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getFrequenciesFromSamples, Merging Samples

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Ralty
2025-08-03 14:10:53 +07:00
parent 303812218c
commit c5918aaa07
1 changed files with 177 additions and 109 deletions
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source/funkin/backend/utils/AudioAnalyzer.hx
+177 -109
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@@ -10,6 +10,10 @@ import lime.media.vorbis.Vorbis;
import lime.media.vorbis.VorbisFile;
#end
#if (target.threaded)
import sys.thread.Mutex;
#end
typedef AudioAnalyzerCallback = Int->Int->Void;
/**
@@ -49,7 +53,7 @@ final class AudioAnalyzer {
* @param maxDb The maximum decibels to cap (Optional, default -10.0, Above 0 is not recommended).
* @param minFreq The minimum frequency to cap (Optional, default 20.0, Below 8.0 is not recommended).
* @param maxFreq The maximum frequency to cap (Optional, default 22000.0, Above 23000.0 is not recommended).
* @return Output of levels/bars that ranges from 0 to 1
* @return Output of levels/bars that ranges from 0 to 1.
*/
public static function getLevelsFromFrequencies(frequencies:Array<Float>, sampleRate:Int, barCount:Int, ?levels:Array<Float>, ratio = 0.0, minDb = -63.0, maxDb = -10.0, minFreq = 20.0, maxFreq = 22000.0):Array<Float> {
if (levels == null) levels = [];
@@ -86,6 +90,138 @@ final class AudioAnalyzer {
return levels;
}
static var __reverseIndices:Array<Array<Int>> = [];
static var __windows:Array<Array<Float>> = [];
static var __twiddleReals:Array<Array<Float>> = [];
static var __twiddleImags:Array<Array<Float>> = [];
static var __freqReals:Array<Array<Float>> = [];
static var __freqImags:Array<Array<Float>> = [];
#if (target.threaded)
static var __mutex:Mutex = new Mutex();
static var __freqCalculating:Int = 0;
#end
/**
* Gets frequencies from the samples.
* @param samples The samples (can be from AudioAnalyzer.getSamples).
* @param fftN How much samples for the fft to get, Has to be power of two, or it won't work.
* @param useWindowing Should fft related stuff use blackman windowing? (Web AnalyzerNode windowing), Most of the time it's not worth it.
* @param frequencies The output for getting the frequencies, to avoid memory leaks (Optional).
* @return Output of frequencies.
*/
public static function getFrequenciesFromSamples(samples:Array<Float>, fftN = 2048, useWindowing = false, ?frequencies:Array<Float>):Array<Float> {
var log = Math.floor(Math.log(fftN) / 0.6931471805599453);
if (log == 0) throw "AudioAnalyzer.getFrequenciesFromSamples: Cannot insert a fftN of 1";
var i = log - 1;
fftN = 1 << log;
#if (target.threaded) __mutex.acquire(); #end
var reals:Array<Float> = __freqReals[__freqCalculating], imags:Array<Float> = __freqImags[__freqCalculating];
if (reals == null) {
__freqReals.push(reals = []);
__freqImags.push(imags = []);
}
__freqCalculating++;
var reverseIndices:Array<Int> = __reverseIndices[i];
var windows:Array<Float> = __windows[i];
var twiddleReals:Array<Float> = __twiddleReals[i];
var twiddleImags:Array<Float> = __twiddleImags[i];
if (reverseIndices == null) {
__reverseIndices.resize(log);
__windows.resize(log);
__twiddleReals.resize(log);
__twiddleImags.resize(log);
(reverseIndices = []).resize(fftN);
(windows = []).resize(fftN);
(twiddleReals = []).resize(fftN);
(twiddleImags = []).resize(fftN);
var f;
for (i in 0...fftN) {
f = 2 * Math.PI * (i / fftN);
windows[i] = 0.42 - 0.5 * Math.cos(f) + 0.08 * Math.cos(2 * f);
reverseIndices[i] = __bitReverse(i, log);
twiddleReals[i] = Math.cos(-f);
twiddleImags[i] = Math.sin(-f);
}
__reverseIndices[i] = reverseIndices;
__windows[i] = windows;
__twiddleReals[i] = twiddleReals;
__twiddleImags[i] = twiddleImags;
}
#if (target.threaded) __mutex.release(); #end
if (fftN > reals.length) {
reals.resize(fftN);
imags.resize(fftN);
}
if (frequencies == null) frequencies = [];
frequencies.resize(1 << i);
i = samples.length;
while (i > 0) {
i--;
if (useWindowing) reals[reverseIndices[i]] = samples[i] * windows[i];
else reals[reverseIndices[i]] = samples[i];
imags[i] = 0;
}
var size = 1, n = fftN, half = 1, k, i0, i1, t, tr:Float, ti:Float;
while ((size <<= 1) < fftN) {
n >>= 1;
i = 0;
while (i < fftN) {
k = 0;
while (k < half) {
i1 = (i0 = i + k) + half;
t = (k * n) % fftN;
tr = reals[i1] * twiddleReals[t] - imags[i1] * twiddleImags[t];
ti = reals[i1] * twiddleImags[t] + imags[i1] * twiddleReals[t];
reals[i1] = reals[i0] - tr;
imags[i1] = imags[i0] - ti;
reals[i0] += tr;
imags[i0] += ti;
k++;
}
i += size;
}
half = size;
}
tr = 1.0 / fftN;
i = 1 << (log - 1);
while (i > 1) {
i--;
frequencies[i] = 2 * Math.sqrt(reals[i] * reals[i] + imags[i] * imags[i]) * tr;
}
frequencies[0] = Math.sqrt(reals[0] * reals[0] + imags[0] * imags[0]) * tr;
#if (target.threaded) __mutex.acquire(); #end
__freqCalculating--;
#if (target.threaded) __mutex.release(); #end
return frequencies;
}
static function __bitReverse(x:Int, log:Int):Int {
var y = 0, i = log;
while (i > 0) {
y = (y << 1) | (x & 1);
x >>= 1;
i--;
}
return y;
}
/**
* The current sound to analyze.
*/
@@ -97,13 +233,13 @@ final class AudioAnalyzer {
*
* Has to be power of two, or it won't work.
*/
public var fftN(default, set):Int;
public var fftN:Int;
/**
* Should fft related stuff use blackman windowing? (Web AnalyzerNode windowing).
* Most of the time looks bad with this.
*/
public var useWindowingFFT:Bool = false;
public var useWindowingFFT:Bool;
/**
* The current buffer from sound.
@@ -137,31 +273,26 @@ final class AudioAnalyzer {
// samples
var __sampleIndex:Int;
var __sampleChannel:Int;
var __sampleToValue:Float;
var __sampleOutputMerge:Bool;
var __sampleOutputLength:Int;
var __sampleOutput:Array<Float>;
// fft
var __N2:Int;
var __logN:Int;
// frequencies
var __freqSamples:Array<Float>;
var __reverseIndices:Array<Int> = [];
var __windows:Array<Float> = [];
var __twiddleReals:Array<Float> = [];
var __twiddleImags:Array<Float> = [];
var __freqReals:Array<Float> = [];
var __freqImags:Array<Float> = [];
// levels
var __frequencies:Array<Float>;
/**
* Creates an analyzer for specified FlxSound
* @param sound An FlxSound to analyze.
* @param fftN How much samples for fft to get (Optional, default 2048, 4096 is recommended for highest quality).
* @param useWindowingFFT Should fft related stuff use blackman windowing? (Web AnalyzerNode windowing).
*/
public function new(sound:FlxSound, fftN = 2048) {
public function new(sound:FlxSound, fftN = 2048, useWindowingFFT = false) {
this.sound = sound;
this.fftN = fftN;
this.useWindowingFFT = useWindowingFFT;
__check();
}
@@ -180,46 +311,6 @@ final class AudioAnalyzer {
__max.resize(buffer.channels);
}
inline function set_fftN(v:Int):Int {
if (fftN == (fftN = nextPow2(v))) return fftN;
__logN = Math.floor(Math.log(fftN) / Math.log(2));
__N2 = fftN >> 1;
__freqReals.resize(fftN);
__freqImags.resize(fftN);
__reverseIndices.resize(fftN);
__windows.resize(fftN);
__twiddleReals.resize(fftN);
__twiddleImags.resize(fftN);
var f, a;
for (i in 0...fftN) {
f = i / (fftN - 1);
__windows[i] = 0.42 - 0.5 * Math.cos(2 * Math.PI * f) + 0.08 * Math.cos(4 * Math.PI * f);
__reverseIndices[i] = __bitReverse(i);
__twiddleReals[i] = Math.cos(a = -2 * Math.PI * i / fftN);
__twiddleImags[i] = Math.sin(a);
}
return fftN;
}
inline function nextPow2(x:Int):Int {
var p = 1;
while (p < x) p <<= 1;
return p;
}
inline function __bitReverse(x:Int):Int {
var y = 0, i = __logN;
while (i > 0) {
y = (y << 1) | (x & 1);
x >>= 1;
i--;
}
return y;
}
/**
* Gets levels from an attached FlxSound from startPos, basically a minimized of frequencies.
* @param startPos Start Position to get from sound in milliseconds.
@@ -230,7 +321,7 @@ final class AudioAnalyzer {
* @param maxDb The maximum decibels to cap (Optional, default -10.0, Above 0 is not recommended).
* @param minFreq The minimum frequency to cap (Optional, default 20.0, Below 8.0 is not recommended).
* @param maxFreq The maximum frequency to cap (Optional, default 22000.0, Above 23000.0 is not recommended).
* @return Output of levels/bars that ranges from 0 to 1
* @return Output of levels/bars that ranges from 0 to 1.
*/
public function getLevels(startPos:Float, barCount:Int, ?levels:Array<Float>, ?ratio:Float, ?minDb:Float, ?maxDb:Float, ?minFreq:Float, ?maxFreq:Float):Array<Float>
return inline getLevelsFromFrequencies(__frequencies = getFrequencies(startPos, __frequencies), buffer.sampleRate, barCount, levels, ratio, minDb, maxDb, minFreq, maxFreq);
@@ -239,50 +330,10 @@ final class AudioAnalyzer {
* Gets frequencies from an attached FlxSound from startPos.
* @param startPos Start Position to get from sound in milliseconds.
* @param frequencies The output for getting the frequencies, to avoid memory leaks (Optional).
* @return Output of frequencies
* @return Output of frequencies.
*/
public function getFrequencies(startPos:Float, ?frequencies:Array<Float>):Array<Float> {
__freqSamples = getSamples(startPos, fftN, true, __freqSamples);
if (frequencies == null) frequencies = [];
frequencies.resize(__N2);
var i = fftN;
while (i > 0) {
i--;
__freqReals[__reverseIndices[i]] = __freqSamples[i] * (useWindowingFFT ? __windows[i] : 1);
__freqImags[i] = 0;
}
var size = 1, n = fftN, half = 1, k, i0, i1, t, tr:Float, ti:Float;
while ((size <<= 1) < fftN) {
n >>= 1;
i = 0;
while (i < fftN) {
k = 0;
while (k < half) {
i1 = (i0 = i + k) + half;
t = (k * n) % fftN;
tr = __freqReals[i1] * __twiddleReals[t] - __freqImags[i1] * __twiddleImags[t];
ti = __freqReals[i1] * __twiddleImags[t] + __freqImags[i1] * __twiddleReals[t];
__freqReals[i1] = __freqReals[i0] - tr;
__freqImags[i1] = __freqImags[i0] - ti;
__freqReals[i0] += tr;
__freqImags[i0] += ti;
k++;
}
i += size;
}
half = size;
}
frequencies[i = 0] = Math.sqrt(__freqReals[0] * __freqReals[0] + __freqImags[0] * __freqImags[0]) * (tr = 1.0 / fftN);
while (++i < __N2) frequencies[i] = 2 * Math.sqrt(__freqReals[i] * __freqReals[i] + __freqImags[i] * __freqImags[i]) * tr;
return frequencies;
}
public function getFrequencies(startPos:Float, ?frequencies:Array<Float>):Array<Float>
return inline getFrequenciesFromSamples(__freqSamples = getSamples(startPos, fftN, true, __freqSamples), fftN, useWindowingFFT, frequencies);
/**
* Analyzes an attached FlxSound from startPos to endPos in milliseconds to get the amplitudes.
@@ -329,32 +380,49 @@ final class AudioAnalyzer {
* @param startPos Start Position to get from sound in milliseconds.
* @param length Length of Samples.
* @param mono Merge all of the byte channels of samples in one channel instead (Optional).
* @param Output that gets passed into this function (Optional).
* @return Output of
* @param channel What channels to get from? (-1 == All Channels, Optional, this will be ignored if mono is enabled).
* @param output An Output that gets passed into this function, usually for to avoid memory leaks (Optional).
* @param outputMerge Merge with previous values (Optional, default false).
* @return Output of samples.
*/
public function getSamples(startPos:Float, length:Int, mono = true, ?output:Array<Float>):Array<Float> {
((output == null) ? (__sampleOutput = output = []) : (__sampleOutput = output)).resize(__sampleOutputLength = length * (mono ? 1 : buffer.channels));
public function getSamples(startPos:Float, length:Int, mono = true, channel = -1, ?output:Array<Float>, ?outputMerge = false):Array<Float> {
((!mono && (__sampleChannel = channel) == -1) ? (__sampleOutputLength = length * buffer.channels) : (__sampleOutputLength = length));
((output == null) ? (__sampleOutput = output = []) : (__sampleOutput = output)).resize(__sampleOutputLength);
((mono) ? (__sampleToValue = 1.0 / (byteSize * buffer.channels)) : (__sampleToValue = 1.0 / byteSize));
__sampleOutputMerge = outputMerge;
__sampleIndex = 0;
__check();
__read(startPos, startPos + (length / __toBits * buffer.channels), mono ? __getSamplesCallbackMerge : __getSamplesCallback);
__read(startPos, startPos + (length / __toBits * buffer.channels), mono ? __getSamplesCallbackMono : (channel == -1 ? __getSamplesCallback : __getSamplesCallbackChannel));
__sampleOutput = null;
return output;
}
function __getSamplesCallbackMerge(b:Int, c:Int):Void if (__sampleIndex < __sampleOutputLength) {
if (c == 0) __sampleOutput[__sampleIndex] = b / buffer.channels / byteSize;
function __getSamplesCallbackMono(b:Int, c:Int):Void if (__sampleIndex < __sampleOutputLength) {
if (c == 0) {
if (__sampleOutputMerge) __sampleOutput[__sampleIndex] += b * __sampleToValue;
else __sampleOutput[__sampleIndex] = b * __sampleToValue;
}
else if (c == buffer.channels) {
__sampleOutput[__sampleIndex] += b / buffer.channels / byteSize;
__sampleOutput[__sampleIndex] += b * __sampleToValue;
__sampleIndex++;
}
else
__sampleOutput[__sampleIndex] += b / buffer.channels / byteSize;
__sampleOutput[__sampleIndex] += b * __sampleToValue;
}
function __getSamplesCallbackChannel(b:Int, c:Int):Void if (__sampleIndex < __sampleOutputLength) {
if (c == __sampleChannel) {
if (__sampleOutputMerge) __sampleOutput[__sampleIndex] += b * __sampleToValue;
else __sampleOutput[__sampleIndex] = b * __sampleToValue;
__sampleIndex++;
}
}
function __getSamplesCallback(b:Int, c:Int):Void if (__sampleIndex < __sampleOutputLength) {
__sampleOutput[__sampleIndex] = b / byteSize;
if (__sampleOutputMerge) __sampleOutput[__sampleIndex] += b * __sampleToValue;
else __sampleOutput[__sampleIndex] = b * __sampleToValue;
__sampleIndex++;
}