
Add support of negotiating simulcast offer/answer. Also fix some minor issues around to make it finally work. Bug: webrtc:10138 Change-Id: I382f5df04ca6ac04d8ed1e030e7b2ae5706dd10c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137425 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Amit Hilbuch <amithi@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28274}
85 lines
3.2 KiB
C++
85 lines
3.2 KiB
C++
/*
|
|
* Copyright (c) 2019 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 TEST_PC_E2E_TEST_PEER_H_
|
|
#define TEST_PC_E2E_TEST_PEER_H_
|
|
|
|
#include <memory>
|
|
#include <string>
|
|
#include <vector>
|
|
|
|
#include "absl/memory/memory.h"
|
|
#include "api/test/peerconnection_quality_test_fixture.h"
|
|
#include "media/base/media_engine.h"
|
|
#include "modules/audio_device/include/test_audio_device.h"
|
|
#include "pc/peer_connection_wrapper.h"
|
|
#include "pc/test/mock_peer_connection_observers.h"
|
|
#include "rtc_base/network.h"
|
|
#include "rtc_base/task_queue.h"
|
|
#include "rtc_base/thread.h"
|
|
#include "test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.h"
|
|
#include "test/pc/e2e/peer_connection_quality_test_params.h"
|
|
|
|
namespace webrtc {
|
|
namespace webrtc_pc_e2e {
|
|
|
|
// Describes a single participant in the call.
|
|
class TestPeer final : public PeerConnectionWrapper {
|
|
public:
|
|
using PeerConnectionWrapper::PeerConnectionWrapper;
|
|
using VideoConfig = PeerConnectionE2EQualityTestFixture::VideoConfig;
|
|
using AudioConfig = PeerConnectionE2EQualityTestFixture::AudioConfig;
|
|
|
|
// Setups all components, that should be provided to WebRTC
|
|
// PeerConnectionFactory and PeerConnection creation methods,
|
|
// also will setup dependencies, that are required for media analyzers
|
|
// injection.
|
|
//
|
|
// |signaling_thread| will be provided by test fixture implementation.
|
|
// |params| - describes current peer paramters, like current peer video
|
|
// streams and audio streams
|
|
// |audio_outpu_file_name| - the name of output file, where incoming audio
|
|
// stream should be written. It should be provided from remote peer
|
|
// |params.audio_config.output_file_name|
|
|
static std::unique_ptr<TestPeer> CreateTestPeer(
|
|
std::unique_ptr<InjectableComponents> components,
|
|
std::unique_ptr<Params> params,
|
|
std::unique_ptr<MockPeerConnectionObserver> observer,
|
|
VideoQualityAnalyzerInjectionHelper* video_analyzer_helper,
|
|
rtc::Thread* signaling_thread,
|
|
absl::optional<std::string> audio_output_file_name,
|
|
double bitrate_multiplier,
|
|
rtc::TaskQueue* task_queue);
|
|
|
|
Params* params() const { return params_.get(); }
|
|
void DetachAecDump() { audio_processing_->DetachAecDump(); }
|
|
|
|
// Adds provided |candidates| to the owned peer connection.
|
|
bool AddIceCandidates(
|
|
std::vector<std::unique_ptr<IceCandidateInterface>> candidates);
|
|
|
|
private:
|
|
TestPeer(rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory,
|
|
rtc::scoped_refptr<PeerConnectionInterface> pc,
|
|
std::unique_ptr<MockPeerConnectionObserver> observer,
|
|
std::unique_ptr<Params> params,
|
|
rtc::scoped_refptr<AudioProcessing> audio_processing);
|
|
|
|
std::unique_ptr<Params> params_;
|
|
rtc::scoped_refptr<AudioProcessing> audio_processing_;
|
|
|
|
std::vector<std::unique_ptr<IceCandidateInterface>> remote_ice_candidates_;
|
|
};
|
|
|
|
} // namespace webrtc_pc_e2e
|
|
} // namespace webrtc
|
|
|
|
#endif // TEST_PC_E2E_TEST_PEER_H_
|