| Index: webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h
|
| diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h b/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..18f645263dce221228f5d1e35ff47dad1003b4ec
|
| --- /dev/null
|
| +++ b/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h
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| @@ -0,0 +1,45 @@
|
| +/*
|
| + * Copyright (c) 2016 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_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_AUDIO_NETWORK_ADAPTOR_H_
|
| +#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_AUDIO_NETWORK_ADAPTOR_H_
|
| +
|
| +#include "testing/gmock/include/gmock/gmock.h"
|
| +#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
|
| +
|
| +namespace webrtc {
|
| +
|
| +class MockAudioNetworkAdaptor : public AudioNetworkAdaptor {
|
| + public:
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| + virtual ~MockAudioNetworkAdaptor() { Die(); }
|
| + MOCK_METHOD0(Die, void());
|
| +
|
| + MOCK_METHOD1(SetUplinkBandwidth, void(int uplink_bandwidth_bps));
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| +
|
| + MOCK_METHOD1(SetUplinkPacketLossFraction,
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| + void(float uplink_packet_loss_fraction));
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| +
|
| + MOCK_METHOD1(SetRtt, void(int rtt_ms));
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| +
|
| + MOCK_METHOD1(SetTargetAudioBitrate, void(int target_audio_bitrate_bps));
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| +
|
| + MOCK_METHOD2(SetReceiverFrameLengthRange,
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| + void(int min_frame_length_ms, int max_frame_length_ms));
|
| +
|
| + MOCK_METHOD0(GetEncoderRuntimeConfig, EncoderRuntimeConfig());
|
| +
|
| + MOCK_METHOD1(StartDebugDump, void(FILE* file_handle));
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| +
|
| + MOCK_METHOD0(StopDebugDump, void());
|
| +};
|
| +
|
| +} // namespace webrtc
|
| +
|
| +#endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_AUDIO_NETWORK_ADAPTOR_H_
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|
|