| Index: webrtc/audio/test/low_bandwidth_audio_test.h
|
| diff --git a/webrtc/audio/test/low_bandwidth_audio_test.h b/webrtc/audio/test/low_bandwidth_audio_test.h
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..eae650af22ca167fde67aff7713f3391d1e554cf
|
| --- /dev/null
|
| +++ b/webrtc/audio/test/low_bandwidth_audio_test.h
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| @@ -0,0 +1,61 @@
|
| +/*
|
| + * 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();
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| + virtual std::string AudioOutputFile();
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| +
|
| + virtual FakeNetworkPipe::Config GetNetworkPipeConfig();
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| +
|
| + size_t GetNumVideoStreams() const override;
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| + size_t GetNumAudioStreams() const override;
|
| + size_t GetNumFlexfecStreams() const override;
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| +
|
| + std::unique_ptr<test::FakeAudioDevice::Capturer> CreateCapturer() override;
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| + std::unique_ptr<test::FakeAudioDevice::Renderer> CreateRenderer() override;
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| +
|
| + void OnFakeAudioDevicesCreated(
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| + test::FakeAudioDevice* send_audio_device,
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| + test::FakeAudioDevice* recv_audio_device) override;
|
| +
|
| + test::PacketTransport* CreateSendTransport(Call* sender_call) override;
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| + test::PacketTransport* CreateReceiveTransport() override;
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| +
|
| + void ModifyAudioConfigs(
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| + AudioSendStream::Config* send_config,
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| + std::vector<AudioReceiveStream::Config>* receive_configs) override;
|
| +
|
| + void PerformTest() override;
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| + void OnTestFinished() override;
|
| +
|
| + private:
|
| + test::FakeAudioDevice* send_audio_device_;
|
| +};
|
| +
|
| +} // namespace test
|
| +} // namespace webrtc
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| +
|
| +#endif // WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_
|
|
|