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/*
* Copyright 2009, 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.
*/
#define LOG_TAG "AudioEqualizer"
#include <assert.h>
#include <stdlib.h>
#include <new>
#include <utils/Log.h>
#include "AudioEqualizer.h"
#include "AudioPeakingFilter.h"
#include "AudioShelvingFilter.h"
#include "EffectsMath.h"
namespace android {
size_t AudioEqualizer::GetInstanceSize(int nBands) {
assert(nBands >= 2);
return sizeof(AudioEqualizer) +
sizeof(AudioShelvingFilter) * 2 +
sizeof(AudioPeakingFilter) * (nBands - 2);
}
AudioEqualizer * AudioEqualizer::CreateInstance(void * pMem, int nBands,
int nChannels, int sampleRate,
const PresetConfig * presets,
int nPresets) {
ALOGV("AudioEqualizer::CreateInstance(pMem=%p, nBands=%d, nChannels=%d, "
"sampleRate=%d, nPresets=%d)",
pMem, nBands, nChannels, sampleRate, nPresets);
assert(nBands >= 2);
bool ownMem = false;
if (pMem == NULL) {
pMem = malloc(GetInstanceSize(nBands));
if (pMem == NULL) {
return NULL;
}
ownMem = true;
}
return new (pMem) AudioEqualizer(pMem, nBands, nChannels, sampleRate,
ownMem, presets, nPresets);
}
void AudioEqualizer::configure(int nChannels, int sampleRate) {
ALOGV("AudioEqualizer::configure(nChannels=%d, sampleRate=%d)", nChannels,
sampleRate);
mpLowShelf->configure(nChannels, sampleRate);
for (int i = 0; i < mNumPeaking; ++i) {
mpPeakingFilters[i].configure(nChannels, sampleRate);
}
mpHighShelf->configure(nChannels, sampleRate);
}
void AudioEqualizer::clear() {
ALOGV("AudioEqualizer::clear()");
mpLowShelf->clear();
for (int i = 0; i < mNumPeaking; ++i) {
mpPeakingFilters[i].clear();
}
mpHighShelf->clear();
}
void AudioEqualizer::free() {
ALOGV("AudioEqualizer::free()");
if (mpMem != NULL) {
::free(mpMem);
}
}
void AudioEqualizer::reset() {
ALOGV("AudioEqualizer::reset()");
const int32_t bottom = Effects_log2(kMinFreq);
const int32_t top = Effects_log2(mSampleRate * 500);
const int32_t jump = (top - bottom) / (mNumPeaking + 2);
int32_t centerFreq = bottom + jump/2;
mpLowShelf->reset();
mpLowShelf->setFrequency(Effects_exp2(centerFreq));
centerFreq += jump;
for (int i = 0; i < mNumPeaking; ++i) {
mpPeakingFilters[i].reset();
mpPeakingFilters[i].setFrequency(Effects_exp2(centerFreq));
centerFreq += jump;
}
mpHighShelf->reset();
mpHighShelf->setFrequency(Effects_exp2(centerFreq));
commit(true);
mCurPreset = PRESET_CUSTOM;
}
void AudioEqualizer::setGain(int band, int32_t millibel) {
ALOGV("AudioEqualizer::setGain(band=%d, millibel=%d)", band, millibel);
assert(band >= 0 && band < mNumPeaking + 2);
if (band == 0) {
mpLowShelf->setGain(millibel);
} else if (band == mNumPeaking + 1) {
mpHighShelf->setGain(millibel);
} else {
mpPeakingFilters[band - 1].setGain(millibel);
}
mCurPreset = PRESET_CUSTOM;
}
void AudioEqualizer::setFrequency(int band, uint32_t millihertz) {
ALOGV("AudioEqualizer::setFrequency(band=%d, millihertz=%d)", band,
millihertz);
assert(band >= 0 && band < mNumPeaking + 2);
if (band == 0) {
mpLowShelf->setFrequency(millihertz);
} else if (band == mNumPeaking + 1) {
mpHighShelf->setFrequency(millihertz);
} else {
mpPeakingFilters[band - 1].setFrequency(millihertz);
}
mCurPreset = PRESET_CUSTOM;
}
void AudioEqualizer::setBandwidth(int band, uint32_t cents) {
ALOGV("AudioEqualizer::setBandwidth(band=%d, cents=%d)", band, cents);
assert(band >= 0 && band < mNumPeaking + 2);
if (band > 0 && band < mNumPeaking + 1) {
mpPeakingFilters[band - 1].setBandwidth(cents);
mCurPreset = PRESET_CUSTOM;
}
}
int32_t AudioEqualizer::getGain(int band) const {
assert(band >= 0 && band < mNumPeaking + 2);
if (band == 0) {
return mpLowShelf->getGain();
} else if (band == mNumPeaking + 1) {
return mpHighShelf->getGain();
} else {
return mpPeakingFilters[band - 1].getGain();
}
}
uint32_t AudioEqualizer::getFrequency(int band) const {
assert(band >= 0 && band < mNumPeaking + 2);
if (band == 0) {
return mpLowShelf->getFrequency();
} else if (band == mNumPeaking + 1) {
return mpHighShelf->getFrequency();
} else {
return mpPeakingFilters[band - 1].getFrequency();
}
}
uint32_t AudioEqualizer::getBandwidth(int band) const {
assert(band >= 0 && band < mNumPeaking + 2);
if (band == 0 || band == mNumPeaking + 1) {
return 0;
} else {
return mpPeakingFilters[band - 1].getBandwidth();
}
}
void AudioEqualizer::getBandRange(int band, uint32_t & low,
uint32_t & high) const {
assert(band >= 0 && band < mNumPeaking + 2);
if (band == 0) {
low = 0;
high = mpLowShelf->getFrequency();
} else if (band == mNumPeaking + 1) {
low = mpHighShelf->getFrequency();
high = mSampleRate * 500;
} else {
mpPeakingFilters[band - 1].getBandRange(low, high);
}
}
const char * AudioEqualizer::getPresetName(int preset) const {
assert(preset < mNumPresets && preset >= PRESET_CUSTOM);
if (preset == PRESET_CUSTOM) {
return "Custom";
} else {
return mpPresets[preset].name;
}
}
int AudioEqualizer::getNumPresets() const {
return mNumPresets;
}
int AudioEqualizer::getPreset() const {
return mCurPreset;
}
void AudioEqualizer::setPreset(int preset) {
ALOGV("AudioEqualizer::setPreset(preset=%d)", preset);
assert(preset < mNumPresets && preset >= 0);
const PresetConfig &presetCfg = mpPresets[preset];
for (int band = 0; band < (mNumPeaking + 2); ++band) {
const BandConfig & bandCfg = presetCfg.bandConfigs[band];
setGain(band, bandCfg.gain);
setFrequency(band, bandCfg.freq);
setBandwidth(band, bandCfg.bandwidth);
}
mCurPreset = preset;
}
void AudioEqualizer::commit(bool immediate) {
ALOGV("AudioEqualizer::commit(immediate=%d)", immediate);
mpLowShelf->commit(immediate);
for (int i = 0; i < mNumPeaking; ++i) {
mpPeakingFilters[i].commit(immediate);
}
mpHighShelf->commit(immediate);
}
void AudioEqualizer::process(const audio_sample_t * pIn,
audio_sample_t * pOut,
int frameCount) {
// ALOGV("AudioEqualizer::process(frameCount=%d)", frameCount);
mpLowShelf->process(pIn, pOut, frameCount);
for (int i = 0; i < mNumPeaking; ++i) {
mpPeakingFilters[i].process(pIn, pOut, frameCount);
}
mpHighShelf->process(pIn, pOut, frameCount);
}
void AudioEqualizer::enable(bool immediate) {
ALOGV("AudioEqualizer::enable(immediate=%d)", immediate);
mpLowShelf->enable(immediate);
for (int i = 0; i < mNumPeaking; ++i) {
mpPeakingFilters[i].enable(immediate);
}
mpHighShelf->enable(immediate);
}
void AudioEqualizer::disable(bool immediate) {
ALOGV("AudioEqualizer::disable(immediate=%d)", immediate);
mpLowShelf->disable(immediate);
for (int i = 0; i < mNumPeaking; ++i) {
mpPeakingFilters[i].disable(immediate);
}
mpHighShelf->disable(immediate);
}
int AudioEqualizer::getMostRelevantBand(uint32_t targetFreq) const {
// First, find the two bands that the target frequency is between.
uint32_t low = mpLowShelf->getFrequency();
if (targetFreq <= low) {
return 0;
}
uint32_t high = mpHighShelf->getFrequency();
if (targetFreq >= high) {
return mNumPeaking + 1;
}
int band = mNumPeaking;
for (int i = 0; i < mNumPeaking; ++i) {
uint32_t freq = mpPeakingFilters[i].getFrequency();
if (freq >= targetFreq) {
high = freq;
band = i;
break;
}
low = freq;
}
// Now, low is right below the target and high is right above. See which one
// is closer on a log scale.
low = Effects_log2(low);
high = Effects_log2(high);
targetFreq = Effects_log2(targetFreq);
if (high - targetFreq < targetFreq - low) {
return band + 1;
} else {
return band;
}
}
AudioEqualizer::AudioEqualizer(void * pMem, int nBands, int nChannels,
int sampleRate, bool ownMem,
const PresetConfig * presets, int nPresets)
: mSampleRate(sampleRate)
, mpPresets(presets)
, mNumPresets(nPresets) {
assert(pMem != NULL);
assert(nPresets == 0 || nPresets > 0 && presets != NULL);
mpMem = ownMem ? pMem : NULL;
pMem = (char *) pMem + sizeof(AudioEqualizer);
mpLowShelf = new (pMem) AudioShelvingFilter(AudioShelvingFilter::kLowShelf,
nChannels, sampleRate);
pMem = (char *) pMem + sizeof(AudioShelvingFilter);
mpHighShelf = new (pMem) AudioShelvingFilter(AudioShelvingFilter::kHighShelf,
nChannels, sampleRate);
pMem = (char *) pMem + sizeof(AudioShelvingFilter);
mNumPeaking = nBands - 2;
if (mNumPeaking > 0) {
mpPeakingFilters = reinterpret_cast<AudioPeakingFilter *>(pMem);
for (int i = 0; i < mNumPeaking; ++i) {
new (&mpPeakingFilters[i]) AudioPeakingFilter(nChannels,
sampleRate);
}
}
reset();
}
}