| /* |
| * 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_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MOCK_WAVREADER_H_ |
| #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MOCK_WAVREADER_H_ |
| |
| #include <cstddef> |
| #include <string> |
| |
| #include "webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h" |
| #include "webrtc/test/gmock.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| namespace test { |
| namespace conversational_speech { |
| |
| class MockWavReader : public WavReaderInterface { |
| public: |
| MockWavReader( |
| int sample_rate, size_t num_channels, size_t num_samples) |
| : sample_rate_(sample_rate), num_channels_(num_channels), |
| num_samples_(num_samples) {} |
| ~MockWavReader() = default; |
| |
| // TOOD(alessiob): use ON_CALL to return random samples. |
| MOCK_METHOD2(ReadFloatSamples, size_t(size_t, float*)); |
| MOCK_METHOD2(ReadInt16Samples, size_t(size_t, int16_t*)); |
| |
| // TOOD(alessiob): use ON_CALL to return properties. |
| MOCK_CONST_METHOD0(sample_rate, int()); |
| MOCK_CONST_METHOD0(num_channels, size_t()); |
| MOCK_CONST_METHOD0(num_samples, size_t()); |
| |
| private: |
| const int sample_rate_; |
| const size_t num_channels_; |
| const size_t num_samples_; |
| }; |
| |
| } // namespace conversational_speech |
| } // namespace test |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MOCK_WAVREADER_H_ |