webrtc/modules/audio_processing/aec3/mock/mock_block_processor.h
Per Åhgren d0fa820559 Allow AEC3 to use any externally reported audio buffer delay in AEC3
This CL adds support for using any externally reported audio buffer
delay to set the initial alignment in AEC3 which is used before the
AEC has been able to detect the delay.

Bug: chromium:834182,webrtc:9163
Change-Id: Ic71355f69b7c4d5815b78e49987043441e7908fb
Reviewed-on: https://webrtc-review.googlesource.com/70580
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22917}
2018-04-18 09:05:54 +00:00

40 lines
1.4 KiB
C++

/*
* 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 MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_BLOCK_PROCESSOR_H_
#define MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_BLOCK_PROCESSOR_H_
#include <vector>
#include "modules/audio_processing/aec3/block_processor.h"
#include "test/gmock.h"
namespace webrtc {
namespace test {
class MockBlockProcessor : public BlockProcessor {
public:
virtual ~MockBlockProcessor() {}
MOCK_METHOD3(ProcessCapture,
void(bool level_change,
bool saturated_microphone_signal,
std::vector<std::vector<float>>* capture_block));
MOCK_METHOD1(BufferRender,
void(const std::vector<std::vector<float>>& block));
MOCK_METHOD1(UpdateEchoLeakageStatus, void(bool leakage_detected));
MOCK_CONST_METHOD1(GetMetrics, void(EchoControl::Metrics* metrics));
MOCK_METHOD1(SetAudioBufferDelay, void(size_t delay_ms));
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_BLOCK_PROCESSOR_H_