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Delete almost all default methods on PeerConnectionInterface

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Keeping default implementations only for methods involved in
ongoing transitions.

Intended to catch inconsistencies between the interface and the
PeerConnectionProxy class, at compile time.

Bug: webrtc:10716
Change-Id: I4cb126c353855f7288ba09273fa6f87aaa0f32eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140860
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29224}
This commit is contained in:
Niels Möller
2019-09-13 15:41:21 +02:00
committed by Commit Bot
parent e607a06338
commit 7b04a91f4a
5 changed files with 63 additions and 159 deletions
Download Patch File Download Diff File Expand all files Collapse all files

119
api/peer_connection_interface.cc
View File

@ -42,103 +42,17 @@ PeerConnectionInterface::RTCConfiguration::RTCConfiguration(
PeerConnectionInterface::RTCConfiguration::~RTCConfiguration() = default;
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>
PeerConnectionInterface::AddTrack(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids) {
return RTCError(RTCErrorType::UNSUPPORTED_OPERATION, "Not implemented");
}
bool PeerConnectionInterface::RemoveTrack(RtpSenderInterface* sender) {
return RemoveTrackNew(sender).ok();
}
RTCError PeerConnectionInterface::RemoveTrackNew(
rtc::scoped_refptr<RtpSenderInterface> sender) {
return RTCError(RemoveTrack(sender) ? RTCErrorType::NONE
: RTCErrorType::INTERNAL_ERROR);
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnectionInterface::AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track) {
return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnectionInterface::AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init) {
return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnectionInterface::AddTransceiver(cricket::MediaType media_type) {
return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnectionInterface::AddTransceiver(cricket::MediaType media_type,
const RtpTransceiverInit& init) {
return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
}
rtc::scoped_refptr<RtpSenderInterface> PeerConnectionInterface::CreateSender(
const std::string& kind,
const std::string& stream_id) {
return rtc::scoped_refptr<RtpSenderInterface>();
}
std::vector<rtc::scoped_refptr<RtpSenderInterface>>
PeerConnectionInterface::GetSenders() const {
return std::vector<rtc::scoped_refptr<RtpSenderInterface>>();
}
std::vector<rtc::scoped_refptr<RtpReceiverInterface>>
PeerConnectionInterface::GetReceivers() const {
return std::vector<rtc::scoped_refptr<RtpReceiverInterface>>();
}
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnectionInterface::GetTransceivers() const {
return std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>();
}
const SessionDescriptionInterface*
PeerConnectionInterface::current_local_description() const {
return nullptr;
}
const SessionDescriptionInterface*
PeerConnectionInterface::current_remote_description() const {
return nullptr;
}
const SessionDescriptionInterface*
PeerConnectionInterface::pending_local_description() const {
return nullptr;
}
const SessionDescriptionInterface*
PeerConnectionInterface::pending_remote_description() const {
return nullptr;
}
PeerConnectionInterface::RTCConfiguration
PeerConnectionInterface::GetConfiguration() {
return PeerConnectionInterface::RTCConfiguration();
}
RTCError PeerConnectionInterface::SetConfiguration(
const PeerConnectionInterface::RTCConfiguration& config) {
return RTCError();
}
bool PeerConnectionInterface::RemoveIceCandidates(
const std::vector<cricket::Candidate>& candidates) {
return false;
}
RTCError PeerConnectionInterface::SetBitrate(const BitrateSettings& bitrate) {
BitrateParameters bitrate_parameters;
bitrate_parameters.min_bitrate_bps = bitrate.min_bitrate_bps;
@ -156,39 +70,6 @@ RTCError PeerConnectionInterface::SetBitrate(
return SetBitrate(bitrate);
}
PeerConnectionInterface::IceConnectionState
PeerConnectionInterface::standardized_ice_connection_state() {
return PeerConnectionInterface::IceConnectionState::kIceConnectionFailed;
}
PeerConnectionInterface::PeerConnectionState
PeerConnectionInterface::peer_connection_state() {
return PeerConnectionInterface::PeerConnectionState::kFailed;
}
bool PeerConnectionInterface::StartRtcEventLog(
std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms) {
return false;
}
bool PeerConnectionInterface::StartRtcEventLog(
std::unique_ptr<RtcEventLogOutput> output) {
return false;
}
rtc::scoped_refptr<DtlsTransportInterface>
PeerConnectionInterface::LookupDtlsTransportByMid(const std::string& mid) {
RTC_NOTREACHED();
return nullptr;
}
rtc::scoped_refptr<SctpTransportInterface>
PeerConnectionInterface::GetSctpTransport() const {
RTC_NOTREACHED();
return nullptr;
}
PeerConnectionInterface::BitrateParameters::BitrateParameters() = default;
PeerConnectionInterface::BitrateParameters::~BitrateParameters() = default;

74
api/peer_connection_interface.h
View File

@ -760,12 +760,12 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// - INVALID_STATE: The PeerConnection is closed.
virtual RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrack(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids);
const std::vector<std::string>& stream_ids) = 0;
// Remove an RtpSender from this PeerConnection.
// Returns true on success.
// TODO(steveanton): Replace with signature that returns RTCError.
virtual bool RemoveTrack(RtpSenderInterface* sender);
virtual bool RemoveTrack(RtpSenderInterface* sender) = 0;
// Plan B semantics: Removes the RtpSender from this PeerConnection.
// Unified Plan semantics: Stop sending on the RtpSender and mark the
@ -798,8 +798,6 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
//
// Common errors:
// - INTERNAL_ERROR: The configuration does not have Unified Plan enabled.
// TODO(steveanton): Make these pure virtual once downstream projects have
// updated.
// Adds a transceiver with a sender set to transmit the given track. The kind
// of the transceiver (and sender/receiver) will be derived from the kind of
@ -807,10 +805,10 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// Errors:
// - INVALID_PARAMETER: |track| is null.
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track);
AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track) = 0;
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init);
const RtpTransceiverInit& init) = 0;
// Adds a transceiver with the given kind. Can either be MEDIA_TYPE_AUDIO or
// MEDIA_TYPE_VIDEO.
@ -818,11 +816,10 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// - INVALID_PARAMETER: |media_type| is not MEDIA_TYPE_AUDIO or
// MEDIA_TYPE_VIDEO.
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
AddTransceiver(cricket::MediaType media_type);
AddTransceiver(cricket::MediaType media_type) = 0;
virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
AddTransceiver(cricket::MediaType media_type, const RtpTransceiverInit& init);
// TODO(deadbeef): Make these pure virtual once all subclasses implement them.
AddTransceiver(cricket::MediaType media_type,
const RtpTransceiverInit& init) = 0;
// Creates a sender without a track. Can be used for "early media"/"warmup"
// use cases, where the application may want to negotiate video attributes
@ -840,7 +837,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// AddTransceiver instead.
virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
const std::string& kind,
const std::string& stream_id);
const std::string& stream_id) = 0;
// If Plan B semantics are specified, gets all RtpSenders, created either
// through AddStream, AddTrack, or CreateSender. All senders of a specific
@ -849,7 +846,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// If Unified Plan semantics are specified, gets the RtpSender for each
// RtpTransceiver.
virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
const;
const = 0;
// If Plan B semantics are specified, gets all RtpReceivers created when a
// remote description is applied. All receivers of a specific media type share
@ -860,7 +857,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// If Unified Plan semantics are specified, gets the RtpReceiver for each
// RtpTransceiver.
virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
const;
const = 0;
// Get all RtpTransceivers, created either through AddTransceiver, AddTrack or
// by a remote description applied with SetRemoteDescription.
@ -868,7 +865,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// Note: This method is only available when Unified Plan is enabled (see
// RTCConfiguration).
virtual std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
GetTransceivers() const;
GetTransceivers() const = 0;
// The legacy non-compliant GetStats() API. This correspond to the
// callback-based version of getStats() in JavaScript. The returned metrics
@ -897,19 +894,17 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// requires stop overriding the current version in third party or making third
// party calls explicit to avoid ambiguity during switch. Make the future
// version abstract as soon as third party projects implement it.
virtual void GetStats(RTCStatsCollectorCallback* callback) {}
virtual void GetStats(RTCStatsCollectorCallback* callback) = 0;
// Spec-compliant getStats() performing the stats selection algorithm with the
// sender. https://w3c.github.io/webrtc-pc/#dom-rtcrtpsender-getstats
// TODO(hbos): Make abstract as soon as third party projects implement it.
virtual void GetStats(
rtc::scoped_refptr<RtpSenderInterface> selector,
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {}
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) = 0;
// Spec-compliant getStats() performing the stats selection algorithm with the
// receiver. https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-getstats
// TODO(hbos): Make abstract as soon as third party projects implement it.
virtual void GetStats(
rtc::scoped_refptr<RtpReceiverInterface> selector,
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {}
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) = 0;
// Clear cached stats in the RTCStatsCollector.
// Exposed for testing while waiting for automatic cache clear to work.
// https://bugs.webrtc.org/8693
@ -933,14 +928,18 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// A "current" description the one currently negotiated from a complete
// offer/answer exchange.
virtual const SessionDescriptionInterface* current_local_description() const;
virtual const SessionDescriptionInterface* current_remote_description() const;
virtual const SessionDescriptionInterface* current_local_description()
const = 0;
virtual const SessionDescriptionInterface* current_remote_description()
const = 0;
// A "pending" description is one that's part of an incomplete offer/answer
// exchange (thus, either an offer or a pranswer). Once the offer/answer
// exchange is finished, the "pending" description will become "current".
virtual const SessionDescriptionInterface* pending_local_description() const;
virtual const SessionDescriptionInterface* pending_remote_description() const;
virtual const SessionDescriptionInterface* pending_local_description()
const = 0;
virtual const SessionDescriptionInterface* pending_remote_description()
const = 0;
// Tells the PeerConnection that ICE should be restarted. This triggers a need
// for negotiation and subsequent CreateOffer() calls will act as if
@ -948,7 +947,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-restartice
// TODO(hbos): Remove default implementation when downstream projects
// implement this.
virtual void RestartIce() {}
virtual void RestartIce() = 0;
// Create a new offer.
// The CreateSessionDescriptionObserver callback will be called when done.
@ -973,14 +972,11 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// TODO(hbos): Remove when Chrome implements the new signature.
virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) {}
// TODO(hbos): Make pure virtual when Chrome has updated its signature.
virtual void SetRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {}
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) = 0;
// TODO(deadbeef): Make this pure virtual once all Chrome subclasses of
// PeerConnectionInterface implement it.
virtual PeerConnectionInterface::RTCConfiguration GetConfiguration();
virtual PeerConnectionInterface::RTCConfiguration GetConfiguration() = 0;
// Sets the PeerConnection's global configuration to |config|.
//
@ -1018,7 +1014,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// continual gathering, to avoid an ever-growing list of candidates as
// networks come and go.
virtual bool RemoveIceCandidates(
const std::vector<cricket::Candidate>& candidates);
const std::vector<cricket::Candidate>& candidates) = 0;
// 0 <= min <= current <= max should hold for set parameters.
struct BitrateParameters {
@ -1062,13 +1058,12 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// In the Javascript API, DtlsTransport is a property of a sender, but
// because the PeerConnection owns the DtlsTransport in this implementation,
// it is better to look them up on the PeerConnection.
// TODO(hta): Remove default implementation after updating Chrome.
virtual rtc::scoped_refptr<DtlsTransportInterface> LookupDtlsTransportByMid(
const std::string& mid);
const std::string& mid) = 0;
// Returns the SCTP transport, if any.
// TODO(hta): Remove default implementation after updating Chrome.
virtual rtc::scoped_refptr<SctpTransportInterface> GetSctpTransport() const;
virtual rtc::scoped_refptr<SctpTransportInterface> GetSctpTransport()
const = 0;
// Returns the current SignalingState.
virtual SignalingState signaling_state() = 0;
@ -1080,10 +1075,10 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
virtual IceConnectionState ice_connection_state() = 0;
// Returns an aggregated state of all ICE transports.
virtual IceConnectionState standardized_ice_connection_state();
virtual IceConnectionState standardized_ice_connection_state() = 0;
// Returns an aggregated state of all ICE and DTLS transports.
virtual PeerConnectionState peer_connection_state();
virtual PeerConnectionState peer_connection_state() = 0;
virtual IceGatheringState ice_gathering_state() = 0;
@ -1097,12 +1092,11 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
// lost in case the application crashes. If the |output_period_ms| argument is
// omitted, webrtc selects a default deemed to be workable in most cases.
virtual bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms);
virtual bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output);
int64_t output_period_ms) = 0;
virtual bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output) = 0;
// Stops logging the RtcEventLog.
// TODO(ivoc): Make this pure virtual when Chrome is updat ed.
virtual void StopRtcEventLog() {}
virtual void StopRtcEventLog() = 0;
// Terminates all media, closes the transports, and in general releases any
// resources used by the PeerConnection. This is an irreversible operation.

1
api/peer_connection_proxy.h
View File

@ -72,6 +72,7 @@ PROXY_METHOD2(void,
GetStats,
rtc::scoped_refptr<RtpReceiverInterface>,
rtc::scoped_refptr<RTCStatsCollectorCallback>)
PROXY_METHOD0(void, ClearStatsCache)
PROXY_METHOD2(rtc::scoped_refptr<DataChannelInterface>,
CreateDataChannel,
const std::string&,

11
api/test/mock_peerconnectioninterface.h
View File

@ -13,6 +13,7 @@
#include <memory>
#include <string>
#include <type_traits>
#include <utility>
#include <vector>
@ -39,6 +40,8 @@ class MockPeerConnectionInterface
MediaStreamTrackInterface*,
std::vector<MediaStreamInterface*>));
MOCK_METHOD1(RemoveTrack, bool(RtpSenderInterface*));
MOCK_METHOD1(RemoveTrackNew,
RTCError(rtc::scoped_refptr<RtpSenderInterface>));
MOCK_METHOD1(AddTransceiver,
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>(
rtc::scoped_refptr<MediaStreamTrackInterface>));
@ -91,6 +94,7 @@ class MockPeerConnectionInterface
const SessionDescriptionInterface*());
MOCK_CONST_METHOD0(pending_remote_description,
const SessionDescriptionInterface*());
MOCK_METHOD0(RestartIce, void());
MOCK_METHOD2(CreateOffer,
void(CreateSessionDescriptionObserver*,
const RTCOfferAnswerOptions&));
@ -116,15 +120,22 @@ class MockPeerConnectionInterface
MOCK_METHOD1(SetBitrate, RTCError(const BitrateParameters&));
MOCK_METHOD1(SetAudioPlayout, void(bool));
MOCK_METHOD1(SetAudioRecording, void(bool));
MOCK_METHOD1(LookupDtlsTransportByMid,
rtc::scoped_refptr<DtlsTransportInterface>(const std::string&));
MOCK_METHOD0(signaling_state, SignalingState());
MOCK_METHOD0(ice_connection_state, IceConnectionState());
MOCK_METHOD0(standardized_ice_connection_state, IceConnectionState());
MOCK_METHOD0(peer_connection_state, PeerConnectionState());
MOCK_METHOD0(ice_gathering_state, IceGatheringState());
MOCK_METHOD2(StartRtcEventLog,
bool(std::unique_ptr<RtcEventLogOutput>, int64_t));
MOCK_METHOD1(StartRtcEventLog, bool(std::unique_ptr<RtcEventLogOutput>));
MOCK_METHOD0(StopRtcEventLog, void());
MOCK_METHOD0(Close, void());
};
static_assert(!std::is_abstract<MockPeerConnectionInterface>::value, "");
} // namespace webrtc
#endif // API_TEST_MOCK_PEERCONNECTIONINTERFACE_H_

17
pc/test/fake_peer_connection_base.h
View File

@ -52,6 +52,11 @@ class FakePeerConnectionBase : public PeerConnectionInternal {
bool RemoveTrack(RtpSenderInterface* sender) override { return false; }
RTCError RemoveTrackNew(
rtc::scoped_refptr<RtpSenderInterface> sender) override {
return RTCError(RTCErrorType::UNSUPPORTED_OPERATION);
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track) override {
return RTCError(RTCErrorType::UNSUPPORTED_OPERATION, "Not implemented");
@ -200,6 +205,14 @@ class FakePeerConnectionBase : public PeerConnectionInternal {
return IceConnectionState::kIceConnectionNew;
}
IceConnectionState standardized_ice_connection_state() override {
return IceConnectionState::kIceConnectionNew;
}
PeerConnectionState peer_connection_state() override {
return PeerConnectionState::kNew;
}
IceGatheringState ice_gathering_state() override {
return IceGatheringState::kIceGatheringNew;
}
@ -209,6 +222,10 @@ class FakePeerConnectionBase : public PeerConnectionInternal {
return false;
}
bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output) override {
return false;
}
void StopRtcEventLog() override {}
void Close() override {}