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/*
* Copyright (c) 2012 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_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_
#include "webrtc/modules/audio_coding/neteq/dtmf_buffer.h"
#include "gmock/gmock.h"
namespace webrtc {
class MockDtmfBuffer : public DtmfBuffer {
public:
MockDtmfBuffer(int fs) : DtmfBuffer(fs) {}
virtual ~MockDtmfBuffer() { Die(); }
MOCK_METHOD0(Die, void());
MOCK_METHOD0(Flush,
void());
MOCK_METHOD1(InsertEvent,
int(const DtmfEvent& event));
MOCK_METHOD2(GetEvent,
bool(uint32_t current_timestamp, DtmfEvent* event));
MOCK_CONST_METHOD0(Length,
size_t());
MOCK_CONST_METHOD0(Empty,
bool());
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_
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