webrtc/modules/audio_processing/aec_dump/mock_aec_dump.h
Alex Loiko e36e8bbf6d Add FixedGainController and move GainController2 in APM.
The FixedGainController (FGC) applies a fixed gain. It will also
control the limiter. The limiter will be landed over the next several
CLs.

The GainController2 is a 'private submodule' of APM. It will control
the new automatic gain controller (AGC). It controls the AGC through
Initialize() and ApplyConfig().

This CL contains

* build changes to make modules/audio_processing/agc2 an independent
  target

* a new MutableFloatAudioFrame which is the audio interface between
  AGC2 and APM

* move of the fixed gain application from GainController2 to
  FixedGainController.

If you are a googler, there is more information in this doc:
https://docs.google.com/document/d/1RV2Doet3MZtUPAHVva61Vjo20iyd1bmmm3aR8znWpzo/edit#

Bug: webrtc:7949
Change-Id: Ief95cbbce83c3aafe54638fd2ab881c9fb8bdc3a
Reviewed-on: https://webrtc-review.googlesource.com/50440
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22046}
2018-02-16 10:56:38 +00:00

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1.7 KiB
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/*
* 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 MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_
#define MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_
#include <memory>
#include "modules/audio_processing/include/aec_dump.h"
#include "modules/include/module_common_types.h"
#include "test/gmock.h"
namespace webrtc {
namespace test {
class MockAecDump : public AecDump {
public:
MockAecDump();
virtual ~MockAecDump();
MOCK_METHOD1(WriteInitMessage,
void(const InternalAPMStreamsConfig& streams_config));
MOCK_METHOD1(AddCaptureStreamInput,
void(const AudioFrameView<const float>& src));
MOCK_METHOD1(AddCaptureStreamOutput,
void(const AudioFrameView<const float>& src));
MOCK_METHOD1(AddCaptureStreamInput, void(const AudioFrame& frame));
MOCK_METHOD1(AddCaptureStreamOutput, void(const AudioFrame& frame));
MOCK_METHOD1(AddAudioProcessingState,
void(const AudioProcessingState& state));
MOCK_METHOD0(WriteCaptureStreamMessage, void());
MOCK_METHOD1(WriteRenderStreamMessage, void(const AudioFrame& frame));
MOCK_METHOD1(WriteRenderStreamMessage,
void(const AudioFrameView<const float>& src));
MOCK_METHOD1(WriteConfig, void(const InternalAPMConfig& config));
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_