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https://github.com/KubaPro010/fm-dx-webserver.git
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3LAS implementation
This commit is contained in:
NoobishSVK
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/*
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MPEG audio format reader is part of 3LAS (Low Latency Live Audio Streaming)
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https://github.com/JoJoBond/3LAS
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*/
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var __extends = (this && this.__extends) || (function () {
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var extendStatics = function (d, b) {
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extendStatics = Object.setPrototypeOf ||
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({ __proto__: [] } instanceof Array && function (d, b) { d.__proto__ = b; }) ||
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function (d, b) { for (var p in b) if (Object.prototype.hasOwnProperty.call(b, p)) d[p] = b[p]; };
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return extendStatics(d, b);
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};
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return function (d, b) {
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if (typeof b !== "function" && b !== null)
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throw new TypeError("Class extends value " + String(b) + " is not a constructor or null");
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extendStatics(d, b);
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function __() { this.constructor = d; }
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d.prototype = b === null ? Object.create(b) : (__.prototype = b.prototype, new __());
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};
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})();
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var MPEGFrameInfo = /** @class */ (function () {
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function MPEGFrameInfo(data, sampleCount, sampleRate) {
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this.Data = data;
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this.SampleCount = sampleCount;
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this.SampleRate = sampleRate;
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}
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return MPEGFrameInfo;
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}());
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var AudioFormatReader_MPEG = /** @class */ (function (_super) {
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__extends(AudioFormatReader_MPEG, _super);
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function AudioFormatReader_MPEG(audio, logger, errorCallback, beforeDecodeCheck, dataReadyCallback, addId3Tag, minDecodeFrames) {
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var _this = _super.call(this, audio, logger, errorCallback, beforeDecodeCheck, dataReadyCallback) || this;
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_this._OnDecodeSuccess = _this.OnDecodeSuccess.bind(_this);
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_this._OnDecodeError = _this.OnDecodeError.bind(_this);
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_this.AddId3Tag = addId3Tag;
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_this.MinDecodeFrames = minDecodeFrames;
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_this.Frames = new Array();
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_this.FrameStartIdx = -1;
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_this.FrameEndIdx = -1;
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_this.FrameSamples = 0;
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_this.FrameSampleRate = 0;
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_this.TimeBudget = 0;
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return _this;
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}
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// Deletes all frames from the databuffer and framearray and all samples from the samplearray
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AudioFormatReader_MPEG.prototype.PurgeData = function () {
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_super.prototype.PurgeData.call(this);
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this.Frames = new Array();
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this.FrameStartIdx = -1;
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this.FrameEndIdx = -1;
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this.FrameSamples = 0;
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this.FrameSampleRate = 0;
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this.TimeBudget = 0;
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};
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// Extracts all currently possible frames
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AudioFormatReader_MPEG.prototype.ExtractAll = function () {
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// Look for frames
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this.FindFrame();
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// Repeat as long as we can extract frames
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while (this.CanExtractFrame()) {
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// Extract frame and push into array
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this.Frames.push(this.ExtractFrame());
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// Look for frames
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this.FindFrame();
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}
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// Check if we have enough frames to decode
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if (this.Frames.length >= this.MinDecodeFrames) {
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// Note:
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// =====
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// mp3 frames have an overlap of [granule size] so we can't use the first or last [granule size] samples.
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// [granule size] is equal to half of a [frame size] in samples (using the mp3's sample rate).
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// Sum up the playback time of each decoded frame and data buffer lengths
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// Note: Since mp3-Frames overlap by half of their sample-length we expect the
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// first and last frame to be only half as long. Some decoders will still output
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// the full frame length by adding zeros.
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var bufferLength = 0;
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var expectedTotalPlayTime_1 = 0;
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expectedTotalPlayTime_1 += this.Frames[0].SampleCount / this.Frames[0].SampleRate / 2.0; // Only half of data is usable due to overlap
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bufferLength += this.Frames[0].Data.length;
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for (var i = 1; i < this.Frames.length - 1; i++) {
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expectedTotalPlayTime_1 += this.Frames[i].SampleCount / this.Frames[i].SampleRate;
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bufferLength += this.Frames[i].Data.length;
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}
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expectedTotalPlayTime_1 += this.Frames[this.Frames.length - 1].SampleCount / this.Frames[this.Frames.length - 1].SampleRate / 2.0; // Only half of data is usable due to overlap
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bufferLength += this.Frames[this.Frames.length - 1].Data.length;
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// If needed, add some space for the ID3v2 tag
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if (this.AddId3Tag) {
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bufferLength += AudioFormatReader_MPEG.Id3v2Tag.length;
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}
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// Create a buffer long enough to hold everything
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var decodeBuffer = new Uint8Array(bufferLength);
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var offset = 0;
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// If needed, add ID3v2 tag to beginning of buffer
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if (this.AddId3Tag) {
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decodeBuffer.set(AudioFormatReader_MPEG.Id3v2Tag, offset);
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offset += AudioFormatReader_MPEG.Id3v2Tag.length;
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}
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// Add the frames to the window
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for (var i = 0; i < this.Frames.length; i++) {
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decodeBuffer.set(this.Frames[i].Data, offset);
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offset += this.Frames[i].Data.length;
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}
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// Remove the used frames from the array
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this.Frames.splice(0, this.Frames.length - 1);
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// Increment Id
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var id_1 = this.Id++;
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// Check if decoded frames might be too far back in the past
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if (!this.OnBeforeDecode(id_1, expectedTotalPlayTime_1))
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return;
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// Push window to the decoder
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this.Audio.decodeAudioData(decodeBuffer.buffer, (function (decodedData) {
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var _id = id_1;
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var _expectedTotalPlayTime = expectedTotalPlayTime_1;
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this._OnDecodeSuccess(decodedData, _id, _expectedTotalPlayTime);
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}).bind(this), this._OnDecodeError.bind(this));
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}
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};
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// Finds frame boundries within the data buffer
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AudioFormatReader_MPEG.prototype.FindFrame = function () {
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// Find frame start
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if (this.FrameStartIdx < 0) {
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var i = 0;
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// Make sure we don't exceed array bounds
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while ((i + 1) < this.DataBuffer.length) {
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// Look for MPEG sync word
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if (this.DataBuffer[i] == 0xFF && (this.DataBuffer[i + 1] & 0xE0) == 0xE0) {
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// Sync found, set frame start
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this.FrameStartIdx = i;
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break;
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}
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i++;
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}
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}
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// Find frame end
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if (this.FrameStartIdx >= 0 && this.FrameEndIdx < 0) {
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// Check if we have enough data to process the header
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if ((this.FrameStartIdx + 2) < this.DataBuffer.length) {
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// Get header data
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// Version index
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var ver = (this.DataBuffer[this.FrameStartIdx + 1] & 0x18) >>> 3;
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// Layer index
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var lyr = (this.DataBuffer[this.FrameStartIdx + 1] & 0x06) >>> 1;
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// Padding? 0/1
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var pad = (this.DataBuffer[this.FrameStartIdx + 2] & 0x02) >>> 1;
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// Bitrate index
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var brx = (this.DataBuffer[this.FrameStartIdx + 2] & 0xF0) >>> 4;
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// SampRate index
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var srx = (this.DataBuffer[this.FrameStartIdx + 2] & 0x0C) >>> 2;
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// Resolve flags to real values
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var bitrate = AudioFormatReader_MPEG.MPEG_bitrates[ver][lyr][brx] * 1000;
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var samprate = AudioFormatReader_MPEG.MPEG_srates[ver][srx];
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var samples = AudioFormatReader_MPEG.MPEG_frame_samples[ver][lyr];
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var slot_size = AudioFormatReader_MPEG.MPEG_slot_size[lyr];
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// In-between calculations
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var bps = samples / 8.0;
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var fsize = ((bps * bitrate) / samprate) + ((pad == 1) ? slot_size : 0);
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// Truncate to integer
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var frameSize = Math.floor(fsize);
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// Store number of samples and samplerate for frame
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this.FrameSamples = samples;
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this.FrameSampleRate = samprate;
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// Set end frame boundry
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this.FrameEndIdx = this.FrameStartIdx + frameSize;
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}
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}
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};
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// Checks if there is a frame ready to be extracted
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AudioFormatReader_MPEG.prototype.CanExtractFrame = function () {
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if (this.FrameStartIdx < 0 || this.FrameEndIdx < 0)
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return false;
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else if (this.FrameEndIdx <= this.DataBuffer.length)
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return true;
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else
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return false;
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};
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// Extract a single frame from the buffer
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AudioFormatReader_MPEG.prototype.ExtractFrame = function () {
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// Extract frame data from buffer
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var frameArray = this.DataBuffer.buffer.slice(this.FrameStartIdx, this.FrameEndIdx);
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// Remove frame from buffer
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if ((this.FrameEndIdx + 1) < this.DataBuffer.length)
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this.DataBuffer = new Uint8Array(this.DataBuffer.buffer.slice(this.FrameEndIdx));
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else
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this.DataBuffer = new Uint8Array(0);
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// Reset Start/End indices
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this.FrameStartIdx = 0;
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this.FrameEndIdx = -1;
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return new MPEGFrameInfo(new Uint8Array(frameArray), this.FrameSamples, this.FrameSampleRate);
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};
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// Is called if the decoding of the window succeeded
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AudioFormatReader_MPEG.prototype.OnDecodeSuccess = function (decodedData, id, expectedTotalPlayTime) {
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var extractSampleCount;
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var extractSampleOffset;
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// Check if we got the expected number of samples
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if (expectedTotalPlayTime > decodedData.duration) {
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// We got less samples than expect, we suspect that they were truncated equally at start and end.
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// This can happen in case of sample rate conversions.
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extractSampleCount = decodedData.length;
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extractSampleOffset = 0;
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this.TimeBudget += (expectedTotalPlayTime - decodedData.duration);
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}
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else if (expectedTotalPlayTime < decodedData.duration) {
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// We got more samples than expect, we suspect that zeros were added equally at start and end.
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// This can happen in case of sample rate conversions or edge frame handling.
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extractSampleCount = Math.ceil(expectedTotalPlayTime * decodedData.sampleRate);
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var budgetSamples = this.TimeBudget * decodedData.sampleRate;
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if (budgetSamples > 1.0) {
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if (budgetSamples > decodedData.length - extractSampleCount) {
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budgetSamples = decodedData.length - extractSampleCount;
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}
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extractSampleCount += budgetSamples;
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this.TimeBudget -= (budgetSamples / decodedData.sampleRate);
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}
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extractSampleOffset = Math.floor((decodedData.length - extractSampleCount) / 2);
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}
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else {
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// We got the expected number of samples, no adaption needed
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extractSampleCount = decodedData.length;
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extractSampleOffset = 0;
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}
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// Create a buffer that can hold the frame to extract
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var audioBuffer = this.Audio.createBuffer(decodedData.numberOfChannels, extractSampleCount, decodedData.sampleRate);
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// Fill buffer with the last part of the decoded frame leave out last granule
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for (var i = 0; i < decodedData.numberOfChannels; i++)
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audioBuffer.getChannelData(i).set(decodedData.getChannelData(i).subarray(extractSampleOffset, extractSampleOffset + extractSampleCount));
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this.OnDataReady(id, audioBuffer);
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};
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// Is called in case the decoding of the window fails
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AudioFormatReader_MPEG.prototype.OnDecodeError = function (_error) {
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this.ErrorCallback();
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};
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// MPEG versions - use [version]
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AudioFormatReader_MPEG.MPEG_versions = new Array(25, 0, 2, 1);
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// Layers - use [layer]
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AudioFormatReader_MPEG.MPEG_layers = new Array(0, 3, 2, 1);
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// Bitrates - use [version][layer][bitrate]
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AudioFormatReader_MPEG.MPEG_bitrates = new Array(new Array(// Version 2.5
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new Array(0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0), // Reserved
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new Array(0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, 0), // Layer 3
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new Array(0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, 0), // Layer 2
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new Array(0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256, 0) // Layer 1
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), new Array(// Reserved
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new Array(0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0), // Invalid
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new Array(0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0), // Invalid
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new Array(0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0), // Invalid
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new Array(0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0) // Invalid
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), new Array(// Version 2
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new Array(0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0), // Reserved
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new Array(0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, 0), // Layer 3
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new Array(0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, 0), // Layer 2
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new Array(0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256, 0) // Layer 1
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), new Array(// Version 1
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new Array(0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0), // Reserved
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new Array(0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 0), // Layer 3
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new Array(0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384, 0), // Layer 2
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new Array(0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448, 0) // Layer 1
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));
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// Sample rates - use [version][srate]
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AudioFormatReader_MPEG.MPEG_srates = new Array(new Array(11025, 12000, 8000, 0), // MPEG 2.5
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new Array(0, 0, 0, 0), // Reserved
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new Array(22050, 24000, 16000, 0), // MPEG 2
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new Array(44100, 48000, 32000, 0) // MPEG 1
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);
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// Samples per frame - use [version][layer]
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AudioFormatReader_MPEG.MPEG_frame_samples = new Array(
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// Rsvd 3 2 1 < Layer v Version
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new Array(0, 576, 1152, 384), // 2.5
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new Array(0, 0, 0, 0), // Reserved
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new Array(0, 576, 1152, 384), // 2
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new Array(0, 1152, 1152, 384) // 1
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);
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AudioFormatReader_MPEG.Id3v2Tag = new Uint8Array(new Array(0x49, 0x44, 0x33, // File identifier: "ID3"
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0x03, 0x00, // Version 2.3
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0x00, // Flags: no unsynchronisation, no extended header, no experimental indicator
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0x00, 0x00, 0x00, 0x0D, // Size of the (tag-)frames, extended header and padding
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0x54, 0x49, 0x54, 0x32, // Title frame: "TIT2"
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0x00, 0x00, 0x00, 0x02, // Size of the frame data
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0x00, 0x00, // Frame Flags
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0x00, 0x20, 0x00 // Frame data (space character) and padding
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));
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// Slot size (MPEG unit of measurement) - use [layer]
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AudioFormatReader_MPEG.MPEG_slot_size = new Array(0, 1, 1, 4); // Rsvd, 3, 2, 1
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return AudioFormatReader_MPEG;
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}(AudioFormatReader));
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//# sourceMappingURL=3las.formatreader.mpeg.js.map
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@@ -0,0 +1,223 @@
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/*
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WAV audio format reader is part of 3LAS (Low Latency Live Audio Streaming)
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https://github.com/JoJoBond/3LAS
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*/
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var __extends = (this && this.__extends) || (function () {
|
||||
var extendStatics = function (d, b) {
|
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extendStatics = Object.setPrototypeOf ||
|
||||
({ __proto__: [] } instanceof Array && function (d, b) { d.__proto__ = b; }) ||
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function (d, b) { for (var p in b) if (Object.prototype.hasOwnProperty.call(b, p)) d[p] = b[p]; };
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return extendStatics(d, b);
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||||
};
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return function (d, b) {
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if (typeof b !== "function" && b !== null)
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||||
throw new TypeError("Class extends value " + String(b) + " is not a constructor or null");
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||||
extendStatics(d, b);
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function __() { this.constructor = d; }
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d.prototype = b === null ? Object.create(b) : (__.prototype = b.prototype, new __());
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};
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})();
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var AudioFormatReader_WAV = /** @class */ (function (_super) {
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__extends(AudioFormatReader_WAV, _super);
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function AudioFormatReader_WAV(audio, logger, errorCallback, beforeDecodeCheck, dataReadyCallback, batchDuration, extraEdgeDuration) {
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var _this = _super.call(this, audio, logger, errorCallback, beforeDecodeCheck, dataReadyCallback) || this;
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_this._OnDecodeSuccess = _this.OnDecodeSuccess.bind(_this);
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_this._OnDecodeError = _this.OnDecodeError.bind(_this);
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_this.BatchDuration = batchDuration;
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_this.ExtraEdgeDuration = extraEdgeDuration;
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_this.GotHeader = false;
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_this.RiffHeader = null;
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_this.WaveSampleRate = 0;
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_this.WaveBitsPerSample = 0;
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_this.WaveBytesPerSample = 0;
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_this.WaveBlockAlign = 0;
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_this.WaveChannels = 0;
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_this.BatchSamples = 0;
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_this.BatchBytes = 0;
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_this.ExtraEdgeSamples = 0;
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_this.TotalBatchSampleSize = 0;
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_this.TotalBatchByteSize = 0;
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_this.SampleBudget = 0;
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return _this;
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}
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// Deletes all samples from the databuffer and the samplearray
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AudioFormatReader_WAV.prototype.PurgeData = function () {
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_super.prototype.PurgeData.call(this);
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this.SampleBudget = 0;
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};
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// Deletes all data from the reader (deos effect headers, etc.)
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AudioFormatReader_WAV.prototype.Reset = function () {
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_super.prototype.Reset.call(this);
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this.GotHeader = false;
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this.RiffHeader = null;
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this.WaveSampleRate = 0;
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this.WaveBitsPerSample = 0;
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this.WaveBytesPerSample = 0;
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this.WaveBlockAlign = 0;
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this.WaveChannels = 0;
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this.BatchSamples = 0;
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this.BatchBytes = 0;
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this.ExtraEdgeSamples = 0;
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this.TotalBatchSampleSize = 0;
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this.TotalBatchByteSize = 0;
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this.SampleBudget = 0;
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||||
};
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AudioFormatReader_WAV.prototype.ExtractAll = function () {
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||||
if (!this.GotHeader)
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this.FindAndExtractHeader();
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else {
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||||
var _loop_1 = function () {
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||||
// Extract samples
|
||||
var tmpSamples = this_1.ExtractIntSamples();
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||||
// Increment Id
|
||||
var id = this_1.Id++;
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||||
if (!this_1.OnBeforeDecode(id, this_1.BatchDuration))
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||||
return "continue";
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||||
// Note:
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||||
// =====
|
||||
// When audio data is resampled we get edge-effects at beginnging and end.
|
||||
// We should be able to compensate for that by keeping the last sample of the
|
||||
// previous batch and adding it to the beginning of the current one, but then
|
||||
// cutting it out AFTER the resampling (since the same effects apply to it)
|
||||
// The effects at the end can be compensated by cutting the resampled samples shorter
|
||||
// This is not trivial for non-natural ratios (e.g. 16kHz -> 44.1kHz). Because we would have
|
||||
// to cut out a non-natural number of samples at beginning and end.
|
||||
// TODO: All of the above...
|
||||
// Create a buffer long enough to hold everything
|
||||
var samplesBuffer = new Uint8Array(this_1.RiffHeader.length + tmpSamples.length);
|
||||
var offset = 0;
|
||||
// Add header
|
||||
samplesBuffer.set(this_1.RiffHeader, offset);
|
||||
offset += this_1.RiffHeader.length;
|
||||
// Add samples
|
||||
samplesBuffer.set(tmpSamples, offset);
|
||||
// Push pages to the decoder
|
||||
this_1.Audio.decodeAudioData(samplesBuffer.buffer, (function (decodedData) {
|
||||
var _id = id;
|
||||
this._OnDecodeSuccess(decodedData, _id);
|
||||
}).bind(this_1), this_1._OnDecodeError);
|
||||
};
|
||||
var this_1 = this;
|
||||
while (this.CanExtractSamples()) {
|
||||
_loop_1();
|
||||
}
|
||||
}
|
||||
};
|
||||
// Finds riff header within the data buffer and extracts it
|
||||
AudioFormatReader_WAV.prototype.FindAndExtractHeader = function () {
|
||||
var curpos = 0;
|
||||
// Make sure a whole header can fit
|
||||
if (!((curpos + 4) < this.DataBuffer.length))
|
||||
return;
|
||||
// Check chunkID, should be "RIFF"
|
||||
if (!(this.DataBuffer[curpos] == 0x52 && this.DataBuffer[curpos + 1] == 0x49 && this.DataBuffer[curpos + 2] == 0x46 && this.DataBuffer[curpos + 3] == 0x46))
|
||||
return;
|
||||
curpos += 8;
|
||||
if (!((curpos + 4) < this.DataBuffer.length))
|
||||
return;
|
||||
// Check riffType, should be "WAVE"
|
||||
if (!(this.DataBuffer[curpos] == 0x57 && this.DataBuffer[curpos + 1] == 0x41 && this.DataBuffer[curpos + 2] == 0x56 && this.DataBuffer[curpos + 3] == 0x45))
|
||||
return;
|
||||
curpos += 4;
|
||||
if (!((curpos + 4) < this.DataBuffer.length))
|
||||
return;
|
||||
// Check for format subchunk, should be "fmt "
|
||||
if (!(this.DataBuffer[curpos] == 0x66 && this.DataBuffer[curpos + 1] == 0x6d && this.DataBuffer[curpos + 2] == 0x74 && this.DataBuffer[curpos + 3] == 0x20))
|
||||
return;
|
||||
curpos += 4;
|
||||
if (!((curpos + 4) < this.DataBuffer.length))
|
||||
return;
|
||||
var subChunkSize = this.DataBuffer[curpos] | this.DataBuffer[curpos + 1] << 8 | this.DataBuffer[curpos + 2] << 16 | this.DataBuffer[curpos + 3] << 24;
|
||||
if (!((curpos + 4 + subChunkSize) < this.DataBuffer.length))
|
||||
return;
|
||||
curpos += 6;
|
||||
this.WaveChannels = this.DataBuffer[curpos] | this.DataBuffer[curpos + 1] << 8;
|
||||
curpos += 2;
|
||||
this.WaveSampleRate = this.DataBuffer[curpos] | this.DataBuffer[curpos + 1] << 8 | this.DataBuffer[curpos + 2] << 16 | this.DataBuffer[curpos + 3] << 24;
|
||||
curpos += 8;
|
||||
this.WaveBlockAlign = this.DataBuffer[curpos] | this.DataBuffer[curpos + 1] << 8;
|
||||
curpos += 2;
|
||||
this.WaveBitsPerSample = this.DataBuffer[curpos] | this.DataBuffer[curpos + 1] << 8;
|
||||
this.WaveBytesPerSample = this.WaveBitsPerSample / 8;
|
||||
curpos += subChunkSize - 14;
|
||||
while (true) {
|
||||
if ((curpos + 8) < this.DataBuffer.length) {
|
||||
subChunkSize = this.DataBuffer[curpos + 4] | this.DataBuffer[curpos + 5] << 8 | this.DataBuffer[curpos + 6] << 16 | this.DataBuffer[curpos + 7] << 24;
|
||||
// Check for data subchunk, should be "data"
|
||||
if (this.DataBuffer[curpos] == 0x64 && this.DataBuffer[curpos + 1] == 0x61 && this.DataBuffer[curpos + 2] == 0x74 && this.DataBuffer[curpos + 3] == 0x61) // Data chunk found
|
||||
break;
|
||||
else
|
||||
curpos += 8 + subChunkSize;
|
||||
}
|
||||
else
|
||||
return;
|
||||
}
|
||||
curpos += 8;
|
||||
this.RiffHeader = new Uint8Array(this.DataBuffer.buffer.slice(0, curpos));
|
||||
this.BatchSamples = Math.ceil(this.BatchDuration * this.WaveSampleRate);
|
||||
this.ExtraEdgeSamples = Math.ceil(this.ExtraEdgeDuration * this.WaveSampleRate);
|
||||
this.BatchBytes = this.BatchSamples * this.WaveBlockAlign;
|
||||
this.TotalBatchSampleSize = (this.BatchSamples + this.ExtraEdgeSamples);
|
||||
this.TotalBatchByteSize = this.TotalBatchSampleSize * this.WaveBlockAlign;
|
||||
var chunkSize = this.RiffHeader.length + this.TotalBatchByteSize - 8;
|
||||
// Fix header chunksizes
|
||||
this.RiffHeader[4] = chunkSize & 0xFF;
|
||||
this.RiffHeader[5] = (chunkSize & 0xFF00) >>> 8;
|
||||
this.RiffHeader[6] = (chunkSize & 0xFF0000) >>> 16;
|
||||
this.RiffHeader[7] = (chunkSize & 0xFF000000) >>> 24;
|
||||
this.RiffHeader[this.RiffHeader.length - 4] = (this.TotalBatchByteSize & 0xFF);
|
||||
this.RiffHeader[this.RiffHeader.length - 3] = (this.TotalBatchByteSize & 0xFF00) >>> 8;
|
||||
this.RiffHeader[this.RiffHeader.length - 2] = (this.TotalBatchByteSize & 0xFF0000) >>> 16;
|
||||
this.RiffHeader[this.RiffHeader.length - 1] = (this.TotalBatchByteSize & 0xFF000000) >>> 24;
|
||||
this.GotHeader = true;
|
||||
};
|
||||
// Checks if there is a samples ready to be extracted
|
||||
AudioFormatReader_WAV.prototype.CanExtractSamples = function () {
|
||||
if (this.DataBuffer.length >= this.TotalBatchByteSize)
|
||||
return true;
|
||||
else
|
||||
return false;
|
||||
};
|
||||
// Extract a single batch of samples from the buffer
|
||||
AudioFormatReader_WAV.prototype.ExtractIntSamples = function () {
|
||||
// Extract sample data from buffer
|
||||
var intSampleArray = new Uint8Array(this.DataBuffer.buffer.slice(0, this.TotalBatchByteSize));
|
||||
// Remove samples from buffer
|
||||
this.DataBuffer = new Uint8Array(this.DataBuffer.buffer.slice(this.BatchBytes));
|
||||
return intSampleArray;
|
||||
};
|
||||
// Is called if the decoding of the samples succeeded
|
||||
AudioFormatReader_WAV.prototype.OnDecodeSuccess = function (decodedData, id) {
|
||||
// Calculate the length of the parts
|
||||
var pickSize = this.BatchDuration * decodedData.sampleRate;
|
||||
this.SampleBudget += (pickSize - Math.ceil(pickSize));
|
||||
pickSize = Math.ceil(pickSize);
|
||||
var pickOffset = (decodedData.length - pickSize) / 2.0;
|
||||
if (pickOffset < 0)
|
||||
pickOffset = 0; // This should never happen!
|
||||
else
|
||||
pickOffset = Math.floor(pickOffset);
|
||||
if (this.SampleBudget < -1.0) {
|
||||
var correction = -1.0 * Math.floor(Math.abs(this.SampleBudget));
|
||||
this.SampleBudget -= correction;
|
||||
pickSize += correction;
|
||||
}
|
||||
else if (this.SampleBudget > 1.0) {
|
||||
var correction = Math.floor(this.SampleBudget);
|
||||
this.SampleBudget -= correction;
|
||||
pickSize += correction;
|
||||
}
|
||||
// Create a buffer that can hold a single part
|
||||
var audioBuffer = this.Audio.createBuffer(decodedData.numberOfChannels, pickSize, decodedData.sampleRate);
|
||||
// Fill buffer with the last part of the decoded frame
|
||||
for (var i = 0; i < decodedData.numberOfChannels; i++)
|
||||
audioBuffer.getChannelData(i).set(decodedData.getChannelData(i).slice(pickOffset, -pickOffset));
|
||||
this.OnDataReady(id, audioBuffer);
|
||||
};
|
||||
// Is called in case the decoding of the window fails
|
||||
AudioFormatReader_WAV.prototype.OnDecodeError = function (_error) {
|
||||
this.ErrorCallback();
|
||||
};
|
||||
return AudioFormatReader_WAV;
|
||||
}(AudioFormatReader));
|
||||
//# sourceMappingURL=3las.formatreader.wav.js.map
|
||||
Reference in New Issue
Block a user