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/*
* 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/test/simulated_network.h"
#include "test/call_test.h"
namespace webrtc {
namespace test {
class AudioEndToEndTest : public test::EndToEndTest {
public:
AudioEndToEndTest();
protected:
TestAudioDeviceModule* send_audio_device() { return send_audio_device_; }
const AudioSendStream* send_stream() const { return send_stream_; }
const AudioReceiveStream* receive_stream() const { return receive_stream_; }
virtual DefaultNetworkSimulationConfig GetNetworkPipeConfig() const;
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(
TestAudioDeviceModule* send_audio_device,
TestAudioDeviceModule* recv_audio_device) override;
test::PacketTransport* CreateSendTransport(
SingleThreadedTaskQueueForTesting* task_queue,
Call* sender_call) override;
test::PacketTransport* CreateReceiveTransport(
SingleThreadedTaskQueueForTesting* task_queue) override;
void ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStream::Config>* receive_configs) override;
void OnAudioStreamsCreated(
AudioSendStream* send_stream,
const std::vector<AudioReceiveStream*>& receive_streams) override;
void PerformTest() override;
private:
TestAudioDeviceModule* send_audio_device_ = nullptr;
AudioSendStream* send_stream_ = nullptr;
AudioReceiveStream* receive_stream_ = nullptr;
};
} // namespace test
} // namespace webrtc
#endif // AUDIO_TEST_AUDIO_END_TO_END_TEST_H_