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https://github.com/gyurix1968/guacamole-client.git
synced 2025-09-06 13:17:41 +00:00
GUAC-1354: Dynamically split and reassemble audio packets to minimize clicking.
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@@ -180,6 +180,18 @@ Guacamole.RawAudioPlayer = function RawAudioPlayer(stream, mimetype) {
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*/
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*/
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var reader = new Guacamole.ArrayBufferReader(stream);
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var reader = new Guacamole.ArrayBufferReader(stream);
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/**
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* The minimum size of an audio packet split by splitAudioPacket(), in
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* seconds. Audio packets smaller than this will not be split, nor will the
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* split result of a larger packet ever be smaller in size than this
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* minimum.
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*
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* @private
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* @constant
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* @type Number
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*/
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var MIN_SPLIT_SIZE = 0.02;
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/**
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/**
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* The maximum amount of latency to allow between the buffered data stream
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* The maximum amount of latency to allow between the buffered data stream
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* and the playback position, in seconds. Initially, this is set to
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* and the playback position, in seconds. Initially, this is set to
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@@ -190,32 +202,209 @@ Guacamole.RawAudioPlayer = function RawAudioPlayer(stream, mimetype) {
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*/
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*/
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var maxLatency = 0.3;
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var maxLatency = 0.3;
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// Play each received raw packet of audio immediately
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/**
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reader.ondata = function playReceivedAudio(data) {
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* The type of typed array that will be used to represent each audio packet
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* internally. This will be either Int8Array or Int16Array, depending on
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* whether the raw audio format is 8-bit or 16-bit.
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*
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* @private
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* @constructor
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*/
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var SampleArray = (format.bytesPerSample === 1) ? window.Int8Array : window.Int16Array;
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/**
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* The maximum absolute value of any sample within a raw audio packet
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* received by this audio player. This depends only on the size of each
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* sample, and will be 128 for 8-bit audio and 32768 for 16-bit audio.
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*
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* @private
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* @type Number
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*/
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var maxSampleValue = (format.bytesPerSample === 1) ? 128 : 32768;
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/**
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* The queue of all pending audio packets, as an array of sample arrays.
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* Audio packets which are pending playback will be added to this queue for
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* further manipulation prior to scheduling via the Web Audio API. Once an
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* audio packet leaves this queue and is scheduled via the Web Audio API,
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* no further modifications can be made to that packet.
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*
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* @private
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* @type SampleArray[]
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*/
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var packetQueue = [];
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/**
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* Given an array of audio packets, returns a single audio packet
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* containing the concatenation of those packets.
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*
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* @private
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* @param {SampleArray[]} packets
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* The array of audio packets to concatenate.
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*
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* @returns {SampleArray}
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* A single audio packet containing the concatenation of all given
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* audio packets. If no packets are provided, this will be undefined.
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*/
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var joinAudioPackets = function joinAudioPackets(packets) {
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// Do not bother joining if one or fewer packets are in the queue
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if (packets.length <= 1)
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return packets[0];
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// Determine total sample length of the entire queue
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var totalLength = 0;
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packets.forEach(function addPacketLengths(packet) {
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totalLength += packet.length;
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});
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// Append each packet within queue
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var offset = 0;
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var joined = new SampleArray(totalLength);
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packets.forEach(function appendPacket(packet) {
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joined.set(packet, offset);
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offset += packet.length;
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});
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return joined;
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};
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/**
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* Given a single packet of audio data, splits off an arbitrary length of
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* audio data from the beginning of that packet, returning the split result
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* as an array of two packets. The split location is determined through an
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* algorithm intended to minimize the liklihood of audible clicking between
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* packets. If no such split location is possible, an array containing only
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* the originally-provided audio packet is returned.
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*
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* @private
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* @param {SampleArray} data
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* The audio packet to split.
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*
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* @returns {SampleArray[]}
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* An array of audio packets containing the result of splitting the
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* provided audio packet. If splitting is possible, this array will
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* contain two packets. If splitting is not possible, this array will
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* contain only the originally-provided packet.
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*/
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var splitAudioPacket = function splitAudioPacket(data) {
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var minValue = Number.MAX_VALUE;
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var optimalSplitLength = data.length;
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// Calculate number of whole samples in the provided audio packet AND
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// in the minimum possible split packet
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var samples = Math.floor(data.length / format.channels);
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var minSplitSamples = Math.floor(format.rate * MIN_SPLIT_SIZE);
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// Calculate the beginning of the "end" of the audio packet
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var start = Math.max(
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format.channels * minSplitSamples,
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format.channels * (samples - minSplitSamples)
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);
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// For all samples at the end of the given packet, find a point where
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// the perceptible volume across all channels is lowest (and thus is
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// the optimal point to split)
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for (var offset = start; offset < data.length; offset += format.channels) {
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// Calculate the sum of all values across all channels (the result
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// will be proportional to the average volume of a sample)
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var totalValue = 0;
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for (var channel = 0; channel < format.channels; channel++) {
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totalValue += Math.abs(data[offset + channel]);
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}
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// If this is the smallest average value thus far, set the split
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// length such that the first packet ends with the current sample
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if (totalValue <= minValue) {
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optimalSplitLength = offset + format.channels;
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minValue = totalValue;
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}
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}
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// If packet is not split, return the supplied packet untouched
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if (optimalSplitLength === data.length)
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return [data];
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// Otherwise, split the packet into two new packets according to the
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// calculated optimal split length
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return [
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data.slice(0, optimalSplitLength),
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data.slice(optimalSplitLength)
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];
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};
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/**
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* Pushes the given packet of audio data onto the playback queue. Unlike
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* other private functions within Guacamole.RawAudioPlayer, the type of the
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* ArrayBuffer packet of audio data here need not be specific to the type
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* of audio (as with SampleArray). The ArrayBuffer type provided by a
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* Guacamole.ArrayBufferReader, for example, is sufficient. Any necessary
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* conversions will be performed automatically internally.
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*
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* @private
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* @param {ArrayBuffer} data
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* A raw packet of audio data that should be pushed onto the audio
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* playback queue.
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*/
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var pushAudioPacket = function pushAudioPacket(data) {
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packetQueue.push(new SampleArray(data));
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};
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/**
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* Shifts off and returns a packet of audio data from the beginning of the
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* playback queue. The length of this audio packet is determined
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* dynamically according to the click-reduction algorithm implemented by
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* splitAudioPacket().
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*
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* @returns {SampleArray}
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* A packet of audio data pulled from the beginning of the playback
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* queue.
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*/
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var shiftAudioPacket = function shiftAudioPacket() {
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// Flatten data in packet queue
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var data = joinAudioPackets(packetQueue);
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if (!data)
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return null;
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// Pull an appropriate amount of data from the front of the queue
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packetQueue = splitAudioPacket(data);
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data = packetQueue.shift();
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return data;
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};
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/**
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* Converts the given audio packet into an AudioBuffer, ready for playback
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* by the Web Audio API. Unlike the raw audio packets received by this
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* audio player, AudioBuffers require floating point samples and are split
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* into isolated planes of channel-specific data.
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*
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* @private
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* @param {SampleArray} data
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* The raw audio packet that should be converted into a Web Audio API
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* AudioBuffer.
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*
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* @returns {AudioBuffer}
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* A new Web Audio API AudioBuffer containing the provided audio data,
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* converted to the format used by the Web Audio API.
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*/
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var toAudioBuffer = function toAudioBuffer(data) {
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// Calculate total number of samples
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// Calculate total number of samples
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var samples = data.byteLength / format.channels / format.bytesPerSample;
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var samples = data.length / format.channels;
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// Calculate overall duration (in seconds)
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var duration = samples / format.rate;
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// Determine exactly when packet CAN play
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// Determine exactly when packet CAN play
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var packetTime = context.currentTime;
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var packetTime = context.currentTime;
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if (nextPacketTime < packetTime)
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if (nextPacketTime < packetTime)
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nextPacketTime = packetTime;
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nextPacketTime = packetTime;
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// Obtain typed array view based on defined bytes per sample
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var maxValue;
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var source;
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if (format.bytesPerSample === 1) {
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source = new Int8Array(data);
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maxValue = 128;
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}
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else {
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source = new Int16Array(data);
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maxValue = 32768;
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}
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// Get audio buffer for specified format
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// Get audio buffer for specified format
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var audioBuffer = context.createBuffer(format.channels, samples, format.rate);
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var audioBuffer = context.createBuffer(format.channels, samples, format.rate);
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@@ -227,12 +416,33 @@ Guacamole.RawAudioPlayer = function RawAudioPlayer(stream, mimetype) {
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// Fill audio buffer with data for channel
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// Fill audio buffer with data for channel
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var offset = channel;
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var offset = channel;
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for (var i = 0; i < samples; i++) {
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for (var i = 0; i < samples; i++) {
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audioData[i] = source[offset] / maxValue;
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audioData[i] = data[offset] / maxSampleValue;
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offset += format.channels;
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offset += format.channels;
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}
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}
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}
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}
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return audioBuffer;
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};
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// Defer playback of received audio packets slightly
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reader.ondata = function playReceivedAudio(data) {
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// Push received samples onto queue
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pushAudioPacket(new SampleArray(data));
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// Shift off an arbitrary packet of audio data from the queue (this may
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// be different in size from the packet just pushed)
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var packet = shiftAudioPacket();
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if (!packet)
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return;
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// Determine exactly when packet CAN play
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var packetTime = context.currentTime;
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if (nextPacketTime < packetTime)
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nextPacketTime = packetTime;
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// Set up buffer source
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// Set up buffer source
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var source = context.createBufferSource();
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var source = context.createBufferSource();
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source.connect(context.destination);
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source.connect(context.destination);
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@@ -242,11 +452,11 @@ Guacamole.RawAudioPlayer = function RawAudioPlayer(stream, mimetype) {
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source.start = source.noteOn;
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source.start = source.noteOn;
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// Schedule packet
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// Schedule packet
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source.buffer = audioBuffer;
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source.buffer = toAudioBuffer(packet);
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source.start(nextPacketTime);
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source.start(nextPacketTime);
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// Update timeline
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// Update timeline by duration of scheduled packet
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nextPacketTime += duration;
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nextPacketTime += packet.length / format.channels / format.rate;
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};
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};
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