| /* | 
 |  *  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_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_ | 
 | #define WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_ | 
 |  | 
 | #include "webrtc/modules/audio_device/include/audio_device_defines.h" | 
 | #include "webrtc/test/gmock.h" | 
 |  | 
 | namespace webrtc { | 
 | namespace test { | 
 |  | 
 | class MockAudioTransport : public AudioTransport { | 
 |  public: | 
 |   MockAudioTransport() {} | 
 |   ~MockAudioTransport() {} | 
 |  | 
 |   MOCK_METHOD10(RecordedDataIsAvailable, | 
 |                 int32_t(const void* audioSamples, | 
 |                         const size_t nSamples, | 
 |                         const size_t nBytesPerSample, | 
 |                         const size_t nChannels, | 
 |                         const uint32_t samplesPerSec, | 
 |                         const uint32_t totalDelayMS, | 
 |                         const int32_t clockDrift, | 
 |                         const uint32_t currentMicLevel, | 
 |                         const bool keyPressed, | 
 |                         uint32_t& newMicLevel)); | 
 |  | 
 |   MOCK_METHOD8(NeedMorePlayData, | 
 |                int32_t(const size_t nSamples, | 
 |                        const size_t nBytesPerSample, | 
 |                        const size_t nChannels, | 
 |                        const uint32_t samplesPerSec, | 
 |                        void* audioSamples, | 
 |                        size_t& nSamplesOut, | 
 |                        int64_t* elapsed_time_ms, | 
 |                        int64_t* ntp_time_ms)); | 
 |  | 
 |   MOCK_METHOD6(PushCaptureData, | 
 |                void(int voe_channel, | 
 |                     const void* audio_data, | 
 |                     int bits_per_sample, | 
 |                     int sample_rate, | 
 |                     size_t number_of_channels, | 
 |                     size_t number_of_frames)); | 
 |  | 
 |   MOCK_METHOD7(PullRenderData, | 
 |                void(int bits_per_sample, | 
 |                     int sample_rate, | 
 |                     size_t number_of_channels, | 
 |                     size_t number_of_frames, | 
 |                     void* audio_data, | 
 |                     int64_t* elapsed_time_ms, | 
 |                     int64_t* ntp_time_ms)); | 
 | }; | 
 |  | 
 | }  // namespace test | 
 | }  // namespace webrtc | 
 |  | 
 | #endif  // WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_ |