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/*
* 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