Mirko Bonadei 92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00

57 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 MODULES_AUDIO_PROCESSING_AGC2_GAIN_CONTROLLER2_H_
#define MODULES_AUDIO_PROCESSING_AGC2_GAIN_CONTROLLER2_H_
#include <memory>
#include <string>
#include "modules/audio_processing/agc2/digital_gain_applier.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "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 // MODULES_AUDIO_PROCESSING_AGC2_GAIN_CONTROLLER2_H_