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Remove friendship between PeerConnection and SdpOfferAnswerHandler

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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}
This commit is contained in:
Harald Alvestrand 2020-10-19 16:05:20 +00:00 committed by Commit Bot
parent e2c548503b
commit 653429c5b3
2 changed files with 105 additions and 92 deletions
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166
pc/peer_connection.h
View File

@ -371,6 +371,97 @@ class PeerConnection : public PeerConnectionInternal,
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 @@ class PeerConnection : public PeerConnectionInternal,
~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 @@ class PeerConnection : public PeerConnectionInternal,
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 @@ class PeerConnection : public PeerConnectionInternal,
// 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 @@ class PeerConnection : public PeerConnectionInternal,
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 @@ class PeerConnection : public PeerConnectionInternal,
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 @@ class PeerConnection : public PeerConnectionInternal,
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 @@ class PeerConnection : public PeerConnectionInternal,
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();

31
pc/sdp_offer_answer.cc
View File

@ -962,20 +962,20 @@ const TransceiverList* SdpOfferAnswerHandler::transceivers() const {
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 @@ cricket::VoiceChannel* SdpOfferAnswerHandler::CreateVoiceChannel(
{
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 @@ cricket::VideoChannel* SdpOfferAnswerHandler::CreateVideoChannel(
{
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 @@ bool SdpOfferAnswerHandler::CreateDataChannel(const std::string& mid) {
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 @@ bool SdpOfferAnswerHandler::CreateDataChannel(const std::string& mid) {
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 @@ void SdpOfferAnswerHandler::DestroyDataChannelTransport() {
RTC_DCHECK_RUN_ON(pc_->network_thread());
pc_->TeardownDataChannelTransport_n();
});
pc_->sctp_mid_s_.reset();
pc_->ResetSctpDataMid();
}
}