83830f316e
Instead use rtc::AutoThread in tests that need that. Bug: webrtc:9714 Change-Id: I1f33b1b2d321770d062504dd9ef86d66a345dd42 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254681 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36950}
118 lines
4.3 KiB
C++
118 lines
4.3 KiB
C++
/*
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* Copyright 2017 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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#include <memory>
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#include <utility>
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#include <vector>
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#include "api/audio/audio_mixer.h"
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#include "api/audio_codecs/builtin_audio_decoder_factory.h"
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#include "api/audio_codecs/builtin_audio_encoder_factory.h"
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#include "api/create_peerconnection_factory.h"
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#include "api/media_types.h"
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#include "api/peer_connection_interface.h"
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#include "api/rtp_transceiver_interface.h"
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#include "api/scoped_refptr.h"
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#include "api/video_codecs/builtin_video_decoder_factory.h"
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#include "api/video_codecs/builtin_video_encoder_factory.h"
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#include "modules/audio_device/include/audio_device.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "p2p/base/port_allocator.h"
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#include "pc/peer_connection_wrapper.h"
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#include "pc/test/fake_audio_capture_module.h"
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#include "pc/test/mock_peer_connection_observers.h"
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#include "rtc_base/rtc_certificate_generator.h"
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#include "rtc_base/thread.h"
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#include "system_wrappers/include/metrics.h"
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#include "test/gtest.h"
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// This file contains unit tests that relate to the behavior of the
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// SdpOfferAnswer module.
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// Tests are writen as integration tests with PeerConnection, since the
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// behaviors are still linked so closely that it is hard to test them in
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// isolation.
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namespace webrtc {
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using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
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namespace {
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std::unique_ptr<rtc::Thread> CreateAndStartThread() {
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auto thread = rtc::Thread::Create();
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thread->Start();
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return thread;
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}
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} // namespace
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class SdpOfferAnswerTest : public ::testing::Test {
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public:
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SdpOfferAnswerTest()
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// Note: We use a PeerConnectionFactory with a distinct
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// signaling thread, so that thread handling can be tested.
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: signaling_thread_(CreateAndStartThread()),
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pc_factory_(
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CreatePeerConnectionFactory(nullptr,
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nullptr,
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signaling_thread_.get(),
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FakeAudioCaptureModule::Create(),
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CreateBuiltinAudioEncoderFactory(),
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CreateBuiltinAudioDecoderFactory(),
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CreateBuiltinVideoEncoderFactory(),
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CreateBuiltinVideoDecoderFactory(),
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nullptr /* audio_mixer */,
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nullptr /* audio_processing */)) {
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webrtc::metrics::Reset();
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}
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std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection() {
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RTCConfiguration config;
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config.sdp_semantics = SdpSemantics::kUnifiedPlan;
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return CreatePeerConnection(config);
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}
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std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection(
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const RTCConfiguration& config) {
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auto observer = std::make_unique<MockPeerConnectionObserver>();
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auto result = pc_factory_->CreatePeerConnectionOrError(
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config, PeerConnectionDependencies(observer.get()));
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EXPECT_TRUE(result.ok());
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observer->SetPeerConnectionInterface(result.value().get());
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return std::make_unique<PeerConnectionWrapper>(
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pc_factory_, result.MoveValue(), std::move(observer));
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}
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protected:
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std::unique_ptr<rtc::Thread> signaling_thread_;
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rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
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private:
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rtc::AutoThread main_thread_;
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};
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TEST_F(SdpOfferAnswerTest, OnTrackReturnsProxiedObject) {
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auto caller = CreatePeerConnection();
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auto callee = CreatePeerConnection();
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auto audio_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
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ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
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// Verify that caller->observer->OnTrack() has been called with a
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// proxied transceiver object.
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ASSERT_EQ(callee->observer()->on_track_transceivers_.size(), 1u);
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auto transceiver = callee->observer()->on_track_transceivers_[0];
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// Since the signaling thread is not the current thread,
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// this will DCHECK if the transceiver is not proxied.
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transceiver->stopped();
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}
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} // namespace webrtc
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