webrtc/modules/audio_processing/agc2/gain_controller2.h
Mirko Bonadei bb547203bf Moving src/webrtc into src/.
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}
2017-09-15 04:25:06 +00:00

56 lines
1.7 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_AGC2_GAIN_CONTROLLER2_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC2_GAIN_CONTROLLER2_H_
#include <memory>
#include <string>
#include "webrtc/modules/audio_processing/agc2/digital_gain_applier.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/rtc_base/constructormagic.h"
namespace webrtc {
class ApmDataDumper;
class AudioBuffer;
// Gain Controller 2 aims to automatically adjust levels by acting on the
// microphone gain and/or applying digital gain.
//
// It temporarily implements a hard-coded gain mode only.
class GainController2 {
public:
explicit GainController2(int sample_rate_hz);
~GainController2();
int sample_rate_hz() { return sample_rate_hz_; }
void Process(AudioBuffer* audio);
static bool Validate(const AudioProcessing::Config::GainController2& config);
static std::string ToString(
const AudioProcessing::Config::GainController2& config);
private:
int sample_rate_hz_;
std::unique_ptr<ApmDataDumper> data_dumper_;
DigitalGainApplier digital_gain_applier_;
static int instance_count_;
// TODO(alessiob): Remove once a meaningful gain controller mode is
// implemented.
const float gain_;
RTC_DISALLOW_COPY_AND_ASSIGN(GainController2);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC2_GAIN_CONTROLLER2_H_