153f11e1b4
This CL adds the Level Estimator of the new gain controller. The Level Estimator divides a 10ms input frame in kSubFramesInFrame=20 sub frames. We take the maximal sample values in every sub frame. We then apply attack/decay smoothing. This is the final level estimate. The results will be used with InterpolatedGainCurve (see this CL https://webrtc-review.googlesource.com/c/src/+/51920). For every level estimate value, we look up a gain with InterpolatedGainCurve::LookUpGainToApply. This gain is then applied to the signal. Bug: webrtc:7949 Change-Id: I2b4b3894a3e945d3dd916ce516c79abacb2b18b1 Reviewed-on: https://webrtc-review.googlesource.com/52381 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22054}
39 lines
1.2 KiB
C++
39 lines
1.2 KiB
C++
/*
|
|
* Copyright (c) 2018 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_AGC2_COMMON_H_
|
|
#define MODULES_AUDIO_PROCESSING_AGC2_AGC2_COMMON_H_
|
|
|
|
#include "rtc_base/basictypes.h"
|
|
|
|
namespace webrtc {
|
|
|
|
constexpr float kMinSampleValue = -32768.f;
|
|
constexpr float kMaxSampleValue = 32767.f;
|
|
|
|
constexpr size_t kFrameDurationMs = 10;
|
|
constexpr size_t kSubFramesInFrame = 20;
|
|
|
|
constexpr float kAttackFilterConstant = 0.f;
|
|
|
|
constexpr size_t kMaximalNumberOfSamplesPerChannel = 480;
|
|
|
|
// This is computed from kDecayMs by
|
|
// 10 ** (-1/20 * subframe_duration / kDecayMs).
|
|
// |subframe_duration| is |kFrameDurationMs / kSubFramesInFrame|.
|
|
// kDecayMs is defined in agc2_testing_common.h
|
|
constexpr float kDecayFilterConstant = 0.9998848773724686f;
|
|
|
|
// TODO(aleloi): add the other constants as more AGC2 components are
|
|
// added.
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_AGC2_AGC2_COMMON_H_
|