mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-18 08:07:56 +01:00
bb547203bf
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
64 lines
2.5 KiB
C++
64 lines
2.5 KiB
C++
/*
|
|
* 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_
|