diff --git a/source/funkin/backend/utils/AudioAnalyzer.hx b/source/funkin/backend/utils/AudioAnalyzer.hx index 0ad9af3d..90d47d38 100644 --- a/source/funkin/backend/utils/AudioAnalyzer.hx +++ b/source/funkin/backend/utils/AudioAnalyzer.hx @@ -18,7 +18,11 @@ class AudioAnalyzer { var __wordSize:Int; var __bitUnsignedSize:Int; var __bitSize:Int; + var __min:Array = []; + var __max:Array = []; + //var __ln10:Float = 2.3025850929940456; + //var __ln2:Float = 0.6931471805599453; #if lime_vorbis var __vorbis:VorbisFile; #end @@ -28,36 +32,47 @@ class AudioAnalyzer { __check(); } + public function getLevels(levels:Array, barCount:Int, duration:Float):Float { + __check(); + // TODO: implement FFT + return 0; + } + inline function __check() if (sound.buffer != buffer) { __bitSize = 1 << ((buffer = sound.buffer).bitsPerSample - 1); __bitUnsignedSize = 1 << buffer.bitsPerSample; __toBits = buffer.sampleRate / 1000 * buffer.channels * (__wordSize = buffer.bitsPerSample >> 3); __vorbis = null; + __min.resize(buffer.channels); + __max.resize(buffer.channels); } - public function analyze(startPos:Float, endPos:Float, ?outSeperate:Array):Float { + /** + * 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 outMin The output minimum value from the analyzer, indices is in channels (0 to -0.5 -> 0 to 0.5) (Optional) + * @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, ?outMin:Array, ?outMax:Array):Float { __check(); - if (buffer.data != null) return __analyze(startPos, endPos, outSeperate); + if (buffer.data != null) return __analyze(startPos, endPos, outMin, outMax); #if lime_vorbis - else if (__prepareVorbis()) return __analyzeVorbis(startPos, endPos, outSeperate); + else if (__prepareVorbis()) return __analyzeVorbis(startPos, endPos, outMin, outMax); #end return 0; } - inline function __analyze(startPos:Float, endPos:Float, ?outSeperate:Array):Float { - var pos = Math.floor(startPos * __toBits), end = Math.floor(endPos * __toBits), buf = buffer.data #if !js .buffer #end; - var max = 0, b = 0, w = 0, c = 0, useSeperate = outSeperate != null; - if (useSeperate) { - while (c < outSeperate.length) outSeperate[c++] *= __bitSize; - while (c++ < buffer.channels) outSeperate.push(0); - c = Math.floor((pos % (__wordSize * buffer.channels)) / __wordSize); - } + function __analyze(startPos:Float, endPos:Float, ?outMin:Array, ?outMax:Array):Float { + __prepareAmplitude(); + var pos = Math.floor(startPos * __toBits), end = Math.floor(endPos * __toBits), buf = buffer.data #if !js .buffer #end; + var c = Math.floor((pos % (__wordSize * buffer.channels)) / __wordSize), b = 0, w = 0; pos -= pos % __wordSize; end -= end % __wordSize; while (pos < end) { - // 8-bit audio data is unsigned (0 is 128, 128 is 256, -128 is 0) if (__wordSize == 1) b = #if js buf[pos++] #else buf.get(pos++) #end - __bitSize; else { while (w < buffer.bitsPerSample) { @@ -67,15 +82,27 @@ class AudioAnalyzer { } if (b > __bitSize) b -= __bitUnsignedSize; } - if (max < b) max = b; - if (useSeperate) { - if (outSeperate[c] < b) outSeperate[c] = b; - if (++c > buffer.channels) c = 0; - } + if (__max[c] < b) __max[c] = b; else if (__min[c] < (b = -b)) __min[c] = b; + if (++c > buffer.channels) c = 0; w = b = 0; } - return max / __bitSize; + return __getAmplitude(outMin, outMax); + } + + inline function __prepareAmplitude() for (i in 0...buffer.channels) __min[i] = __max[i] = 0; + inline function __getAmplitude(?outMin:Array, ?outMax:Array):Float { + var min = 0, max = 0, useOutput = outMin != null && outMax != null, v:Float; + for (i in 0...buffer.channels) { + if (__min[i] > min) min = __min[i]; + if (__max[i] > max) max = __max[i]; + if (useOutput) { + if (outMin[i] < (v = __min[i] / __bitSize)) outMin[i] = v; + if (outMax[i] < (v = __max[i] / __bitSize)) outMax[i] = v; + } + } + + return (max + min) / __bitSize; } #if lime_vorbis // As far i know, only native supports vorbis @@ -91,17 +118,13 @@ class AudioAnalyzer { return false; } - inline function __analyzeVorbis(startPos:Float, endPos:Float, ?outSeperate:Array):Float @:privateAccess { + function __analyzeVorbis(startPos:Float, endPos:Float, ?outMin:Array, ?outMax:Array):Float @:privateAccess { var n = Math.floor((endPos - startPos) * __toBits); if ((n -= n % __wordSize) < __wordSize) return 0; - var pos = 0, max = 0, b = 0, w = 0, c = 0, useSeperate = outSeperate != null; - if (useSeperate) { - while (c < outSeperate.length) outSeperate[c++] *= __bitSize; - while (c++ < buffer.channels) outSeperate.push(0); - } + var pos = 0, c = 0, b = 0, w = 0; + __prepareAmplitude(); - // IT SHOULD BE ALWAYS BACKEND.STREAMED IF THIS GETS CALLED. var backend = sound._source != null ? sound._source.__backend : null; if (backend != null && backend.streamTimer != null) { var i = backend.bufferSizes.length - backend.queuedBuffers; @@ -109,29 +132,21 @@ class AudioAnalyzer { if (startPos >= time && startPos < backend.bufferTimes[backend.bufferSizes.length - 1] * 1000) { var buf = backend.bufferDatas[i].buffer, size = backend.bufferSizes[i]; - pos = Math.floor((startPos - time) * __toBits); - c = Math.floor((pos % (__wordSize * buffer.channels)) / __wordSize); + c = Math.floor(((pos = Math.floor((startPos - time) * __toBits)) % (__wordSize * buffer.channels)) / __wordSize); pos -= pos % __wordSize; while (n > 0) { - if (__wordSize == 1) { - b = buf.get(pos++) - __bitSize; - n--; - } + if (__wordSize == 1) b = buf.get(pos++) - __bitSize; else { while (w < buffer.bitsPerSample) { b |= buf.get(pos) << w; w += 8; pos++; - n--; } if (b > __bitSize) b -= __bitUnsignedSize; } - if (max < b) max = b; - if (useSeperate) { - if (outSeperate[c] < b) outSeperate[c] = b; - if (++c > buffer.channels) c = 0; - } + if (__max[c] < b) __max[c] = b; else if (__min[c] < (b = -b)) __min[c] = b; + if (++c > buffer.channels) c = 0; if (pos >= size) { if (++i >= backend.bufferDatas.length) break; size = backend.bufferSizes[i]; @@ -139,20 +154,20 @@ class AudioAnalyzer { pos = 0; } w = b = 0; + n -= __wordSize; } + if (n == 0) return __getAmplitude(outMin, outMax); startPos = pos / __toBits + time; - // Cannot inline a not final return + pos = c = 0; } } - if (n == 0) return max / __bitSize; if (Math.abs(startPos - __vorbis.timeTell() * 1000) > 4) { if (startPos < 1) __vorbis.rawSeek(0); else __vorbis.timeSeek(startPos / 1000); } - pos = c = 0; var isBigEndian = lime.system.System.endianness == lime.system.Endian.BIG_ENDIAN, result; while (n > 0) { n -= (result = __vorbis.read(__buffer, 0, n < __bufferSize ? n : __bufferSize, isBigEndian, __wordSize, true)); @@ -170,17 +185,14 @@ class AudioAnalyzer { } if (b > __bitSize) b -= __bitUnsignedSize; } - if (max < b) max = b; - if (useSeperate) { - if (outSeperate[c] < b) outSeperate[c] = b; - if (++c > buffer.channels) c = 0; - } + if (__max[c] < b) __max[c] = b; else if (__min[c] < (b = -b)) __min[c] = b; + if (++c > buffer.channels) c = 0; w = b = 0; } pos = 0; } - return max / __bitSize; + return __getAmplitude(outMin, outMax); } #end } \ No newline at end of file