mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00

Bug: webrtc:15874 Change-Id: I5bdb19d5e710838b41e6ca283d406c9f1f21286b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348060 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42137}
63 lines
2.2 KiB
C++
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/audio/audio_device.h"
|
|
#include "api/task_queue/task_queue_base.h"
|
|
#include "api/test/simulated_network.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_
|