| /* |
| * 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_BUFFER_LEVEL_FILTER_H_ |
| #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_ |
| |
| #include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h" |
| |
| #include "testing/gmock/include/gmock/gmock.h" |
| |
| namespace webrtc { |
| |
| class MockBufferLevelFilter : public BufferLevelFilter { |
| public: |
| virtual ~MockBufferLevelFilter() { Die(); } |
| MOCK_METHOD0(Die, |
| void()); |
| MOCK_METHOD0(Reset, |
| void()); |
| MOCK_METHOD3(Update, |
| void(size_t buffer_size_packets, int time_stretched_samples, |
| size_t packet_len_samples)); |
| MOCK_METHOD1(SetTargetBufferLevel, |
| void(int target_buffer_level)); |
| MOCK_CONST_METHOD0(filtered_current_level, |
| int()); |
| }; |
| |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_ |