Remove friendship between PeerConnection and SdpOfferAnswerHandler
Add multiple accessors to PeerConnection, and make multiple
formerly private functions public for access from SdpOfferAnswerHandler.
Reducing the surface of PeerConnection is a job to be done iteratively.
Bug: webrtc:11995
Change-Id: Iab176824ae557af84ac934e40ff674a1008a29d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189540
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32459}
diff --git a/pc/peer_connection.h b/pc/peer_connection.h
index 582c76d..dd7a3b4a 100644
--- a/pc/peer_connection.h
+++ b/pc/peer_connection.h
@@ -371,6 +371,97 @@
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_; }
+ rtc::UniqueRandomIdGenerator* ssrc_generator() { return &ssrc_generator_; }
+ const cricket::AudioOptions& audio_options() { return audio_options_; }
+ const cricket::VideoOptions& video_options() { return video_options_; }
+ VideoBitrateAllocatorFactory* video_bitrate_allocator_factory() {
+ return video_bitrate_allocator_factory_.get();
+ }
+
+ cricket::DataChannelType data_channel_type() const;
+ void SetIceConnectionState(IceConnectionState new_state);
+ void NoteUsageEvent(UsageEvent event);
+
+ // Report the UMA metric SdpFormatReceived for the given remote offer.
+ void ReportSdpFormatReceived(const SessionDescriptionInterface& remote_offer);
+ // Signals from MediaStreamObserver.
+ void OnAudioTrackAdded(AudioTrackInterface* track,
+ MediaStreamInterface* stream)
+ RTC_RUN_ON(signaling_thread());
+ void OnAudioTrackRemoved(AudioTrackInterface* track,
+ MediaStreamInterface* stream)
+ RTC_RUN_ON(signaling_thread());
+ void OnVideoTrackAdded(VideoTrackInterface* track,
+ MediaStreamInterface* stream)
+ RTC_RUN_ON(signaling_thread());
+ void OnVideoTrackRemoved(VideoTrackInterface* track,
+ MediaStreamInterface* stream)
+ RTC_RUN_ON(signaling_thread());
+
+ // 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 configuration_.sdp_semantics == SdpSemantics::kUnifiedPlan;
+ }
+ bool ValidateBundleSettings(const cricket::SessionDescription* desc);
+
+ // Returns the MID for the data section associated with either the
+ // RtpDataChannel or SCTP data channel, if it has been set. If no data
+ // channels are configured this will return nullopt.
+ absl::optional<std::string> GetDataMid() const;
+
+ void SetSctpDataMid(const std::string& mid) {
+ RTC_DCHECK_RUN_ON(signaling_thread());
+ sctp_mid_s_ = mid;
+ }
+ void ResetSctpDataMid() {
+ RTC_DCHECK_RUN_ON(signaling_thread());
+ sctp_mid_s_.reset();
+ }
+
+ // 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.
+ RtpTransportInternal* GetRtpTransport(const std::string& mid) {
+ RTC_DCHECK_RUN_ON(signaling_thread());
+ auto rtp_transport = transport_controller_->GetRtpTransport(mid);
+ RTC_DCHECK(rtp_transport);
+ return rtp_transport;
+ }
+
+ // Returns true if SRTP (either using DTLS-SRTP or SDES) is required by
+ // this session.
+ bool SrtpRequired() const RTC_RUN_ON(signaling_thread());
+
+ void OnSentPacket_w(const rtc::SentPacket& sent_packet);
+
+ 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);
// Functions made public for testing.
void ReturnHistogramVeryQuicklyForTesting() {
@@ -383,23 +474,10 @@
~PeerConnection() override;
private:
- // While refactoring: Allow access from SDP negotiation
- // TOOD(https://bugs.webrtc.org/11995): Remove friendship.
- friend class SdpOfferAnswerHandler;
-
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
FindTransceiverBySender(rtc::scoped_refptr<RtpSenderInterface> sender)
RTC_RUN_ON(signaling_thread());
- // 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);
-
- void SetIceConnectionState(IceConnectionState new_state);
void SetStandardizedIceConnectionState(
PeerConnectionInterface::IceConnectionState new_state)
RTC_RUN_ON(signaling_thread());
@@ -428,39 +506,10 @@
const cricket::CandidatePairChangeEvent& event)
RTC_RUN_ON(signaling_thread());
- // Signals from MediaStreamObserver.
- void OnAudioTrackAdded(AudioTrackInterface* track,
- MediaStreamInterface* stream)
- RTC_RUN_ON(signaling_thread());
- void OnAudioTrackRemoved(AudioTrackInterface* track,
- MediaStreamInterface* stream)
- RTC_RUN_ON(signaling_thread());
- void OnVideoTrackAdded(VideoTrackInterface* track,
- MediaStreamInterface* stream)
- RTC_RUN_ON(signaling_thread());
- void OnVideoTrackRemoved(VideoTrackInterface* track,
- MediaStreamInterface* stream)
- RTC_RUN_ON(signaling_thread());
void OnNegotiationNeeded();
- // Returns the MID for the data section associated with either the
- // RtpDataChannel or SCTP data channel, if it has been set. If no data
- // channels are configured this will return nullopt.
- absl::optional<std::string> GetDataMid() const;
-
- // 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 configuration_.sdp_semantics == SdpSemantics::kUnifiedPlan;
- }
-
// Returns the specified SCTP DataChannel in sctp_data_channels_,
// or nullptr if not found.
SctpDataChannel* FindDataChannelBySid(int sid) const
@@ -501,14 +550,9 @@
// Returns RTCError::OK() if there are no issues.
RTCError ValidateConfiguration(const RTCConfiguration& config) const;
- cricket::ChannelManager* channel_manager() const;
-
- cricket::ChannelInterface* GetChannel(const std::string& content_name);
-
cricket::IceConfig ParseIceConfig(
const PeerConnectionInterface::RTCConfiguration& config) const;
- cricket::DataChannelType data_channel_type() const;
// Called when an RTCCertificate is generated or retrieved by
// WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
@@ -529,21 +573,12 @@
int* sdp_mline_index)
RTC_RUN_ON(signaling_thread());
- bool SetupDataChannelTransport_n(const std::string& mid)
- RTC_RUN_ON(network_thread());
- void TeardownDataChannelTransport_n() RTC_RUN_ON(network_thread());
-
- bool ValidateBundleSettings(const cricket::SessionDescription* desc);
bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
// Verifies a=setup attribute as per RFC 5763.
bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
SdpType type);
- // Returns true if SRTP (either using DTLS-SRTP or SDES) is required by
- // this session.
- bool SrtpRequired() const RTC_RUN_ON(signaling_thread());
-
// JsepTransportController signal handlers.
void OnTransportControllerConnectionState(cricket::IceConnectionState state)
RTC_RUN_ON(signaling_thread());
@@ -564,9 +599,6 @@
RTC_RUN_ON(signaling_thread());
void OnTransportControllerDtlsHandshakeError(rtc::SSLHandshakeError error);
- // Report the UMA metric SdpFormatReceived for the given remote offer.
- void ReportSdpFormatReceived(const SessionDescriptionInterface& remote_offer);
-
// Invoked when TransportController connection completion is signaled.
// Reports stats for all transports in use.
void ReportTransportStats() RTC_RUN_ON(signaling_thread());
@@ -580,11 +612,8 @@
void ReportIceCandidateCollected(const cricket::Candidate& candidate)
RTC_RUN_ON(signaling_thread());
- void NoteUsageEvent(UsageEvent event);
void ReportUsagePattern() const RTC_RUN_ON(signaling_thread());
- void OnSentPacket_w(const rtc::SentPacket& sent_packet);
-
// JsepTransportController::Observer override.
//
// Called by |transport_controller_| when processing transport information
@@ -597,19 +626,6 @@
rtc::scoped_refptr<DtlsTransport> dtls_transport,
DataChannelTransportInterface* data_channel_transport) override;
- // 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();
-
- // Returns rtp transport, result can not be nullptr.
- RtpTransportInternal* GetRtpTransport(const std::string& mid) {
- RTC_DCHECK_RUN_ON(signaling_thread());
- auto rtp_transport = transport_controller_->GetRtpTransport(mid);
- RTC_DCHECK(rtp_transport);
- return rtp_transport;
- }
-
std::function<void(const rtc::CopyOnWriteBuffer& packet,
int64_t packet_time_us)>
InitializeRtcpCallback();
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index b40d00c..81df99a 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -962,20 +962,20 @@
return pc_->rtp_manager()->transceivers();
}
JsepTransportController* SdpOfferAnswerHandler::transport_controller() {
- return pc_->transport_controller_.get();
+ return pc_->transport_controller();
}
DataChannelController* SdpOfferAnswerHandler::data_channel_controller() {
- return &pc_->data_channel_controller_;
+ return pc_->data_channel_controller();
}
const DataChannelController* SdpOfferAnswerHandler::data_channel_controller()
const {
- return &pc_->data_channel_controller_;
+ return pc_->data_channel_controller();
}
cricket::PortAllocator* SdpOfferAnswerHandler::port_allocator() {
- return pc_->port_allocator_.get();
+ return pc_->port_allocator();
}
const cricket::PortAllocator* SdpOfferAnswerHandler::port_allocator() const {
- return pc_->port_allocator_.get();
+ return pc_->port_allocator();
}
RtpTransmissionManager* SdpOfferAnswerHandler::rtp_manager() {
return pc_->rtp_manager();
@@ -4509,9 +4509,9 @@
{
RTC_DCHECK_RUN_ON(pc_->signaling_thread());
voice_channel = channel_manager()->CreateVoiceChannel(
- pc_->call_ptr_, pc_->configuration_.media_config, rtp_transport,
+ pc_->call_ptr(), pc_->configuration()->media_config, rtp_transport,
signaling_thread(), mid, pc_->SrtpRequired(), pc_->GetCryptoOptions(),
- &pc_->ssrc_generator_, pc_->audio_options_);
+ pc_->ssrc_generator(), pc_->audio_options());
}
if (!voice_channel) {
return nullptr;
@@ -4536,10 +4536,10 @@
{
RTC_DCHECK_RUN_ON(pc_->signaling_thread());
video_channel = channel_manager()->CreateVideoChannel(
- pc_->call_ptr_, pc_->configuration_.media_config, rtp_transport,
+ pc_->call_ptr(), pc_->configuration()->media_config, rtp_transport,
signaling_thread(), mid, pc_->SrtpRequired(), pc_->GetCryptoOptions(),
- &pc_->ssrc_generator_, pc_->video_options_,
- pc_->video_bitrate_allocator_factory_.get());
+ pc_->ssrc_generator(), pc_->video_options(),
+ pc_->video_bitrate_allocator_factory());
}
if (!video_channel) {
return nullptr;
@@ -4559,10 +4559,7 @@
RTC_FROM_HERE,
rtc::Bind(&PeerConnection::SetupDataChannelTransport_n, pc_,
mid))) {
- {
- RTC_DCHECK_RUN_ON(pc_->signaling_thread());
- pc_->sctp_mid_s_ = mid;
- }
+ pc_->SetSctpDataMid(mid);
} else {
return false;
}
@@ -4576,9 +4573,9 @@
RTC_DCHECK_RUN_ON(pc_->signaling_thread());
data_channel_controller()->set_rtp_data_channel(
channel_manager()->CreateRtpDataChannel(
- pc_->configuration_.media_config, rtp_transport,
+ pc_->configuration()->media_config, rtp_transport,
signaling_thread(), mid, pc_->SrtpRequired(),
- pc_->GetCryptoOptions(), &pc_->ssrc_generator_));
+ pc_->GetCryptoOptions(), pc_->ssrc_generator()));
}
if (!data_channel_controller()->rtp_data_channel()) {
return false;
@@ -4626,7 +4623,7 @@
RTC_DCHECK_RUN_ON(pc_->network_thread());
pc_->TeardownDataChannelTransport_n();
});
- pc_->sctp_mid_s_.reset();
+ pc_->ResetSctpDataMid();
}
}
