text: /*
   *  Copyright (c) 2018 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.
   */
  
  #include "modules/audio_processing/aec3/mock/mock_render_delay_buffer.h"
  
  namespace webrtc {
  namespace test {
  
  MockRenderDelayBuffer::MockRenderDelayBuffer(int sample_rate_hz,
                                               size_t num_channels)
      : block_buffer_(GetRenderDelayBufferSize(4, 4, 12),
                      NumBandsForRate(sample_rate_hz),
                      num_channels),
        spectrum_buffer_(block_buffer_.buffer.size(), num_channels),
        fft_buffer_(block_buffer_.buffer.size(), num_channels),
        render_buffer_(&block_buffer_, &spectrum_buffer_, &fft_buffer_),
        downsampled_render_buffer_(GetDownSampledBufferSize(4, 4)) {
    ON_CALL(*this, GetRenderBuffer())
        .WillByDefault(
            ::testing::Invoke(this, &MockRenderDelayBuffer::FakeGetRenderBuffer));
    ON_CALL(*this, GetDownsampledRenderBuffer())
        .WillByDefault(::testing::Invoke(
            this, &MockRenderDelayBuffer::FakeGetDownsampledRenderBuffer));
  }
  
  MockRenderDelayBuffer::~MockRenderDelayBuffer() = default;
  
  }  // namespace test
  }  // namespace webrtc
  