webrtc/audio/test/audio_end_to_end_test.h
Artem Titov fb8e3de0a8 Use AudioDeviceModule instead of TestAudioDeviceModule.
This is step to allow migration of Test ADM to the AudioDeviceModuleImpl
as a base class to include AudioDeviceBuffer into SUT.

Also it will allow to remove WaitForRecordingEnd() method from Test
ADM

Bug: b/272350185, webrtc:15081
Change-Id: If2aa43ec0c31f6ad9aab8aa3e36cabc4a7a73c22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300862
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39849}
2023-04-13 12:31:34 +00:00

63 lines
2.2 KiB
C++

/*
* Copyright (c) 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 AUDIO_TEST_AUDIO_END_TO_END_TEST_H_
#define AUDIO_TEST_AUDIO_END_TO_END_TEST_H_
#include <memory>
#include <string>
#include <vector>
#include "api/task_queue/task_queue_base.h"
#include "api/test/simulated_network.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "test/call_test.h"
namespace webrtc {
namespace test {
class AudioEndToEndTest : public test::EndToEndTest {
public:
AudioEndToEndTest();
protected:
AudioDeviceModule* send_audio_device() { return send_audio_device_; }
const AudioSendStream* send_stream() const { return send_stream_; }
const AudioReceiveStreamInterface* receive_stream() const {
return receive_stream_;
}
size_t GetNumVideoStreams() const override;
size_t GetNumAudioStreams() const override;
size_t GetNumFlexfecStreams() const override;
std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer() override;
std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer() override;
void OnFakeAudioDevicesCreated(AudioDeviceModule* send_audio_device,
AudioDeviceModule* recv_audio_device) override;
void ModifyAudioConfigs(AudioSendStream::Config* send_config,
std::vector<AudioReceiveStreamInterface::Config>*
receive_configs) override;
void OnAudioStreamsCreated(AudioSendStream* send_stream,
const std::vector<AudioReceiveStreamInterface*>&
receive_streams) override;
private:
AudioDeviceModule* send_audio_device_ = nullptr;
AudioSendStream* send_stream_ = nullptr;
AudioReceiveStreamInterface* receive_stream_ = nullptr;
};
} // namespace test
} // namespace webrtc
#endif // AUDIO_TEST_AUDIO_END_TO_END_TEST_H_