webrtc_m130/pc/peer_connection_wrapper.h
Philipp Hancke cfaba8fd2d Measure SDP munging
by storing
  [[LastCreatedOffer]] / [[LastCreatedAnswer]]
which are similar to the W3C equivalent but as
description objects instead of serialized SDP strings.

While rejecting all SDP munging is not feasible, this lets us
measure and reject certain modifications gradually.

Chromium metrics CL:
  https://chromium-review.googlesource.com/c/chromium/src/+/6089633

This is measured at three points during the lifetime of a peerconnection:
* for the first SLD call
* when the connection is first established
* when the connection was established and is being closed

Note that the "first" SDP munging detected is returned which may hide that something uses more than one modification.

BUG=chromium:40567530

Change-Id: I964e3ee6e75f73b777d90556fac8691a6f3dc27f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370680
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43741}
2025-01-15 07:38:45 -08:00

207 lines
9.3 KiB
C++

/*
* Copyright 2017 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_WRAPPER_H_
#define PC_PEER_CONNECTION_WRAPPER_H_
#include <memory>
#include <optional>
#include <string>
#include <vector>
#include "api/data_channel_interface.h"
#include "api/function_view.h"
#include "api/jsep.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_error.h"
#include "api/rtp_parameters.h"
#include "api/rtp_sender_interface.h"
#include "api/rtp_transceiver_interface.h"
#include "api/scoped_refptr.h"
#include "api/stats/rtc_stats_report.h"
#include "pc/test/mock_peer_connection_observers.h"
namespace webrtc {
// Class that wraps a PeerConnection so that it is easier to use in unit tests.
// Namely, gives a synchronous API for the event-callback-based API of
// PeerConnection and provides an observer object that stores information from
// PeerConnectionObserver callbacks.
//
// This is intended to be subclassed if additional information needs to be
// stored with the PeerConnection (e.g., fake PeerConnection parameters so that
// tests can be written against those interactions). The base
// PeerConnectionWrapper should only have helper methods that are broadly
// useful. More specific helper methods should be created in the test-specific
// subclass.
//
// The wrapper is intended to be constructed by specialized factory methods on
// a test fixture class then used as a local variable in each test case.
class PeerConnectionWrapper {
public:
// Constructs a PeerConnectionWrapper from the given PeerConnection.
// The given PeerConnectionFactory should be the factory that created the
// PeerConnection and the MockPeerConnectionObserver should be the observer
// that is watching the PeerConnection.
PeerConnectionWrapper(
rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory,
rtc::scoped_refptr<PeerConnectionInterface> pc,
std::unique_ptr<MockPeerConnectionObserver> observer);
virtual ~PeerConnectionWrapper();
PeerConnectionFactoryInterface* pc_factory();
PeerConnectionInterface* pc();
MockPeerConnectionObserver* observer();
// Calls the underlying PeerConnection's CreateOffer method and returns the
// resulting SessionDescription once it is available. If the method call
// failed, null is returned.
std::unique_ptr<SessionDescriptionInterface> CreateOffer(
const PeerConnectionInterface::RTCOfferAnswerOptions& options,
std::string* error_out = nullptr);
// Calls CreateOffer with default options.
std::unique_ptr<SessionDescriptionInterface> CreateOffer();
// Calls CreateOffer and sets a copy of the offer as the local description.
std::unique_ptr<SessionDescriptionInterface> CreateOfferAndSetAsLocal(
const PeerConnectionInterface::RTCOfferAnswerOptions& options);
// Calls CreateOfferAndSetAsLocal with default options.
std::unique_ptr<SessionDescriptionInterface> CreateOfferAndSetAsLocal();
// Calls the underlying PeerConnection's CreateAnswer method and returns the
// resulting SessionDescription once it is available. If the method call
// failed, null is returned.
std::unique_ptr<SessionDescriptionInterface> CreateAnswer(
const PeerConnectionInterface::RTCOfferAnswerOptions& options,
std::string* error_out = nullptr);
// Calls CreateAnswer with the default options.
std::unique_ptr<SessionDescriptionInterface> CreateAnswer();
// Calls CreateAnswer and sets a copy of the offer as the local description.
std::unique_ptr<SessionDescriptionInterface> CreateAnswerAndSetAsLocal(
const PeerConnectionInterface::RTCOfferAnswerOptions& options);
// Calls CreateAnswerAndSetAsLocal with default options.
std::unique_ptr<SessionDescriptionInterface> CreateAnswerAndSetAsLocal();
std::unique_ptr<SessionDescriptionInterface> CreateRollback();
// Calls the underlying PeerConnection's SetLocalDescription method with the
// given session description and waits for the success/failure response.
// Returns true if the description was successfully set.
bool SetLocalDescription(std::unique_ptr<SessionDescriptionInterface> desc,
std::string* error_out = nullptr);
bool SetLocalDescription(std::unique_ptr<SessionDescriptionInterface> desc,
RTCError* error_out);
// Calls the underlying PeerConnection's SetRemoteDescription method with the
// given session description and waits for the success/failure response.
// Returns true if the description was successfully set.
bool SetRemoteDescription(std::unique_ptr<SessionDescriptionInterface> desc,
std::string* error_out = nullptr);
bool SetRemoteDescription(std::unique_ptr<SessionDescriptionInterface> desc,
RTCError* error_out);
// Does a round of offer/answer with the local PeerConnectionWrapper
// generating the offer and the given PeerConnectionWrapper generating the
// answer.
// Equivalent to:
// 1. this->CreateOffer(offer_options)
// 2. this->SetLocalDescription(offer)
// 3. answerer->SetRemoteDescription(offer)
// 4. answerer->CreateAnswer(answer_options)
// 5. answerer->SetLocalDescription(answer)
// 6. this->SetRemoteDescription(answer)
// Returns true if all steps succeed, false otherwise.
// Suggested usage:
// ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
bool ExchangeOfferAnswerWith(PeerConnectionWrapper* answerer);
bool ExchangeOfferAnswerWith(
PeerConnectionWrapper* answerer,
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_options,
const PeerConnectionInterface::RTCOfferAnswerOptions& answer_options);
// The following are wrappers for the underlying PeerConnection's
// AddTransceiver method. They return the result of calling AddTransceiver
// with the given arguments, DCHECKing if there is an error.
rtc::scoped_refptr<RtpTransceiverInterface> AddTransceiver(
cricket::MediaType media_type);
rtc::scoped_refptr<RtpTransceiverInterface> AddTransceiver(
cricket::MediaType media_type,
const RtpTransceiverInit& init);
rtc::scoped_refptr<RtpTransceiverInterface> AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track);
rtc::scoped_refptr<RtpTransceiverInterface> AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init);
// Returns a new dummy audio track with the given label.
rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
const std::string& label);
// Returns a new dummy video track with the given label.
rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
const std::string& label);
// Wrapper for the underlying PeerConnection's AddTrack method. DCHECKs if
// AddTrack fails.
rtc::scoped_refptr<RtpSenderInterface> AddTrack(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids = {});
rtc::scoped_refptr<RtpSenderInterface> AddTrack(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids,
const std::vector<RtpEncodingParameters>& init_send_encodings);
// Calls the underlying PeerConnection's AddTrack method with an audio media
// stream track not bound to any source.
rtc::scoped_refptr<RtpSenderInterface> AddAudioTrack(
const std::string& track_label,
const std::vector<std::string>& stream_ids = {});
// Calls the underlying PeerConnection's AddTrack method with a video media
// stream track fed by a FakeVideoTrackSource.
rtc::scoped_refptr<RtpSenderInterface> AddVideoTrack(
const std::string& track_label,
const std::vector<std::string>& stream_ids = {});
// Calls the underlying PeerConnection's CreateDataChannel method with default
// initialization parameters.
rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
const std::string& label,
const std::optional<DataChannelInit>& config = std::nullopt);
// Returns the signaling state of the underlying PeerConnection.
PeerConnectionInterface::SignalingState signaling_state();
// Returns true if ICE has finished gathering candidates.
bool IsIceGatheringDone();
// Returns true if ICE has established a connection.
bool IsIceConnected();
// Calls GetStats() on the underlying PeerConnection and returns the resulting
// report. If GetStats() fails, this method returns null and fails the test.
rtc::scoped_refptr<const RTCStatsReport> GetStats();
private:
std::unique_ptr<SessionDescriptionInterface> CreateSdp(
rtc::FunctionView<void(CreateSessionDescriptionObserver*)> fn,
std::string* error_out);
bool SetSdp(rtc::FunctionView<void(SetSessionDescriptionObserver*)> fn,
std::string* error_out);
rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
std::unique_ptr<MockPeerConnectionObserver> observer_;
rtc::scoped_refptr<PeerConnectionInterface> pc_;
};
} // namespace webrtc
#endif // PC_PEER_CONNECTION_WRAPPER_H_