| Index: webrtc/audio/test/low_bandwidth_audio_test.h
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| diff --git a/webrtc/audio/test/low_bandwidth_audio_test.h b/webrtc/audio/test/low_bandwidth_audio_test.h
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| new file mode 100644
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| index 0000000000000000000000000000000000000000..eae650af22ca167fde67aff7713f3391d1e554cf
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| --- /dev/null
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| +++ b/webrtc/audio/test/low_bandwidth_audio_test.h
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| @@ -0,0 +1,61 @@
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| +/*
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| + *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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| + *
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| + *  Use of this source code is governed by a BSD-style license
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| + *  that can be found in the LICENSE file in the root of the source
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| + *  tree. An additional intellectual property rights grant can be found
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| + *  in the file PATENTS.  All contributing project authors may
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| + *  be found in the AUTHORS file in the root of the source tree.
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| + */
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| +#ifndef WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_
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| +#define WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_
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| +
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| +#include <memory>
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| +#include <string>
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| +#include <vector>
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| +
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| +#include "webrtc/test/call_test.h"
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| +#include "webrtc/test/fake_audio_device.h"
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| +
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| +namespace webrtc {
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| +namespace test {
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| +
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| +class AudioQualityTest : public test::EndToEndTest {
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| + public:
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| +  AudioQualityTest();
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| +
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| + protected:
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| +  virtual std::string AudioInputFile();
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| +  virtual std::string AudioOutputFile();
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| +
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| +  virtual FakeNetworkPipe::Config GetNetworkPipeConfig();
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| +
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| +  size_t GetNumVideoStreams() const override;
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| +  size_t GetNumAudioStreams() const override;
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| +  size_t GetNumFlexfecStreams() const override;
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| +
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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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| +
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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;
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| +
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| +  test::PacketTransport* CreateSendTransport(Call* sender_call) override;
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| +  test::PacketTransport* CreateReceiveTransport() override;
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| +
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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;
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| +
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| +  void PerformTest() override;
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| +  void OnTestFinished() override;
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| +
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| + private:
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| +  test::FakeAudioDevice* send_audio_device_;
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| +};
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| +
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| +}  // namespace test
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| +}  // namespace webrtc
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| +
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| +#endif  // WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_
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| 
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