5fe37c6838
First effect factory and effect library API implementation. Also added default effect libraries for reverb and equalizer effects. These libraries are for functional test only and are not fine tuned with regard to audio quality. They will probably be replaced by other implementations before the release. Change-Id: I6868f8612146ae282c64052765c61a52ec789ec8
316 lines
10 KiB
C++
316 lines
10 KiB
C++
/*
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* Copyright 2009, The Android Open Source Project
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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#define LOG_TAG "AudioEqualizer"
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#include <assert.h>
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#include <stdlib.h>
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#include <new>
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#include <utils/Log.h>
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#include "AudioEqualizer.h"
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#include "AudioPeakingFilter.h"
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#include "AudioShelvingFilter.h"
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#include "EffectsMath.h"
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namespace android {
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size_t AudioEqualizer::GetInstanceSize(int nBands) {
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assert(nBands >= 2);
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return sizeof(AudioEqualizer) +
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sizeof(AudioShelvingFilter) * 2 +
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sizeof(AudioPeakingFilter) * (nBands - 2);
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}
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AudioEqualizer * AudioEqualizer::CreateInstance(void * pMem, int nBands,
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int nChannels, int sampleRate,
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const PresetConfig * presets,
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int nPresets) {
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LOGV("AudioEqualizer::CreateInstance(pMem=%p, nBands=%d, nChannels=%d, "
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"sampleRate=%d, nPresets=%d)",
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pMem, nBands, nChannels, sampleRate, nPresets);
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assert(nBands >= 2);
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bool ownMem = false;
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if (pMem == NULL) {
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pMem = malloc(GetInstanceSize(nBands));
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if (pMem == NULL) {
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return NULL;
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}
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ownMem = true;
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}
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return new (pMem) AudioEqualizer(pMem, nBands, nChannels, sampleRate,
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ownMem, presets, nPresets);
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}
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void AudioEqualizer::configure(int nChannels, int sampleRate) {
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LOGV("AudioEqualizer::configure(nChannels=%d, sampleRate=%d)", nChannels,
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sampleRate);
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mpLowShelf->configure(nChannels, sampleRate);
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for (int i = 0; i < mNumPeaking; ++i) {
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mpPeakingFilters[i].configure(nChannels, sampleRate);
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}
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mpHighShelf->configure(nChannels, sampleRate);
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}
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void AudioEqualizer::clear() {
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LOGV("AudioEqualizer::clear()");
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mpLowShelf->clear();
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for (int i = 0; i < mNumPeaking; ++i) {
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mpPeakingFilters[i].clear();
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}
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mpHighShelf->clear();
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}
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void AudioEqualizer::free() {
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LOGV("AudioEqualizer::free()");
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if (mpMem != NULL) {
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::free(mpMem);
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}
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}
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void AudioEqualizer::reset() {
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LOGV("AudioEqualizer::reset()");
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const int32_t bottom = Effects_log2(kMinFreq);
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const int32_t top = Effects_log2(mSampleRate * 500);
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const int32_t jump = (top - bottom) / (mNumPeaking + 2);
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int32_t centerFreq = bottom + jump/2;
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mpLowShelf->reset();
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mpLowShelf->setFrequency(Effects_exp2(centerFreq));
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centerFreq += jump;
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for (int i = 0; i < mNumPeaking; ++i) {
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mpPeakingFilters[i].reset();
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mpPeakingFilters[i].setFrequency(Effects_exp2(centerFreq));
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centerFreq += jump;
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}
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mpHighShelf->reset();
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mpHighShelf->setFrequency(Effects_exp2(centerFreq));
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commit(true);
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mCurPreset = PRESET_CUSTOM;
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}
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void AudioEqualizer::setGain(int band, int32_t millibel) {
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LOGV("AudioEqualizer::setGain(band=%d, millibel=%d)", band, millibel);
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assert(band >= 0 && band < mNumPeaking + 2);
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if (band == 0) {
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mpLowShelf->setGain(millibel);
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} else if (band == mNumPeaking + 1) {
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mpHighShelf->setGain(millibel);
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} else {
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mpPeakingFilters[band - 1].setGain(millibel);
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}
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mCurPreset = PRESET_CUSTOM;
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}
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void AudioEqualizer::setFrequency(int band, uint32_t millihertz) {
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LOGV("AudioEqualizer::setFrequency(band=%d, millihertz=%d)", band,
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millihertz);
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assert(band >= 0 && band < mNumPeaking + 2);
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if (band == 0) {
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mpLowShelf->setFrequency(millihertz);
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} else if (band == mNumPeaking + 1) {
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mpHighShelf->setFrequency(millihertz);
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} else {
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mpPeakingFilters[band - 1].setFrequency(millihertz);
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}
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mCurPreset = PRESET_CUSTOM;
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}
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void AudioEqualizer::setBandwidth(int band, uint32_t cents) {
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LOGV("AudioEqualizer::setBandwidth(band=%d, cents=%d)", band, cents);
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assert(band >= 0 && band < mNumPeaking + 2);
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if (band > 0 && band < mNumPeaking + 1) {
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mpPeakingFilters[band - 1].setBandwidth(cents);
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mCurPreset = PRESET_CUSTOM;
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}
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}
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int32_t AudioEqualizer::getGain(int band) const {
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assert(band >= 0 && band < mNumPeaking + 2);
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if (band == 0) {
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return mpLowShelf->getGain();
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} else if (band == mNumPeaking + 1) {
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return mpHighShelf->getGain();
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} else {
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return mpPeakingFilters[band - 1].getGain();
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}
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}
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uint32_t AudioEqualizer::getFrequency(int band) const {
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assert(band >= 0 && band < mNumPeaking + 2);
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if (band == 0) {
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return mpLowShelf->getFrequency();
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} else if (band == mNumPeaking + 1) {
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return mpHighShelf->getFrequency();
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} else {
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return mpPeakingFilters[band - 1].getFrequency();
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}
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}
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uint32_t AudioEqualizer::getBandwidth(int band) const {
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assert(band >= 0 && band < mNumPeaking + 2);
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if (band == 0 || band == mNumPeaking + 1) {
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return 0;
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} else {
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return mpPeakingFilters[band - 1].getBandwidth();
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}
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}
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void AudioEqualizer::getBandRange(int band, uint32_t & low,
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uint32_t & high) const {
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assert(band >= 0 && band < mNumPeaking + 2);
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if (band == 0) {
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low = 0;
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high = mpLowShelf->getFrequency();
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} else if (band == mNumPeaking + 1) {
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low = mpHighShelf->getFrequency();
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high = mSampleRate * 500;
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} else {
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mpPeakingFilters[band - 1].getBandRange(low, high);
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}
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}
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const char * AudioEqualizer::getPresetName(int preset) const {
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assert(preset < mNumPresets && preset >= PRESET_CUSTOM);
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if (preset == PRESET_CUSTOM) {
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return "Custom";
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} else {
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return mpPresets[preset].name;
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}
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}
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int AudioEqualizer::getNumPresets() const {
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return mNumPresets;
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}
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int AudioEqualizer::getPreset() const {
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return mCurPreset;
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}
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void AudioEqualizer::setPreset(int preset) {
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LOGV("AudioEqualizer::setPreset(preset=%d)", preset);
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assert(preset < mNumPresets && preset >= 0);
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const PresetConfig &presetCfg = mpPresets[preset];
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for (int band = 0; band < (mNumPeaking + 2); ++band) {
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const BandConfig & bandCfg = presetCfg.bandConfigs[band];
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setGain(band, bandCfg.gain);
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setFrequency(band, bandCfg.freq);
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setBandwidth(band, bandCfg.bandwidth);
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}
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mCurPreset = preset;
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}
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void AudioEqualizer::commit(bool immediate) {
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LOGV("AudioEqualizer::commit(immediate=%d)", immediate);
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mpLowShelf->commit(immediate);
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for (int i = 0; i < mNumPeaking; ++i) {
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mpPeakingFilters[i].commit(immediate);
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}
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mpHighShelf->commit(immediate);
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}
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void AudioEqualizer::process(const audio_sample_t * pIn,
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audio_sample_t * pOut,
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int frameCount) {
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// LOGV("AudioEqualizer::process(frameCount=%d)", frameCount);
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mpLowShelf->process(pIn, pOut, frameCount);
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for (int i = 0; i < mNumPeaking; ++i) {
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mpPeakingFilters[i].process(pIn, pOut, frameCount);
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}
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mpHighShelf->process(pIn, pOut, frameCount);
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}
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void AudioEqualizer::enable(bool immediate) {
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LOGV("AudioEqualizer::enable(immediate=%d)", immediate);
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mpLowShelf->enable(immediate);
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for (int i = 0; i < mNumPeaking; ++i) {
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mpPeakingFilters[i].enable(immediate);
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}
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mpHighShelf->enable(immediate);
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}
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void AudioEqualizer::disable(bool immediate) {
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LOGV("AudioEqualizer::disable(immediate=%d)", immediate);
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mpLowShelf->disable(immediate);
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for (int i = 0; i < mNumPeaking; ++i) {
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mpPeakingFilters[i].disable(immediate);
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}
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mpHighShelf->disable(immediate);
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}
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int AudioEqualizer::getMostRelevantBand(uint32_t targetFreq) const {
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// First, find the two bands that the target frequency is between.
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uint32_t low = mpLowShelf->getFrequency();
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if (targetFreq <= low) {
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return 0;
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}
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uint32_t high = mpHighShelf->getFrequency();
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if (targetFreq >= high) {
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return mNumPeaking + 1;
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}
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int band = mNumPeaking;
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for (int i = 0; i < mNumPeaking; ++i) {
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uint32_t freq = mpPeakingFilters[i].getFrequency();
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if (freq >= targetFreq) {
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high = freq;
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band = i;
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break;
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}
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low = freq;
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}
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// Now, low is right below the target and high is right above. See which one
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// is closer on a log scale.
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low = Effects_log2(low);
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high = Effects_log2(high);
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targetFreq = Effects_log2(targetFreq);
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if (high - targetFreq < targetFreq - low) {
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return band + 1;
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} else {
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return band;
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}
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}
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AudioEqualizer::AudioEqualizer(void * pMem, int nBands, int nChannels,
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int sampleRate, bool ownMem,
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const PresetConfig * presets, int nPresets)
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: mSampleRate(sampleRate)
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, mpPresets(presets)
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, mNumPresets(nPresets) {
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assert(pMem != NULL);
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assert(nPresets == 0 || nPresets > 0 && presets != NULL);
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mpMem = ownMem ? pMem : NULL;
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pMem = (char *) pMem + sizeof(AudioEqualizer);
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mpLowShelf = new (pMem) AudioShelvingFilter(AudioShelvingFilter::kLowShelf,
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nChannels, sampleRate);
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pMem = (char *) pMem + sizeof(AudioShelvingFilter);
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mpHighShelf = new (pMem) AudioShelvingFilter(AudioShelvingFilter::kHighShelf,
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nChannels, sampleRate);
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pMem = (char *) pMem + sizeof(AudioShelvingFilter);
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mNumPeaking = nBands - 2;
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if (mNumPeaking > 0) {
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mpPeakingFilters = reinterpret_cast<AudioPeakingFilter *>(pMem);
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for (int i = 0; i < mNumPeaking; ++i) {
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new (&mpPeakingFilters[i]) AudioPeakingFilter(nChannels,
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sampleRate);
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}
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}
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reset();
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}
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}
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