webrtc_m130/webrtc/audio/test/low_bandwidth_audio_test.h
oprypin 92220ffe9f Low-bandwidth audio testing
The C++ part of the test uses CallTest to set up an audio-only call. It reads an audio file, plays it through a FakeAudioDevice which transfers data through a FakeNetworkPipe for another FakeAudioDevice to receive it and write it to a file. Information about these files is printed to stdout.

The test cases are meant to try different network and audio configs (more are planned in the future).

The Python part of the test runs the C++ part and scans stdout for tests to perform, runs the pairs of files (original and degraded) through the PESQ tool to receive a score and writes that to perf dashboard.

BUG=webrtc:7229
NOTRY=True

Review-Url: https://codereview.webrtc.org/2694203002
Cr-Commit-Position: refs/heads/master@{#17356}
2017-03-23 10:40:03 +00:00

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1.8 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 WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_
#define WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_
#include <memory>
#include <string>
#include <vector>
#include "webrtc/test/call_test.h"
#include "webrtc/test/fake_audio_device.h"
namespace webrtc {
namespace test {
class AudioQualityTest : public test::EndToEndTest {
public:
AudioQualityTest();
protected:
virtual std::string AudioInputFile();
virtual std::string AudioOutputFile();
virtual FakeNetworkPipe::Config GetNetworkPipeConfig();
size_t GetNumVideoStreams() const override;
size_t GetNumAudioStreams() const override;
size_t GetNumFlexfecStreams() const override;
std::unique_ptr<test::FakeAudioDevice::Capturer> CreateCapturer() override;
std::unique_ptr<test::FakeAudioDevice::Renderer> CreateRenderer() override;
void OnFakeAudioDevicesCreated(
test::FakeAudioDevice* send_audio_device,
test::FakeAudioDevice* recv_audio_device) override;
test::PacketTransport* CreateSendTransport(Call* sender_call) override;
test::PacketTransport* CreateReceiveTransport() override;
void ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStream::Config>* receive_configs) override;
void PerformTest() override;
void OnTestFinished() override;
private:
test::FakeAudioDevice* send_audio_device_;
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_