package funkin.backend.utils; import flixel.sound.FlxSound; import lime.media.AudioBuffer; import lime.utils.ArrayBufferView.ArrayBufferIO; import lime.utils.ArrayBuffer; #if (lime_cffi && lime_vorbis) import lime.media.vorbis.Vorbis; import lime.media.vorbis.VorbisFile; #end /** * An utility that analyze FlxSounds, * can be used to make waveform or real-time audio visualizer. * * FlxSound.amplitude does work in CNE so if any case if your only checking for peak of current * time, use that instead. **/ class AudioAnalyzer { public static function getByte(buf:ArrayBuffer, pos:Int, wordSize:Int):Int { if (wordSize == 2) return ArrayBufferIO.getInt16(buf, pos); else if (wordSize == 3) { var b = ArrayBufferIO.getUint16(buf, pos) | (buf.get(pos + 2) << 16); if (b > 8388608) return b - 16777216; else return b; } else if (wordSize == 4) return ArrayBufferIO.getInt32(buf, pos); else return ArrayBufferIO.getUint8(buf, pos) - 128; } public var sound:FlxSound; public var buffer:AudioBuffer; public var fftSize:Int; public var byteSize:Int; var __toBits:Float; var __wordSize:Int; var __sampleSize:Int; #if (lime_cffi && lime_vorbis) var __vorbis:VorbisFile; var __buffer:ArrayBuffer; var __bufferSize:Int; #end // analyze var __min:Array = []; var __max:Array = []; var __minByte:Int; var __maxByte:Int; /** * Creates an analyzer for specified FlxSound * @param sound An FlxSound to analyze. **/ public function new(sound:FlxSound, fftSize = 1024) { this.sound = sound; this.fftSize = fftSize; __check(); } function __check() if (sound.buffer != buffer) { byteSize = 1 << ((buffer = sound.buffer).bitsPerSample - 1); __toBits = buffer.sampleRate / 1000 * (__sampleSize = buffer.channels * (__wordSize = buffer.bitsPerSample >> 3)); __min.resize(buffer.channels); __max.resize(buffer.channels); #if (lime_cffi && lime_vorbis) __vorbis = null; #end } /** * TODO: IMPLEMENT FFT **/ public function getLevels(levels:Array, barCount:Int, duration:Float):Float { return 0; } /** * Returns a peak of an attached FlxSound from startPos to endPos in milliseconds. * @param startPos Start Position of the FlxSound in milliseconds. * @param endPos End Position of the FlxSound in milliseconds. * @param outOrOutMin The output minimum value from the analyzer, indices is in channels (0 to -0.5 -> 0 to 0.5) (Optional, if outMax doesn't get passed in, it will be [min, max] with all channels combined instead) * @param outMax The output maximum value from the analyzer, indices is in channels (Optional) * @return Output Amplitude value **/ public function analyze(startPos:Float, endPos:Float, ?outOrOutMin:Array, ?outMax:Array):Float { __minByte = __maxByte = 0; if (outOrOutMin != null) { var f:Float; if (outMax != null) { for (i in 0...buffer.channels) __min[i] = __max[i] = 0; read(startPos, endPos, __analyzeCallback); for (i in 0...buffer.channels) { if (outOrOutMin[i] < (f = __min[i] / byteSize)) outOrOutMin[i] = f; if (outMax[i] < (f = __max[i] / byteSize)) outMax[i] = f; } } else { read(startPos, endPos, __analyzeCallbackSimple); outOrOutMin.resize(2); if (outOrOutMin[0] < (f = __minByte / byteSize)) outOrOutMin[0] = f; if (outOrOutMin[1] < (f = __maxByte / byteSize)) outOrOutMin[1] = f; } } else read(startPos, endPos, __analyzeCallbackSimple); return (__maxByte + __minByte) / byteSize; } function __analyzeCallback(b:Int, c:Int):Void ((b > __max[c]) ? (if ((__max[c] = b) > __maxByte) (__maxByte = b)) : (if (-b > __min[c]) (if ((__min[c] = -b) > __minByte) (__minByte = __min[c])))); function __analyzeCallbackSimple(b:Int, c:Int):Void ((b > __maxByte) ? (__maxByte = b) : (if (-b > __minByte) (__minByte = -b))); public function read(startPos:Float, endPos:Float, callback:AudioAnalyzerCallback) { __check(); if (buffer.data != null) __read(startPos, endPos, callback); #if lime_cffi else if (__canReadStream() && (startPos += __readStream(startPos, endPos, callback)) >= endPos) return; #if lime_vorbis else if (__prepareDecoder()) __readDecoder(startPos, endPos, callback); #end #end } inline function __read(startPos:Float, endPos:Float, callback:AudioAnalyzerCallback) { var pos = Math.floor(startPos * __toBits), end = Math.floor(endPos * __toBits), c = 0; pos -= pos % __sampleSize; end -= end % __sampleSize; while (pos < end) { callback(getByte(buffer.data.buffer, pos, __wordSize), c); if (++c > buffer.channels) c = 0; pos += __wordSize; } } #if lime_cffi inline function __canReadStream():Bool @:privateAccess return sound._source != null && sound._source.__backend != null && sound._source.__backend.streamTimer != null; inline function __readStream(startPos:Float, endPos:Float, callback:AudioAnalyzerCallback):Float @:privateAccess { var backend = sound._source.__backend; var i = backend.bufferSizes.length - backend.queuedBuffers; var time = backend.bufferTimes[i] * 1000; var n = Math.floor((endPos - startPos) * __toBits); if (startPos >= time && startPos < backend.bufferTimes[backend.bufferSizes.length - 1] * 1000) { var pos = Math.floor((startPos - time) * __toBits), buf = backend.bufferDatas[i].buffer, size = backend.bufferSizes[i], c = 0; pos -= pos % __sampleSize; while (n > 0) { callback(getByte(buf, pos, __wordSize), c); if (++c > buffer.channels) c = 0; if ((pos += __wordSize) >= size) { if (++i >= backend.bufferDatas.length) break; buf = backend.bufferDatas[i].buffer; size = backend.bufferSizes[i]; pos = 0; } n -= __wordSize; } } return endPos - (n / __toBits); } #if lime_vorbis inline function __prepareDecoder():Bool @:privateAccess { if (buffer.__srcVorbisFile == null) return __vorbis != null; if (__vorbis != null) return true; if ((__vorbis = buffer.__srcVorbisFile.clone()) != null) { // IM HOPING IT HAVE A GC CLOSURE. __buffer = new ArrayBuffer(__bufferSize = 0x400 * __sampleSize); return true; } return false; } inline function __readDecoder(startPos:Float, endPos:Float, callback:AudioAnalyzerCallback) { var time = startPos / 1000; if (Math.abs(time - __vorbis.timeTell()) > 0.004) { if (startPos < 1) __vorbis.rawSeek(0); else __vorbis.timeSeek(time); } var isBigEndian = lime.system.System.endianness == lime.system.Endian.BIG_ENDIAN, result; var n = Math.floor((endPos - startPos) * __toBits), pos = 0, c = 0; n -= n % __sampleSize; while (n > 0) { result = __vorbis.read(__buffer, 0, n < __bufferSize ? n : __bufferSize, isBigEndian, __wordSize, true); if (result == Vorbis.HOLE) continue; else if (result <= 0) break; while (pos < result) { callback(getByte(__buffer, pos, __wordSize), c); if (++c > buffer.channels) c = 0; if ((pos += __wordSize) >= n) break; } pos = 0; n -= result; } } #end #end } private typedef AudioAnalyzerCallback = Int->Int->Void; /* abstract Complex(Array) { public var real(get, never):Float; inline function get_real() return this[0]; public var imag(get, never):Float; inline function get_imag() return this[1]; public inline function new(real:Float, imag:Float) this = [real, imag]; public static inline function fromReal(real:Float):Complex return new Complex(real, 0); public static inline function exp(w:Float):Complex return new Complex(Math.cos(w), Math.sin(w)); public var angle(get, never):Float; inline function get_angle() return Math.atan2(imag, real); public var magnitude(get, never):Float; inline function get_magnitude():Float return Math.sqrt(real*real + imag*imag); @:op(A + B) public inline function add(rhs:Complex):Complex return new Complex(real + rhs.real, imag + rhs.imag); @:op(A - B) public inline function sub(rhs:Complex):Complex return new Complex(real - rhs.real, imag - rhs.imag); @:op(A * B) public inline function mult(rhs:Complex):Complex return new Complex(real*rhs.real - imag*rhs.imag, real*rhs.imag + imag*rhs.real); @:op(A / B) public inline function div(rhs:Complex):Complex { var m = rhs.magnitude; return new Complex((real*rhs.real + imag*rhs.imag) / m, (imag*rhs.real - real*rhs.imag) / m); } public inline function conj():Complex return new Complex(real, -imag); } */