/*
 *  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_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_

#include <vector>

#include "webrtc/modules/audio_processing/aec3/aec3_common.h"
#include "webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h"
#include "webrtc/modules/audio_processing/aec3/render_buffer.h"
#include "webrtc/modules/audio_processing/aec3/render_delay_buffer.h"
#include "webrtc/test/gmock.h"

namespace webrtc {
namespace test {

class MockRenderDelayBuffer : public RenderDelayBuffer {
 public:
  explicit MockRenderDelayBuffer(int sample_rate_hz)
      : render_buffer_(Aec3Optimization::kNone,
                       NumBandsForRate(sample_rate_hz),
                       kRenderDelayBufferSize,
                       std::vector<size_t>(1, kAdaptiveFilterLength)) {
    ON_CALL(*this, GetRenderBuffer())
        .WillByDefault(
            testing::Invoke(this, &MockRenderDelayBuffer::FakeGetRenderBuffer));
    ON_CALL(*this, GetDownsampledRenderBuffer())
        .WillByDefault(testing::Invoke(
            this, &MockRenderDelayBuffer::FakeGetDownsampledRenderBuffer));
  }
  virtual ~MockRenderDelayBuffer() = default;

  MOCK_METHOD0(Reset, void());
  MOCK_METHOD1(Insert, bool(const std::vector<std::vector<float>>& block));
  MOCK_METHOD0(UpdateBuffers, bool());
  MOCK_METHOD1(SetDelay, void(size_t delay));
  MOCK_CONST_METHOD0(Delay, size_t());
  MOCK_CONST_METHOD0(MaxDelay, size_t());
  MOCK_CONST_METHOD0(IsBlockAvailable, bool());
  MOCK_CONST_METHOD0(GetRenderBuffer, const RenderBuffer&());
  MOCK_CONST_METHOD0(GetDownsampledRenderBuffer,
                     const DownsampledRenderBuffer&());

 private:
  const RenderBuffer& FakeGetRenderBuffer() const { return render_buffer_; }
  const DownsampledRenderBuffer& FakeGetDownsampledRenderBuffer() const {
    return downsampled_render_buffer_;
  }
  RenderBuffer render_buffer_;
  DownsampledRenderBuffer downsampled_render_buffer_;
};

}  // namespace test
}  // namespace webrtc

#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_