| /* |
| * Licensed to the Apache Software Foundation (ASF) under one |
| * or more contributor license agreements. See the NOTICE file |
| * distributed with this work for additional information |
| * regarding copyright ownership. The ASF licenses this file |
| * to you under the Apache License, Version 2.0 (the |
| * "License"); you may not use this file except in compliance |
| * with the License. You may obtain a copy of the License at |
| * |
| * http://www.apache.org/licenses/LICENSE-2.0 |
| * |
| * Unless required by applicable law or agreed to in writing, |
| * software distributed under the License is distributed on an |
| * "AS IS" BASIS, WITHOUT WARRANTIES OR CONDITIONS OF ANY |
| * KIND, either express or implied. See the License for the |
| * specific language governing permissions and limitations |
| * under the License. |
| */ |
| |
| var Guacamole = Guacamole || {}; |
| |
| /** |
| * Abstract audio player which accepts, queues and plays back arbitrary audio |
| * data. It is up to implementations of this class to provide some means of |
| * handling a provided Guacamole.InputStream. Data received along the provided |
| * stream is to be played back immediately. |
| * |
| * @constructor |
| */ |
| Guacamole.AudioPlayer = function AudioPlayer() { |
| |
| /** |
| * Notifies this Guacamole.AudioPlayer that all audio up to the current |
| * point in time has been given via the underlying stream, and that any |
| * difference in time between queued audio data and the current time can be |
| * considered latency. |
| */ |
| this.sync = function sync() { |
| // Default implementation - do nothing |
| }; |
| |
| }; |
| |
| /** |
| * Determines whether the given mimetype is supported by any built-in |
| * implementation of Guacamole.AudioPlayer, and thus will be properly handled |
| * by Guacamole.AudioPlayer.getInstance(). |
| * |
| * @param {String} mimetype |
| * The mimetype to check. |
| * |
| * @returns {Boolean} |
| * true if the given mimetype is supported by any built-in |
| * Guacamole.AudioPlayer, false otherwise. |
| */ |
| Guacamole.AudioPlayer.isSupportedType = function isSupportedType(mimetype) { |
| |
| return Guacamole.RawAudioPlayer.isSupportedType(mimetype); |
| |
| }; |
| |
| /** |
| * Returns a list of all mimetypes supported by any built-in |
| * Guacamole.AudioPlayer, in rough order of priority. Beware that only the core |
| * mimetypes themselves will be listed. Any mimetype parameters, even required |
| * ones, will not be included in the list. For example, "audio/L8" is a |
| * supported raw audio mimetype that is supported, but it is invalid without |
| * additional parameters. Something like "audio/L8;rate=44100" would be valid, |
| * however (see https://tools.ietf.org/html/rfc4856). |
| * |
| * @returns {String[]} |
| * A list of all mimetypes supported by any built-in Guacamole.AudioPlayer, |
| * excluding any parameters. |
| */ |
| Guacamole.AudioPlayer.getSupportedTypes = function getSupportedTypes() { |
| |
| return Guacamole.RawAudioPlayer.getSupportedTypes(); |
| |
| }; |
| |
| /** |
| * Returns an instance of Guacamole.AudioPlayer providing support for the given |
| * audio format. If support for the given audio format is not available, null |
| * is returned. |
| * |
| * @param {Guacamole.InputStream} stream |
| * The Guacamole.InputStream to read audio data from. |
| * |
| * @param {String} mimetype |
| * The mimetype of the audio data in the provided stream. |
| * |
| * @return {Guacamole.AudioPlayer} |
| * A Guacamole.AudioPlayer instance supporting the given mimetype and |
| * reading from the given stream, or null if support for the given mimetype |
| * is absent. |
| */ |
| Guacamole.AudioPlayer.getInstance = function getInstance(stream, mimetype) { |
| |
| // Use raw audio player if possible |
| if (Guacamole.RawAudioPlayer.isSupportedType(mimetype)) |
| return new Guacamole.RawAudioPlayer(stream, mimetype); |
| |
| // No support for given mimetype |
| return null; |
| |
| }; |
| |
| /** |
| * Implementation of Guacamole.AudioPlayer providing support for raw PCM format |
| * audio. This player relies only on the Web Audio API and does not require any |
| * browser-level support for its audio formats. |
| * |
| * @constructor |
| * @augments Guacamole.AudioPlayer |
| * @param {Guacamole.InputStream} stream |
| * The Guacamole.InputStream to read audio data from. |
| * |
| * @param {String} mimetype |
| * The mimetype of the audio data in the provided stream, which must be a |
| * "audio/L8" or "audio/L16" mimetype with necessary parameters, such as: |
| * "audio/L16;rate=44100,channels=2". |
| */ |
| Guacamole.RawAudioPlayer = function RawAudioPlayer(stream, mimetype) { |
| |
| /** |
| * The format of audio this player will decode. |
| * |
| * @private |
| * @type {Guacamole.RawAudioFormat} |
| */ |
| var format = Guacamole.RawAudioFormat.parse(mimetype); |
| |
| /** |
| * An instance of a Web Audio API AudioContext object, or null if the |
| * Web Audio API is not supported. |
| * |
| * @private |
| * @type {AudioContext} |
| */ |
| var context = Guacamole.AudioContextFactory.getAudioContext(); |
| |
| /** |
| * The earliest possible time that the next packet could play without |
| * overlapping an already-playing packet, in seconds. Note that while this |
| * value is in seconds, it is not an integer value and has microsecond |
| * resolution. |
| * |
| * @private |
| * @type {Number} |
| */ |
| var nextPacketTime = context.currentTime; |
| |
| /** |
| * Guacamole.ArrayBufferReader wrapped around the audio input stream |
| * provided with this Guacamole.RawAudioPlayer was created. |
| * |
| * @private |
| * @type {Guacamole.ArrayBufferReader} |
| */ |
| var reader = new Guacamole.ArrayBufferReader(stream); |
| |
| /** |
| * The minimum size of an audio packet split by splitAudioPacket(), in |
| * seconds. Audio packets smaller than this will not be split, nor will the |
| * split result of a larger packet ever be smaller in size than this |
| * minimum. |
| * |
| * @private |
| * @constant |
| * @type {Number} |
| */ |
| var MIN_SPLIT_SIZE = 0.02; |
| |
| /** |
| * The maximum amount of latency to allow between the buffered data stream |
| * and the playback position, in seconds. Initially, this is set to |
| * roughly one third of a second. |
| * |
| * @private |
| * @type {Number} |
| */ |
| var maxLatency = 0.3; |
| |
| /** |
| * The type of typed array that will be used to represent each audio packet |
| * internally. This will be either Int8Array or Int16Array, depending on |
| * whether the raw audio format is 8-bit or 16-bit. |
| * |
| * @private |
| * @constructor |
| */ |
| var SampleArray = (format.bytesPerSample === 1) ? window.Int8Array : window.Int16Array; |
| |
| /** |
| * The maximum absolute value of any sample within a raw audio packet |
| * received by this audio player. This depends only on the size of each |
| * sample, and will be 128 for 8-bit audio and 32768 for 16-bit audio. |
| * |
| * @private |
| * @type {Number} |
| */ |
| var maxSampleValue = (format.bytesPerSample === 1) ? 128 : 32768; |
| |
| /** |
| * The queue of all pending audio packets, as an array of sample arrays. |
| * Audio packets which are pending playback will be added to this queue for |
| * further manipulation prior to scheduling via the Web Audio API. Once an |
| * audio packet leaves this queue and is scheduled via the Web Audio API, |
| * no further modifications can be made to that packet. |
| * |
| * @private |
| * @type {SampleArray[]} |
| */ |
| var packetQueue = []; |
| |
| /** |
| * Given an array of audio packets, returns a single audio packet |
| * containing the concatenation of those packets. |
| * |
| * @private |
| * @param {SampleArray[]} packets |
| * The array of audio packets to concatenate. |
| * |
| * @returns {SampleArray} |
| * A single audio packet containing the concatenation of all given |
| * audio packets. If no packets are provided, this will be undefined. |
| */ |
| var joinAudioPackets = function joinAudioPackets(packets) { |
| |
| // Do not bother joining if one or fewer packets are in the queue |
| if (packets.length <= 1) |
| return packets[0]; |
| |
| // Determine total sample length of the entire queue |
| var totalLength = 0; |
| packets.forEach(function addPacketLengths(packet) { |
| totalLength += packet.length; |
| }); |
| |
| // Append each packet within queue |
| var offset = 0; |
| var joined = new SampleArray(totalLength); |
| packets.forEach(function appendPacket(packet) { |
| joined.set(packet, offset); |
| offset += packet.length; |
| }); |
| |
| return joined; |
| |
| }; |
| |
| /** |
| * Given a single packet of audio data, splits off an arbitrary length of |
| * audio data from the beginning of that packet, returning the split result |
| * as an array of two packets. The split location is determined through an |
| * algorithm intended to minimize the liklihood of audible clicking between |
| * packets. If no such split location is possible, an array containing only |
| * the originally-provided audio packet is returned. |
| * |
| * @private |
| * @param {SampleArray} data |
| * The audio packet to split. |
| * |
| * @returns {SampleArray[]} |
| * An array of audio packets containing the result of splitting the |
| * provided audio packet. If splitting is possible, this array will |
| * contain two packets. If splitting is not possible, this array will |
| * contain only the originally-provided packet. |
| */ |
| var splitAudioPacket = function splitAudioPacket(data) { |
| |
| var minValue = Number.MAX_VALUE; |
| var optimalSplitLength = data.length; |
| |
| // Calculate number of whole samples in the provided audio packet AND |
| // in the minimum possible split packet |
| var samples = Math.floor(data.length / format.channels); |
| var minSplitSamples = Math.floor(format.rate * MIN_SPLIT_SIZE); |
| |
| // Calculate the beginning of the "end" of the audio packet |
| var start = Math.max( |
| format.channels * minSplitSamples, |
| format.channels * (samples - minSplitSamples) |
| ); |
| |
| // For all samples at the end of the given packet, find a point where |
| // the perceptible volume across all channels is lowest (and thus is |
| // the optimal point to split) |
| for (var offset = start; offset < data.length; offset += format.channels) { |
| |
| // Calculate the sum of all values across all channels (the result |
| // will be proportional to the average volume of a sample) |
| var totalValue = 0; |
| for (var channel = 0; channel < format.channels; channel++) { |
| totalValue += Math.abs(data[offset + channel]); |
| } |
| |
| // If this is the smallest average value thus far, set the split |
| // length such that the first packet ends with the current sample |
| if (totalValue <= minValue) { |
| optimalSplitLength = offset + format.channels; |
| minValue = totalValue; |
| } |
| |
| } |
| |
| // If packet is not split, return the supplied packet untouched |
| if (optimalSplitLength === data.length) |
| return [data]; |
| |
| // Otherwise, split the packet into two new packets according to the |
| // calculated optimal split length |
| return [ |
| new SampleArray(data.buffer.slice(0, optimalSplitLength * format.bytesPerSample)), |
| new SampleArray(data.buffer.slice(optimalSplitLength * format.bytesPerSample)) |
| ]; |
| |
| }; |
| |
| /** |
| * Pushes the given packet of audio data onto the playback queue. Unlike |
| * other private functions within Guacamole.RawAudioPlayer, the type of the |
| * ArrayBuffer packet of audio data here need not be specific to the type |
| * of audio (as with SampleArray). The ArrayBuffer type provided by a |
| * Guacamole.ArrayBufferReader, for example, is sufficient. Any necessary |
| * conversions will be performed automatically internally. |
| * |
| * @private |
| * @param {ArrayBuffer} data |
| * A raw packet of audio data that should be pushed onto the audio |
| * playback queue. |
| */ |
| var pushAudioPacket = function pushAudioPacket(data) { |
| packetQueue.push(new SampleArray(data)); |
| }; |
| |
| /** |
| * Shifts off and returns a packet of audio data from the beginning of the |
| * playback queue. The length of this audio packet is determined |
| * dynamically according to the click-reduction algorithm implemented by |
| * splitAudioPacket(). |
| * |
| * @private |
| * @returns {SampleArray} |
| * A packet of audio data pulled from the beginning of the playback |
| * queue. |
| */ |
| var shiftAudioPacket = function shiftAudioPacket() { |
| |
| // Flatten data in packet queue |
| var data = joinAudioPackets(packetQueue); |
| if (!data) |
| return null; |
| |
| // Pull an appropriate amount of data from the front of the queue |
| packetQueue = splitAudioPacket(data); |
| data = packetQueue.shift(); |
| |
| return data; |
| |
| }; |
| |
| /** |
| * Converts the given audio packet into an AudioBuffer, ready for playback |
| * by the Web Audio API. Unlike the raw audio packets received by this |
| * audio player, AudioBuffers require floating point samples and are split |
| * into isolated planes of channel-specific data. |
| * |
| * @private |
| * @param {SampleArray} data |
| * The raw audio packet that should be converted into a Web Audio API |
| * AudioBuffer. |
| * |
| * @returns {AudioBuffer} |
| * A new Web Audio API AudioBuffer containing the provided audio data, |
| * converted to the format used by the Web Audio API. |
| */ |
| var toAudioBuffer = function toAudioBuffer(data) { |
| |
| // Calculate total number of samples |
| var samples = data.length / format.channels; |
| |
| // Determine exactly when packet CAN play |
| var packetTime = context.currentTime; |
| if (nextPacketTime < packetTime) |
| nextPacketTime = packetTime; |
| |
| // Get audio buffer for specified format |
| var audioBuffer = context.createBuffer(format.channels, samples, format.rate); |
| |
| // Convert each channel |
| for (var channel = 0; channel < format.channels; channel++) { |
| |
| var audioData = audioBuffer.getChannelData(channel); |
| |
| // Fill audio buffer with data for channel |
| var offset = channel; |
| for (var i = 0; i < samples; i++) { |
| audioData[i] = data[offset] / maxSampleValue; |
| offset += format.channels; |
| } |
| |
| } |
| |
| return audioBuffer; |
| |
| }; |
| |
| // Defer playback of received audio packets slightly |
| reader.ondata = function playReceivedAudio(data) { |
| |
| // Push received samples onto queue |
| pushAudioPacket(new SampleArray(data)); |
| |
| // Shift off an arbitrary packet of audio data from the queue (this may |
| // be different in size from the packet just pushed) |
| var packet = shiftAudioPacket(); |
| if (!packet) |
| return; |
| |
| // Determine exactly when packet CAN play |
| var packetTime = context.currentTime; |
| if (nextPacketTime < packetTime) |
| nextPacketTime = packetTime; |
| |
| // Set up buffer source |
| var source = context.createBufferSource(); |
| source.connect(context.destination); |
| |
| // Use noteOn() instead of start() if necessary |
| if (!source.start) |
| source.start = source.noteOn; |
| |
| // Schedule packet |
| source.buffer = toAudioBuffer(packet); |
| source.start(nextPacketTime); |
| |
| // Update timeline by duration of scheduled packet |
| nextPacketTime += packet.length / format.channels / format.rate; |
| |
| }; |
| |
| /** @override */ |
| this.sync = function sync() { |
| |
| // Calculate elapsed time since last sync |
| var now = context.currentTime; |
| |
| // Reschedule future playback time such that playback latency is |
| // bounded within a reasonable latency threshold |
| nextPacketTime = Math.min(nextPacketTime, now + maxLatency); |
| |
| }; |
| |
| }; |
| |
| Guacamole.RawAudioPlayer.prototype = new Guacamole.AudioPlayer(); |
| |
| /** |
| * Determines whether the given mimetype is supported by |
| * Guacamole.RawAudioPlayer. |
| * |
| * @param {String} mimetype |
| * The mimetype to check. |
| * |
| * @returns {Boolean} |
| * true if the given mimetype is supported by Guacamole.RawAudioPlayer, |
| * false otherwise. |
| */ |
| Guacamole.RawAudioPlayer.isSupportedType = function isSupportedType(mimetype) { |
| |
| // No supported types if no Web Audio API |
| if (!Guacamole.AudioContextFactory.getAudioContext()) |
| return false; |
| |
| return Guacamole.RawAudioFormat.parse(mimetype) !== null; |
| |
| }; |
| |
| /** |
| * Returns a list of all mimetypes supported by Guacamole.RawAudioPlayer. Only |
| * the core mimetypes themselves will be listed. Any mimetype parameters, even |
| * required ones, will not be included in the list. For example, "audio/L8" is |
| * a raw audio mimetype that may be supported, but it is invalid without |
| * additional parameters. Something like "audio/L8;rate=44100" would be valid, |
| * however (see https://tools.ietf.org/html/rfc4856). |
| * |
| * @returns {String[]} |
| * A list of all mimetypes supported by Guacamole.RawAudioPlayer, excluding |
| * any parameters. If the necessary JavaScript APIs for playing raw audio |
| * are absent, this list will be empty. |
| */ |
| Guacamole.RawAudioPlayer.getSupportedTypes = function getSupportedTypes() { |
| |
| // No supported types if no Web Audio API |
| if (!Guacamole.AudioContextFactory.getAudioContext()) |
| return []; |
| |
| // We support 8-bit and 16-bit raw PCM |
| return [ |
| 'audio/L8', |
| 'audio/L16' |
| ]; |
| |
| }; |