This is a reland of ad5c4accad00e04de08e2b62d366cc1f8e0320a5 It was flaky due to starting ICE signaling before SDP negotiation finished. This was solved by adding an helper for adding ice candidates which will wait until the peer connection is ready if needed. Original change's description: > Adds PeerConnection scenario test framework. > > Bug: webrtc:10839 > Change-Id: If67eeb680d016d66c69d8e761a88c240e4931a5d > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147276 > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28754} Bug: webrtc:10839 Change-Id: I6eb8f482561c87e7b0f20d2431d21a41b26c91d5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147877 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28777}
105 lines
4.7 KiB
C++
105 lines
4.7 KiB
C++
/*
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* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef TEST_PEER_SCENARIO_PEER_SCENARIO_H_
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#define TEST_PEER_SCENARIO_PEER_SCENARIO_H_
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// The peer connection scenario test framework enables writing end to end unit
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// tests on the peer connection level. It's similar to the Scenario test but
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// uses the full stack, including SDP and ICE negotiation. This ensures that
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// features work end to end. It's also diffferent from the other tests on peer
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// connection level in that it does not rely on any mocks or fakes other than
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// for media input and networking. Additionally it provides direct access to the
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// underlying peer connection class.
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#include <list>
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#include <vector>
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#include "test/network/network_emulation_manager.h"
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#include "test/peer_scenario/peer_scenario_client.h"
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#include "test/peer_scenario/signaling_route.h"
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#include "test/scenario/stats_collection.h"
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#include "test/scenario/video_frame_matcher.h"
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namespace webrtc {
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namespace test {
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// The PeerScenario class represents a PeerConnection simulation scenario. The
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// main purpose is to maintain ownership and ensure safe destruction order of
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// clients and network emulation. Additionally it reduces the amount of bolier
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// plate requited for some actions. For example usage see the existing tests
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// using this class. Note that it should be used from a single calling thread.
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// This thread will also be assigned as the signaling thread for all peer
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// connections that are created. This means that the process methods must be
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// used when waiting to ensure that messages are processed on the signaling
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// thread.
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class PeerScenario {
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public:
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PeerScenario();
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NetworkEmulationManagerImpl* net() { return &net_; }
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rtc::Thread* thread() { return signaling_thread_; }
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// Creates a client wrapping a peer connection conforming to the given config.
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// The client will share the signaling thread with the scenario. To maintain
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// control of destruction order, ownership is kept within the scenario.
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PeerScenarioClient* CreateClient(PeerScenarioClient::Config config);
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// Sets up a signaling route that can be used for SDP and ICE.
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SignalingRoute ConnectSignaling(PeerScenarioClient* caller,
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PeerScenarioClient* callee,
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std::vector<EmulatedNetworkNode*> send_link,
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std::vector<EmulatedNetworkNode*> ret_link);
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// Connects two clients over given links. This will also start ICE signaling
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// and SDP negotiation with default behavior. For customized behavior,
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// ConnectSignaling should be used to allow more detailed control, for
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// instance to allow different signaling and media routes.
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void SimpleConnection(PeerScenarioClient* caller,
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PeerScenarioClient* callee,
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std::vector<EmulatedNetworkNode*> send_link,
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std::vector<EmulatedNetworkNode*> ret_link);
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// Starts feeding the results of comparing captured frames from |send_track|
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// with decoded frames on |receiver| to |analyzer|.
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// TODO(srte): Provide a way to detach to allow removal of tracks.
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void AttachVideoQualityAnalyzer(VideoQualityAnalyzer* analyzer,
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VideoTrackInterface* send_track,
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PeerScenarioClient* receiver);
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// Waits on |event| while processing messages on the signaling thread.
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bool WaitAndProcess(rtc::Event* event,
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TimeDelta max_duration = TimeDelta::seconds(5));
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// Process messages on the signaling thread for the given duration.
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void ProcessMessages(TimeDelta duration);
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private:
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// Helper struct to maintain ownership of the matcher and taps.
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struct PeerVideoQualityPair {
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public:
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PeerVideoQualityPair(Clock* capture_clock, VideoQualityAnalyzer* analyzer)
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: matcher_({analyzer->Handler()}),
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capture_tap_(capture_clock, &matcher_),
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decode_tap_(capture_clock, &matcher_, 0) {}
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VideoFrameMatcher matcher_;
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CapturedFrameTap capture_tap_;
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DecodedFrameTap decode_tap_;
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};
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Clock* clock() { return Clock::GetRealTimeClock(); }
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rtc::Thread* const signaling_thread_;
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std::list<PeerVideoQualityPair> video_quality_pairs_;
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NetworkEmulationManagerImpl net_;
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std::list<PeerScenarioClient> peer_clients_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // TEST_PEER_SCENARIO_PEER_SCENARIO_H_
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