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/* //device/include/server/AudioFlinger/AudioPeakingFilter.cpp
**
** Copyright 2007, The Android Open Source Project
**
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
**
** http://www.apache.org/licenses/LICENSE-2.0
**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
** limitations under the License.
*/
#include "AudioPeakingFilter.h"
#include "AudioCommon.h"
#include "EffectsMath.h"
#include <new>
#include <assert.h>
#include <cutils/compiler.h>
namespace android {
// Format of the coefficient table:
// kCoefTable[freq][gain][bw][coef]
// freq - peak frequency, in octaves below Nyquist,from -9 to -1.
// gain - gain, in millibel, starting at -9600, jumps of 1024, to 4736 millibel.
// bw - bandwidth, starting at 1 cent, jumps of 1024, to 3073 cents.
// coef - 0: b0
// 1: b1
// 2: b2
// 3: -a1
// 4: -a2
static const size_t kInDims[3] = {9, 15, 4};
static const audio_coef_t kCoefTable[9*15*4*5] = {
#include "AudioPeakingFilterCoef.inl"
};
AudioCoefInterpolator AudioPeakingFilter::mCoefInterp(3, kInDims, 5, (const audio_coef_t*) kCoefTable);
AudioPeakingFilter::AudioPeakingFilter(int nChannels, int sampleRate)
: mBiquad(nChannels, sampleRate) {
configure(nChannels, sampleRate);
reset();
}
void AudioPeakingFilter::configure(int nChannels, int sampleRate) {
mNiquistFreq = sampleRate * 500;
mFrequencyFactor = ((1ull) << 42) / mNiquistFreq;
mBiquad.configure(nChannels, sampleRate);
setFrequency(mNominalFrequency);
commit(true);
}
void AudioPeakingFilter::reset() {
setGain(0);
setFrequency(0);
setBandwidth(2400);
commit(true);
}
void AudioPeakingFilter::setFrequency(uint32_t millihertz) {
mNominalFrequency = millihertz;
if (CC_UNLIKELY(millihertz > mNiquistFreq / 2)) {
millihertz = mNiquistFreq / 2;
}
uint32_t normFreq = static_cast<uint32_t>(
(static_cast<uint64_t>(millihertz) * mFrequencyFactor) >> 10);
if (CC_LIKELY(normFreq > (1 << 23))) {
mFrequency = (Effects_log2(normFreq) - ((32-9) << 15)) << (FREQ_PRECISION_BITS - 15);
} else {
mFrequency = 0;
}
}
void AudioPeakingFilter::setGain(int32_t millibel) {
mGain = millibel + 9600;
}
void AudioPeakingFilter::setBandwidth(uint32_t cents) {
mBandwidth = cents - 1;
}
void AudioPeakingFilter::commit(bool immediate) {
audio_coef_t coefs[5];
int intCoord[3] = {
mFrequency >> FREQ_PRECISION_BITS,
mGain >> GAIN_PRECISION_BITS,
mBandwidth >> BANDWIDTH_PRECISION_BITS
};
uint32_t fracCoord[3] = {
mFrequency << (32 - FREQ_PRECISION_BITS),
static_cast<uint32_t>(mGain) << (32 - GAIN_PRECISION_BITS),
mBandwidth << (32 - BANDWIDTH_PRECISION_BITS)
};
mCoefInterp.getCoef(intCoord, fracCoord, coefs);
mBiquad.setCoefs(coefs, immediate);
}
void AudioPeakingFilter::getBandRange(uint32_t & low, uint32_t & high) const {
// Half bandwidth, in octaves, 15-bit precision
int32_t halfBW = (((mBandwidth + 1) / 2) << 15) / 1200;
low = static_cast<uint32_t>((static_cast<uint64_t>(mNominalFrequency) * Effects_exp2(-halfBW + (16 << 15))) >> 16);
if (CC_UNLIKELY(halfBW >= (16 << 15))) {
high = mNiquistFreq;
} else {
high = static_cast<uint32_t>((static_cast<uint64_t>(mNominalFrequency) * Effects_exp2(halfBW + (16 << 15))) >> 16);
if (CC_UNLIKELY(high > mNiquistFreq)) {
high = mNiquistFreq;
}
}
}
}
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