webrtc_m130/test/pc/e2e/test_peer.h
Artem Titov 70f80e5962 Add support for creation of AEC dump during the test with PC framework.
Also add conversational speech into PC smoke test (with resource files).

Bug: webrtc:10138
Change-Id: I415a5565bc9146821476ffc60f57f47ed51f89c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132323
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27592}
2019-04-12 13:09:12 +00:00

83 lines
3.1 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 "absl/memory/memory.h"
#include "api/array_view.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(
rtc::ArrayView<const IceCandidateInterface* const> 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_;
};
} // namespace webrtc_pc_e2e
} // namespace webrtc
#endif // TEST_PC_E2E_TEST_PEER_H_