diff --git a/source/funkin/backend/utils/AudioAnalyzer.hx b/source/funkin/backend/utils/AudioAnalyzer.hx index 6364a0ee..0ad9af3d 100644 --- a/source/funkin/backend/utils/AudioAnalyzer.hx +++ b/source/funkin/backend/utils/AudioAnalyzer.hx @@ -12,6 +12,7 @@ import lime.utils.ArrayBuffer; // ORIGINAL CODES FROM YOSH & LUNAR https://github.com/CodenameCrew/YoshiCrafterEngine/blob/main/source/WaveformSprite.hx // REWRITTEN BY RALTYRO class AudioAnalyzer { + public var sound:FlxSound; public var buffer:AudioBuffer; var __toBits:Float; var __wordSize:Int; @@ -23,18 +24,19 @@ class AudioAnalyzer { #end public function new(sound:FlxSound) { - buffer = sound.buffer; + this.sound = sound; + __check(); + } + inline function __check() if (sound.buffer != buffer) { + __bitSize = 1 << ((buffer = sound.buffer).bitsPerSample - 1); __bitUnsignedSize = 1 << buffer.bitsPerSample; - __bitSize = 1 << (buffer.bitsPerSample - 1); __toBits = buffer.sampleRate / 1000 * buffer.channels * (__wordSize = buffer.bitsPerSample >> 3); - - #if lime_vorbis - __prepareVorbis(); - #end + __vorbis = null; } public function analyze(startPos:Float, endPos:Float, ?outSeperate:Array):Float { + __check(); if (buffer.data != null) return __analyze(startPos, endPos, outSeperate); #if lime_vorbis else if (__prepareVorbis()) return __analyzeVorbis(startPos, endPos, outSeperate); @@ -56,7 +58,7 @@ class AudioAnalyzer { 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; + if (__wordSize == 1) b = #if js buf[pos++] #else buf.get(pos++) #end - __bitSize; else { while (w < buffer.bitsPerSample) { b |= #if js buf[pos] #else buf.get(pos) #end << w; @@ -78,43 +80,88 @@ class AudioAnalyzer { #if lime_vorbis // As far i know, only native supports vorbis var __buffer:ArrayBuffer; + var __bufferSize:Int; function __prepareVorbis():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(0x400 * buffer.channels * (buffer.bitsPerSample >> 3)); + __buffer = new ArrayBuffer(__bufferSize = 0x400 * buffer.channels * (buffer.bitsPerSample >> 3)); return true; } return false; } - var prevPos:Int = 0; - inline function __analyzeVorbis(startPos:Float, endPos:Float, ?outSeperate:Array):Float { + inline function __analyzeVorbis(startPos:Float, endPos:Float, ?outSeperate:Array):Float @:privateAccess { var n = Math.floor((endPos - startPos) * __toBits); - if (n < __wordSize) return 0; + if ((n -= n % __wordSize) < __wordSize) return 0; - n -= n % __wordSize; - - if (prevPos != Math.floor(startPos)) { - __vorbis.timeSeek(startPos / 1000); - prevPos = Math.floor(endPos); - } - - var isBigEndian = lime.system.System.endianness == lime.system.Endian.BIG_ENDIAN, bufferSize = __buffer.length; - var max = 0, b = 0, c = 0, useSeperate = outSeperate != null, pos = 0, w = 0, result; + 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); - c = 0; } + + // 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; + var time = backend.bufferTimes[i] * 1000; + + 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); + pos -= pos % __wordSize; + + while (n > 0) { + if (__wordSize == 1) { + b = buf.get(pos++) - __bitSize; + n--; + } + 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 (pos >= size) { + if (++i >= backend.bufferDatas.length) break; + size = backend.bufferSizes[i]; + buf = backend.bufferDatas[i].buffer; + pos = 0; + } + w = b = 0; + } + + startPos = pos / __toBits + time; + // Cannot inline a not final return + } + } + 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)); + n -= (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) { - if (__wordSize == 1) b = __buffer.get(pos) - __bitSize; + if (__wordSize == 1) b = __buffer.get(pos++) - __bitSize; else { while (w < buffer.bitsPerSample) { b |= __buffer.get(pos) << w; diff --git a/source/openfl/media/SoundChannel.hx b/source/openfl/media/SoundChannel.hx index 9de0928e..6ccbf8ac 100644 --- a/source/openfl/media/SoundChannel.hx +++ b/source/openfl/media/SoundChannel.hx @@ -8,8 +8,6 @@ import openfl.events.EventDispatcher; #if lime import lime.media.AudioSource; import lime.media.openal.AL; - -#if lime_cffi import lime._internal.backend.native.NativeAudioSource; #end #end /** @@ -157,28 +155,32 @@ import lime.media.openal.AL; __lastPeakTime = time; __leftPeak = __rightPeak = 0; - #if (lime_cffi && !macro) + + #if !macro if (!__isValid) return false; - var backend = __source.__backend, buffer = __source.buffer; + var buffer = __source.buffer; var wordSize = buffer.bitsPerSample >> 3, bitUnsignedSize = 1 << buffer.bitsPerSample, bitSize = 1 << (buffer.bitsPerSample - 1); - var pos = Math.floor(time * buffer.sampleRate / 1000 * buffer.channels * wordSize), size = 0, i = 0, buf; + var pos = Math.floor(time * buffer.sampleRate / 1000 * buffer.channels * wordSize), size = 0, buf; + #if lime_cffi + var backend = __source.__backend, i = 0; if (backend.streamed) { - buf = backend.bufferDatas[i = NativeAudioSource.STREAM_MAX_BUFFERS - backend.queuedBuffers] #if !js .buffer #end; - size = backend.bufferSize; + size = backend.bufferSizes[i = backend.bufferSizes.length - backend.queuedBuffers]; + buf = backend.bufferDatas[i].buffer; pos -= Math.floor(backend.bufferTimes[i] * buffer.sampleRate * buffer.channels * wordSize); } - else { + else + #end { buf = buffer.data #if !js .buffer #end; size = #if js buf.byteLength #else buf.length #end; } - var n = Math.floor(Math.min(buffer.sampleRate / 80, 512)), c = Math.floor((pos % (wordSize * buffer.channels)) / wordSize), b = 0, w = 0; + var s = Math.floor(Math.min(buffer.sampleRate / 80, 512)), c = Math.floor((pos % (wordSize * buffer.channels)) / wordSize), b = 0, w = 0; pos -= pos % wordSize; - while (n > 0) { - if (wordSize == 1) b = #if js buf[pos] #else buf.get(pos) #end - bitSize; + while (s > 0) { + if (wordSize == 1) b = #if js buf[pos++] #else buf.get(pos++) #end - bitSize; else { while (w < buffer.bitsPerSample) { b |= #if js buf[pos] #else buf.get(pos) #end << w; @@ -189,15 +191,17 @@ import lime.media.openal.AL; } ((c % 2 == 0) ? (if (__leftPeak < b) __leftPeak = b) : (if (__rightPeak < b) __rightPeak = b)); - if (pos >= size) { + if (pos >= size) #if lime_cffi { if (!backend.streamed || ++i >= backend.bufferDatas.length) break; - buf = backend.bufferDatas[i] #if !js .buffer #end; + size = backend.bufferSizes[i]; + buf = backend.bufferDatas[i].buffer; pos = 0; } + #else break; #end if (++c > buffer.channels) { c = 0; - n--; + s--; } w = b = 0; }