Stable Reworked Audio
- Reworked Internal Audio to work with Streaming Vorbis Files (for now?) - FlxSound.amplitude, FlxSound.amplitudeLeft, FlxSound.amplitudeRight, SoundChannel.leftPeak, SoundChannel.rightPeak now works. (Streamed Audio on this seems to be half-broken) - Reworked AudioAnalyzer to work with all type of audios (hopefully) and work with Streamed Audio. - Redo AudioSwitchFix to work in time instead of when switching states. gosh, and all that i hope it doesnt break anything.
This commit is contained in:
Ralty
@@ -16,7 +16,6 @@ import lime.media.vorbis.VorbisFile;
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import lime.math.Vector4;
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import lime.media.AudioBuffer;
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import lime.media.AudioSource;
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import lime.system.System;
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import lime.utils.ArrayBufferView;
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#if !lime_debug
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@@ -26,7 +25,7 @@ import lime.utils.ArrayBufferView;
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@:access(lime.media.AudioBuffer)
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@:access(lime.utils.ArrayBufferView)
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class NativeAudioSource {
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private static var STREAM_BUFFER_SIZE:Int = 0x4000;
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private static var STREAM_BUFFER_SIZE:Int = 0x4000; // what if it was odd channels (excluding mono).
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private static var STREAM_MAX_BUFFERS:Int = 32;
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private static var STREAM_TIMER_FREQUENCY:Int = 100;
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private static var STREAM_BUFFER_FREQUENCY:Int = 6;
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@@ -47,6 +46,7 @@ class NativeAudioSource {
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private var streamTimer:Timer;
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private var bufferDatas:Array<ArrayBufferView>;
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private var bufferTimes:Array<Float>;
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private var bufferSizes:Array<Int>;
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private var requestBuffers:Int;
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private var queuedBuffers:Int;
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private var toLoop:Int;
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@@ -74,30 +74,48 @@ class NativeAudioSource {
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buffers = null;
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bufferDatas = null;
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bufferTimes = null;
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bufferSizes = null;
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unusedBuffers = null;
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}
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public function init() {
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if (handle != null) return;
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var buffer = parent.buffer;
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if (buffer.channels == 1) format = buffer.bitsPerSample == 8 ? AL.FORMAT_MONO8 : AL.FORMAT_MONO16;
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else format = buffer.bitsPerSample == 8 ? AL.FORMAT_STEREO8 : AL.FORMAT_STEREO16;
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if (disposed = (handle = AL.createSource()) == null) return;
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AL.sourcef(handle, AL.MAX_GAIN, 10);
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if (streamed = buffer.__srcVorbisFile != null) {
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dataLength = (samples = getFloat(buffer.__srcVorbisFile.pcmTotal())) * buffer.channels * (buffer.bitsPerSample == 8 ? 1 : 2);
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var buffer = parent.buffer;
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var channels = buffer.channels, bitsPerSample = buffer.bitsPerSample;
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var constructor = buffer.bitsPerSample == 8 ? Int8 : Int16;
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// Default is just AL.FORMAT_MONO8 if it doesn't match any to avoid yo ears getting blasted
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var isCreativeXFi = AL.getString(AL.RENDERER) == "X-Fi";
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if (channels > 2 && (isCreativeXFi || AL.isExtensionPresent("AL_EXT_MCFORMATS"))) { // https://github.com/openalext/openalext/blob/master/AL_EXT_MCFORMATS.txt
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if (channels == 3) format = bitsPerSample == 32 ? 0x1209 : (bitsPerSample == 16 ? 0x1208 : 0x1207);
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else if (channels == 4) format = bitsPerSample == 32 ? 0x1206 : (bitsPerSample == 16 ? 0x1205 : 0x1204);
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else if (channels == 6) format = bitsPerSample == 32 ? 0x120C : (bitsPerSample == 16 ? 0x120B : 0x120A);
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else if (channels == 7) format = bitsPerSample == 32 ? 0x120F : (bitsPerSample == 16 ? 0x120E : 0x120D);
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else if (channels == 8) format = bitsPerSample == 32 ? 0x1212 : (bitsPerSample == 16 ? 0x1211 : 0x1210);
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}
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// The 32 bit for STEREO and MONO are undocumented? https://github.com/kcat/openal-soft/issues/934, though it only works in X-Fi OpenAL??
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else if (bitsPerSample == 32 && isCreativeXFi) format = channels == 2 ? 0x1203 : 0x1202;
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else if (channels == 2) format = bitsPerSample == 16 ? AL.FORMAT_STEREO16 : AL.FORMAT_STEREO8;
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else format = bitsPerSample == 16 ? AL.FORMAT_MONO16 : AL.FORMAT_MONO8;
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#if lime_vorbis
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if (buffer.__srcVorbisFile != null) {
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streamed = true;
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dataLength = (samples = getFloat(buffer.__srcVorbisFile.pcmTotal())) * buffer.channels * (buffer.bitsPerSample >> 3);
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var constructor = buffer.bitsPerSample == 32 ? Int32 : buffer.bitsPerSample == 16 ? Int16 : Int8;
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buffers = AL.genBuffers(STREAM_MAX_BUFFERS);
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bufferDatas = [for (i in 0...STREAM_MAX_BUFFERS) new ArrayBufferView(STREAM_BUFFER_SIZE, constructor)];
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bufferTimes = [for (i in 0...STREAM_MAX_BUFFERS) 0];
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bufferSizes = [for (i in 0...STREAM_MAX_BUFFERS) 0];
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unusedBuffers = [];
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}
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else {
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samples = (dataLength = getFloat(Int64.make(0, buffer.data.length))) / buffer.channels / (buffer.bitsPerSample == 8 ? 1 : 2);
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else #end {
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streamed = false;
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samples = (dataLength = getFloat(Int64.make(0, buffer.data.length))) / buffer.channels / (buffer.bitsPerSample >> 3);
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if (buffer.__srcBuffer == null && (buffer.__srcBuffer = AL.createBuffer()) != null)
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AL.bufferData(buffer.__srcBuffer, format, buffer.data, buffer.data.length, buffer.sampleRate);
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@@ -129,12 +147,8 @@ class NativeAudioSource {
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public function stop() {
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if (!disposed) {
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if (AL.getSourcei(handle, AL.SOURCE_STATE) != AL.STOPPED) AL.sourceStop(handle);
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if (streamed) {
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AL.sourceUnqueueBuffers(handle, AL.getSourcei(handle, AL.BUFFERS_QUEUED));
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unusedBuffers.resize(0);
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}
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requestBuffers = queuedBuffers = toLoop = 0;
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AL.sourceStop(handle);
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toLoop = 0;
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}
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playing = false;
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@@ -152,31 +166,29 @@ class NativeAudioSource {
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// just incase if we support more than 1 in the future?? which i doubt
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private function streamTell():Float {
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#if lime_vorbis
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/*if (parent.buffer.__srcVorbisFile != null) */return parent.buffer.__srcVorbisFile.timeTell();
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#end
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return parent.buffer.__srcVorbisFile.timeTell();
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#else
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return 0;
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#end
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}
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private function streamSeek(samples:Int64) {
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#if lime_vorbis
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//if (parent.buffer.__srcVorbisFile != null) {
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// There's apparently a bug in libvorbis <= 1.3.4, hhttps://github.com/xiph/vorbis/blob/master/CHANGES#L39
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// But it seems to also happen in this lime libvorbis version which is 1.3.7...? -ralty
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if (samples < 4) parent.buffer.__srcVorbisFile.rawSeek(0); else parent.buffer.__srcVorbisFile.pcmSeek(samples);
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// break;
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//}
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// There's apparently a bug in libvorbis <= 1.3.4, hhttps://github.com/xiph/vorbis/blob/master/CHANGES#L39
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// But it seems to also happen in this lime libvorbis version which is 1.3.7...? -ralty
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if (samples < 4) parent.buffer.__srcVorbisFile.rawSeek(0); else parent.buffer.__srcVorbisFile.pcmSeek(samples);
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#end
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}
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private function readToBufferData(data:ArrayBufferView):Int {
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var total = 0, result = 0, wasEOF = false;
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var bigEndian = System.endianness == lime.system.Endian.BIG_ENDIAN, wordSize = parent.buffer.bitsPerSample == 8 ? 1 : 2;
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#if lime_vorbis
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var isBigEndian = lime.system.System.endianness == lime.system.Endian.BIG_ENDIAN, wordSize = parent.buffer.bitsPerSample >> 3;
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var size = dataLength - (streamTell() * parent.buffer.sampleRate * parent.buffer.channels * wordSize);
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var n:Int = size < STREAM_BUFFER_SIZE ? Math.floor(size) : STREAM_BUFFER_SIZE;
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var vorbisFile = parent.buffer.__srcVorbisFile;
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var total = 0, result = 0, wasEOF = false;
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while (total < STREAM_BUFFER_SIZE) {
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result = n > 0 ? vorbisFile.read(data.buffer, total, n, bigEndian, wordSize, true) : 0;
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result = n > 0 ? parent.buffer.__srcVorbisFile.read(data.buffer, total, n, isBigEndian, wordSize, true) : 0;
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if (result == Vorbis.HOLE) continue;
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else if (result <= Vorbis.EREAD) break;
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@@ -205,36 +217,37 @@ class NativeAudioSource {
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return result;
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}
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return total;
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}
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private function cycleBuffer() {
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var i = STREAM_MAX_BUFFERS - (queuedBuffers = requestBuffers), n = STREAM_MAX_BUFFERS - 1;
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var data = bufferDatas[i], time = bufferTimes[i];
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while (i < n) {
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bufferDatas[i] = bufferDatas[i + 1];
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bufferTimes[i] = bufferTimes[++i];
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}
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bufferDatas[n] = data;
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bufferTimes[n] = time;
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#else
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return 0;
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#end
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}
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private function fillBuffer(buffer:ALBuffer):Int {
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var i = STREAM_MAX_BUFFERS - requestBuffers, time = streamTell();
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var decoded = readToBufferData(bufferDatas[i]);
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var i = STREAM_MAX_BUFFERS - requestBuffers;
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var data = bufferDatas[i], time = streamTell();
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var decoded = readToBufferData(data);
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if (decoded > 0) {
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AL.bufferData(buffer, format, bufferDatas[i], decoded, parent.buffer.sampleRate);
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bufferTimes[i] = time;
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cycleBuffer();
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AL.bufferData(buffer, format, data, decoded, parent.buffer.sampleRate);
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var n = STREAM_MAX_BUFFERS - 1;
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while (i < n) {
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bufferDatas[i] = bufferDatas[i + 1];
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bufferTimes[i] = bufferTimes[i + 1];
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bufferSizes[i] = bufferSizes[++i];
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}
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queuedBuffers = requestBuffers;
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bufferDatas[n] = data;
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bufferTimes[n] = time;
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bufferSizes[n] = decoded;
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}
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return decoded;
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}
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private function streamRun() {
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#if lime_vorbis
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if (parent == null || parent.buffer == null || handle == null) return dispose();
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if (!playing || disposed) stopStreamTimer();
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if (!playing || disposed) return stopStreamTimer();
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try {
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var processed = AL.getSourcei(handle, AL.BUFFERS_PROCESSED), n = STREAM_BUFFER_FREQUENCY, buffer;
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@@ -264,25 +277,28 @@ class NativeAudioSource {
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else requestBuffers--;
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}
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}
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if (AL.getSourcei(handle, AL.SOURCE_STATE) == AL.STOPPED) {
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AL.sourcePlay(handle);
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resetTimer((getLength() - getCurrentTime()) / getPitch());
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}
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}
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catch(e) trace('NativeAudioSource Bug! in streamRun $e');
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#end
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if (AL.getSourcei(handle, AL.SOURCE_STATE) == AL.STOPPED) AL.sourcePlay(handle);
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if (streamEnded && timer == null) resetTimer((getLength() - getCurrentTime()) / getPitch());
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}
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// Timers
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inline function stopStreamTimer() if (streamTimer != null) streamTimer.stop();
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inline function stopStreamTimer() {
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if (streamTimer != null) streamTimer.stop();
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streamTimer = null;
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}
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private function resetStreamTimer() {
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stopStreamTimer();
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if (streamed) (streamTimer = new Timer(STREAM_TIMER_FREQUENCY)).run = streamRun;
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}
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inline function stopTimer() if (timer != null) timer.stop();
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inline function stopTimer() {
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if (timer != null) timer.stop();
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timer = null;
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}
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private function resetTimer(timeRemaining:Float) {
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stopTimer();
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@@ -290,6 +306,8 @@ class NativeAudioSource {
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}
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private function timer_onRun() {
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stopTimer();
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var timeRemaining = (getLength() - getCurrentTime()) / getPitch();
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if (timeRemaining > 30 && AL.getSourcei(handle, AL.SOURCE_STATE) == AL.PLAYING) {
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resetTimer(timeRemaining);
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@@ -317,66 +335,69 @@ class NativeAudioSource {
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// Get & Set Methods
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public function getCurrentTime():Float {
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if (completed) return getLength();
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else if (!disposed) {
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var time = ((streamed ? bufferTimes[STREAM_MAX_BUFFERS - queuedBuffers] : 0) + AL.getSourcef(handle, AL.SEC_OFFSET)) * 1000 - parent.offset;
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if (loops > 0 && time > getLength()) {
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var start = (loopTime != null ? Math.max(0, loopTime) : 0) + parent.offset;
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return ((time - start) % (getLength() - start)) + start;
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}
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else if (time > 0) return time;
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else if (disposed) return 0;
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var time = AL.getSourcef(handle, AL.SAMPLE_OFFSET) / parent.buffer.sampleRate;
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if (streamed) time += bufferTimes[STREAM_MAX_BUFFERS - queuedBuffers];
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time *= 1000;
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var length = if (length == null) getRealLength(); else length;
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if (loops > 0 && time > length) {
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var start = loopTime != null ? Math.max(0, loopTime) : 0;
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return ((time - start) % (length - start)) + start - parent.offset;
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}
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return 0;
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return time - parent.offset;
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}
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public function setCurrentTime(value:Float):Float {
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if (disposed) return value;
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if (streamed) {
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AL.sourceStop(handle);
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AL.sourceUnqueueBuffers(handle, AL.getSourcei(handle, AL.BUFFERS_QUEUED));
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unusedBuffers.resize(0);
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streamSeek(getSamples(value + parent.offset));
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}
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else
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AL.sourcei(handle, AL.SAMPLE_OFFSET, getSamples(value + parent.offset));
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if (streamed) AL.sourceStop(handle);
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else AL.sourcei(handle, AL.SAMPLE_OFFSET, getSamples(value + parent.offset));
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if (playing) {
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var timeRemaining = (getLength() - value) / AL.getSourcef(handle, AL.PITCH);
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if (timeRemaining > 0) {
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var timeRemaining = (getLength() - value) / getPitch();
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if (completed = timeRemaining <= 0) complete();
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else {
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if (streamed) {
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AL.sourceUnqueueBuffers(handle, AL.getSourcei(handle, AL.BUFFERS_QUEUED));
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unusedBuffers.resize(0);
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streamSeek(getSamples(value + parent.offset));
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for (i in 0...(requestBuffers = queuedBuffers = STREAM_BUFFER_FREQUENCY))
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if (fillBuffer(buffers[i]) > 0) AL.sourceQueueBuffer(handle, buffers[i]);
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resetStreamTimer();
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}
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resetTimer(timeRemaining);
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else
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resetTimer(timeRemaining);
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AL.sourcePlay(handle);
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}
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else {
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stopStreamTimer();
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complete();
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}
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}
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if (!playing && streamed) bufferTimes[STREAM_MAX_BUFFERS - (requestBuffers = queuedBuffers = 1)] = value / 1000;
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return value;
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}
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inline private function getRealLength():Float return samples / parent.buffer.sampleRate * 1000;
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public function getLength():Null<Float> {
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return if (length == null) getRealLength() - parent.offset;
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else length - parent.offset;
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}
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public function getLength():Null<Float>
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return if (disposed) 0; else if (length == null) getRealLength() - parent.offset; else length - parent.offset;
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public function setLength(value:Null<Float>):Null<Float> {
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if (value == length || disposed) return length = value;
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var buffer = parent.buffer, wordSize = buffer.bitsPerSample == 8 ? 1 : 2;
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var buffer = parent.buffer, wordSize = buffer.bitsPerSample >> 3;
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if ((length = value) == null) dataLength = streamed ? samples * buffer.channels * wordSize : getFloat(Int64.make(0, buffer.data.length));
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else dataLength = Math.ffloor(Math.max(0, Math.min(value, getRealLength())) / 1000 * buffer.sampleRate) * buffer.channels * wordSize;
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if (playing) {
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if (streamed) setCurrentTime(getCurrentTime());
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if (streamed) {
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if (streamEnded || toLoop > 0) setCurrentTime(getCurrentTime());
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}
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else {
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var timeRemaining = ((getLength() - parent.offset) - getCurrentTime()) / getPitch();
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if (timeRemaining > 0) resetTimer(timeRemaining);
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@@ -392,12 +413,9 @@ class NativeAudioSource {
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public function setPitch(value:Float):Float {
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if (disposed || value == AL.getSourcef(handle, AL.PITCH)) return value;
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if (playing) {
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if (value <= 0) stopTimer();
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else {
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var timeRemaining = (getLength() - getCurrentTime()) / value;
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if (timeRemaining > 0) resetTimer(timeRemaining);
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}
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if (playing && (streamEnded || !streamed)) {
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var timeRemaining = (getLength() - getCurrentTime()) / value;
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if (timeRemaining > 0) resetTimer(timeRemaining);
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}
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AL.sourcef(handle, AL.PITCH, value);
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return value;
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@@ -454,6 +472,6 @@ class NativeAudioSource {
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return position;
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}
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inline private function getFloat(x:Int64):Float return x.high * 4294967296. + (x.low >>> 0);
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inline private function getFloat(x:Int64):Float return x.high * 4294967296. + (x.low >> 0);
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inline private function getSamples(ms:Float):Int64 return Int64.fromFloat(Math.max(0, Math.min(ms / 1000 * parent.buffer.sampleRate, samples)));
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}
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}
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