack(AudioTrackInterface* track, MediaStreamInterface* stream)
+ RTC_RUN_ON(signaling_thread());
void RemoveAudioTrack(AudioTrackInterface* track,
- MediaStreamInterface* stream);
- void AddVideoTrack(VideoTrackInterface* track, MediaStreamInterface* stream);
+ MediaStreamInterface* stream)
+ RTC_RUN_ON(signaling_thread());
+ void AddVideoTrack(VideoTrackInterface* track, MediaStreamInterface* stream)
+ RTC_RUN_ON(signaling_thread());
void RemoveVideoTrack(VideoTrackInterface* track,
- MediaStreamInterface* stream);
+ MediaStreamInterface* stream)
+ RTC_RUN_ON(signaling_thread());
// AddTrack implementation when Unified Plan is specified.
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrackUnifiedPlan(
@@ -346,7 +353,8 @@
// AddTrack implementation when Plan B is specified.
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrackPlanB(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
- const std::vector<std::string>& stream_ids);
+ const std::vector<std::string>& stream_ids)
+ RTC_RUN_ON(signaling_thread());
// Returns the first RtpTransceiver suitable for a newly added track, if such
// transceiver is available.
@@ -450,7 +458,7 @@
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
transceiver,
const cricket::ContentInfo& content,
- const cricket::ContentGroup* bundle_group);
+ const cricket::ContentGroup* bundle_group) RTC_RUN_ON(signaling_thread());
// Either creates or destroys the local data channel according to the given
// media section.
@@ -467,21 +475,25 @@
size_t mline_index,
const cricket::ContentInfo& content,
const cricket::ContentInfo* old_local_content,
- const cricket::ContentInfo* old_remote_content);
+ const cricket::ContentInfo* old_remote_content)
+ RTC_RUN_ON(signaling_thread());
// Returns the RtpTransceiver, if found, that is associated to the given MID.
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
- GetAssociatedTransceiver(const std::string& mid) const;
+ GetAssociatedTransceiver(const std::string& mid) const
+ RTC_RUN_ON(signaling_thread());
// Returns the RtpTransceiver, if found, that was assigned to the given mline
// index in CreateOffer.
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
- GetTransceiverByMLineIndex(size_t mline_index) const;
+ GetTransceiverByMLineIndex(size_t mline_index) const
+ RTC_RUN_ON(signaling_thread());
// Returns an RtpTransciever, if available, that can be used to receive the
// given media type according to JSEP rules.
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
- FindAvailableTransceiverToReceive(cricket::MediaType media_type) const;
+ FindAvailableTransceiverToReceive(cricket::MediaType media_type) const
+ RTC_RUN_ON(signaling_thread());
// Returns the media section in the given session description that is
// associated with the RtpTransceiver. Returns null if none found or this
@@ -490,7 +502,8 @@
const cricket::ContentInfo* FindMediaSectionForTransceiver(
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
transceiver,
- const SessionDescriptionInterface* sdesc) const;
+ const SessionDescriptionInterface* sdesc) const
+ RTC_RUN_ON(signaling_thread());
// Runs the algorithm **set the associated remote streams** specified in
// https://w3c.github.io/webrtc-pc/#set-associated-remote-streams.
@@ -530,17 +543,21 @@
// the local MediaStreams and DataChannels.
void GetOptionsForOffer(const PeerConnectionInterface::RTCOfferAnswerOptions&
offer_answer_options,
- cricket::MediaSessionOptions* session_options);
+ cricket::MediaSessionOptions* session_options)
+ RTC_RUN_ON(signaling_thread());
void GetOptionsForPlanBOffer(
const PeerConnectionInterface::RTCOfferAnswerOptions&
offer_answer_options,
- cricket::MediaSessionOptions* session_options);
+ cricket::MediaSessionOptions* session_options)
+ RTC_RUN_ON(signaling_thread());
void GetOptionsForUnifiedPlanOffer(
const PeerConnectionInterface::RTCOfferAnswerOptions&
offer_answer_options,
- cricket::MediaSessionOptions* session_options);
+ cricket::MediaSessionOptions* session_options)
+ RTC_RUN_ON(signaling_thread());
- RTCError HandleLegacyOfferOptions(const RTCOfferAnswerOptions& options);
+ RTCError HandleLegacyOfferOptions(const RTCOfferAnswerOptions& options)
+ RTC_RUN_ON(signaling_thread());
void RemoveRecvDirectionFromReceivingTransceiversOfType(
cricket::MediaType media_type);
void AddUpToOneReceivingTransceiverOfType(cricket::MediaType media_type);
@@ -551,15 +568,18 @@
// Returns a MediaSessionOptions struct with options decided by
// |constraints|, the local MediaStreams and DataChannels.
void GetOptionsForAnswer(const RTCOfferAnswerOptions& offer_answer_options,
- cricket::MediaSessionOptions* session_options);
+ cricket::MediaSessionOptions* session_options)
+ RTC_RUN_ON(signaling_thread());
void GetOptionsForPlanBAnswer(
const PeerConnectionInterface::RTCOfferAnswerOptions&
offer_answer_options,
- cricket::MediaSessionOptions* session_options);
+ cricket::MediaSessionOptions* session_options)
+ RTC_RUN_ON(signaling_thread());
void GetOptionsForUnifiedPlanAnswer(
const PeerConnectionInterface::RTCOfferAnswerOptions&
offer_answer_options,
- cricket::MediaSessionOptions* session_options);
+ cricket::MediaSessionOptions* session_options)
+ RTC_RUN_ON(signaling_thread());
// Generates MediaDescriptionOptions for the |session_opts| based on existing
// local description or remote description.
@@ -628,7 +648,8 @@
// For each new or removed StreamParam, OnLocalSenderSeen or
// OnLocalSenderRemoved is invoked.
void UpdateLocalSenders(const std::vector<cricket::StreamParams>& streams,
- cricket::MediaType media_type);
+ cricket::MediaType media_type)
+ RTC_RUN_ON(signaling_thread());
// Triggered when a local sender has been seen for the first time in a local
// session description.
@@ -636,7 +657,8 @@
// streams in the local SessionDescription can be mapped to a MediaStreamTrack
// in a MediaStream in |local_streams_|
void OnLocalSenderAdded(const RtpSenderInfo& sender_info,
- cricket::MediaType media_type);
+ cricket::MediaType media_type)
+ RTC_RUN_ON(signaling_thread());
// Triggered when a local sender has been removed from a local session
// description.
@@ -686,7 +708,7 @@
// 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 {
+ bool IsUnifiedPlan() const RTC_RUN_ON(signaling_thread()) {
return configuration_.sdp_semantics == SdpSemantics::kUnifiedPlan;
}
@@ -694,10 +716,12 @@
// unique. To support legacy end points that do not supply a=mid lines, this
// method will modify the session description to add MIDs generated according
// to the SDP semantics.
- void FillInMissingRemoteMids(cricket::SessionDescription* remote_description);
+ void FillInMissingRemoteMids(cricket::SessionDescription* remote_description)
+ RTC_RUN_ON(signaling_thread());
// Is there an RtpSender of the given type?
- bool HasRtpSender(cricket::MediaType type) const;
+ bool HasRtpSender(cricket::MediaType type) const
+ RTC_RUN_ON(signaling_thread());
// Return the RtpSender with the given track attached.
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
@@ -818,8 +842,8 @@
const cricket::SessionDescription* description);
// Push the media parts of the local or remote session description
// down to all of the channels.
- RTCError PushdownMediaDescription(SdpType type,
- cricket::ContentSource source);
+ RTCError PushdownMediaDescription(SdpType type, cricket::ContentSource source)
+ RTC_RUN_ON(signaling_thread());
bool PushdownSctpParameters_n(cricket::ContentSource source);
RTCError PushdownTransportDescription(cricket::ContentSource source,
@@ -854,12 +878,14 @@
RTC_RUN_ON(signaling_thread());
// Deletes the corresponding channel of contents that don't exist in |desc|.
// |desc| can be null. This means that all channels are deleted.
- void RemoveUnusedChannels(const cricket::SessionDescription* desc);
+ void RemoveUnusedChannels(const cricket::SessionDescription* desc)
+ RTC_RUN_ON(signaling_thread());
// Allocates media channels based on the |desc|. If |desc| doesn't have
// the BUNDLE option, this method will disable BUNDLE in PortAllocator.
// This method will also delete any existing media channels before creating.
- RTCError CreateChannels(const cricket::SessionDescription& desc);
+ RTCError CreateChannels(const cricket::SessionDescription& desc)
+ RTC_RUN_ON(signaling_thread());
// If the BUNDLE policy is max-bundle, then we know for sure that all
// transports will be bundled from the start. This method returns the BUNDLE
@@ -867,12 +893,15 @@
// error is returned if max-bundle is specified but the session description
// does not have a BUNDLE group.
RTCErrorOr<const cricket::ContentGroup*> GetEarlyBundleGroup(
- const cricket::SessionDescription& desc) const;
+ const cricket::SessionDescription& desc) const
+ RTC_RUN_ON(signaling_thread());
// Helper methods to create media channels.
- cricket::VoiceChannel* CreateVoiceChannel(const std::string& mid);
- cricket::VideoChannel* CreateVideoChannel(const std::string& mid);
- bool CreateDataChannel(const std::string& mid);
+ cricket::VoiceChannel* CreateVoiceChannel(const std::string& mid)
+ RTC_RUN_ON(signaling_thread());
+ cricket::VideoChannel* CreateVideoChannel(const std::string& mid)
+ RTC_RUN_ON(signaling_thread());
+ bool CreateDataChannel(const std::string& mid) RTC_RUN_ON(signaling_thread());
bool CreateSctpTransport_n(const std::string& mid);
// For bundling.
@@ -901,7 +930,8 @@
bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
// Below methods are helper methods which verifies SDP.
RTCError ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
- cricket::ContentSource source);
+ cricket::ContentSource source)
+ RTC_RUN_ON(signaling_thread());
// Check if a call to SetLocalDescription is acceptable with a session
// description of the given type.
@@ -996,7 +1026,7 @@
// 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();
+ CryptoOptions GetCryptoOptions() RTC_RUN_ON(signaling_thread());
// Returns rtp transport, result can not be nullptr.
RtpTransportInternal* GetRtpTransport(const std::string& mid) {
@@ -1007,7 +1037,8 @@
// Returns media transport, if PeerConnection was created with configuration
// to use media transport. Otherwise returns nullptr.
- MediaTransportInterface* GetMediaTransport(const std::string& mid) {
+ MediaTransportInterface* GetMediaTransport(const std::string& mid)
+ RTC_RUN_ON(signaling_thread()) {
auto media_transport = transport_controller_->GetMediaTransport(mid);
RTC_DCHECK((configuration_.use_media_transport ||
configuration_.use_media_transport_for_data_channels) ==
@@ -1050,7 +1081,16 @@
IceGatheringState ice_gathering_state_ RTC_GUARDED_BY(signaling_thread()) =
kIceGatheringNew;
- PeerConnectionInterface::RTCConfiguration configuration_;
+ PeerConnectionInterface::RTCConfiguration configuration_
+ RTC_GUARDED_BY(signaling_thread());
+
+ // Cache configuration_.use_media_transport so that we can access it from
+ // other threads.
+ // TODO(bugs.webrtc.org/9987): Caching just this bool and allowing the data
+ // it's derived from to change is not necessarily sound. Stop doing it.
+ rtc::RaceChecker use_media_transport_race_checker_;
+ bool use_media_transport_ RTC_GUARDED_BY(use_media_transport_race_checker_) =
+ configuration_.use_media_transport;
// TODO(zstein): |async_resolver_factory_| can currently be nullptr if it
// is not injected. It should be required once chromium supplies it.
