/*
 *  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_