GuacLite
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*
* @param {Number} a
* The window size to use for the Lanczos kernel.
*
* @returns {number}
* The value of the Lanczos kernel at the given point for the given
* window size.
*/
var lanczos = function lanczos(x, a) {
// Lanczos is sinc(x) * sinc(x / a) for -a < x < a ...
if (-a < x && x < a)
return sinc(x) * sinc(x / a);
// ... and 0 otherwise
return 0;
};
/**
* Determines the value of the waveform represented by the audio data at
* the given location. If the value cannot be determined exactly as it does
* not correspond to an exact sample within the audio data, the value will
* be derived through interpolating nearby samples.
*
* @private
* @param {Float32Array} audioData
* An array of audio data, as returned by AudioBuffer.getChannelData().
*
* @param {Number} t
* The relative location within the waveform from which the value
* should be retrieved, represented as a floating point number between
* 0 and 1 inclusive, where 0 represents the earliest point in time and
* 1 represents the latest.
*
* @returns {number}
* The value of the waveform at the given location.
*/
var interpolateSample = function getValueAt(audioData, t) {
// Convert [0, 1] range to [0, audioData.length - 1]
var index = (audioData.length - 1) * t;
// Determine the start and end points for the summation used by the
// Lanczos interpolation algorithm (see: https://en.wikipedia.org/wiki/Lanczos_resampling)
var start = Math.floor(index) - LANCZOS_WINDOW_SIZE + 1;
var end = Math.floor(index) + LANCZOS_WINDOW_SIZE;
// Calculate the value of the Lanczos interpolation function for the
// required range
var sum = 0;
for (var i = start; i <= end; i++) {
sum += (audioData[i] || 0) * lanczos(index - i, LANCZOS_WINDOW_SIZE);
}
return sum;
};
/**
* Converts the given AudioBuffer into an audio packet, ready for streaming
* along the underlying output stream. Unlike the raw audio packets used by
* this audio recorder, AudioBuffers require floating point samples and are
* split into isolated planes of channel-specific data.
*
* @private
* @param {AudioBuffer} audioBuffer
* The Web Audio API AudioBuffer that should be converted to a raw
* audio packet.
*
* @returns {!SampleArray}
* A new raw audio packet containing the audio data from the provided
* AudioBuffer.
*/
var toSampleArray = function toSampleArray(audioBuffer) {
// Track overall amount of data read
var inSamples = audioBuffer.length;
readSamples += inSamples;
// Calculate the total number of samples that should be written as of
// the audio data just received and adjust the size of the output
// packet accordingly
var expectedWrittenSamples = Math.round(readSamples * format.rate / audioBuffer.sampleRate);
var outSamples = expectedWrittenSamples - writtenSamples;
// Update number of samples written
writtenSamples += outSamples;
// Get array for raw PCM storage
var data = new SampleArray(outSamples * format.channels);
// Convert each channel
for (var channel = 0; channel < format.channels; channel++) {
var audioData = audioBuffer.getChannelData(channel);
// Fill array with data from audio buffer channel
var offset = channel;
for (var i = 0; i < outSamples; i++) {
data[offset] = interpolateSample(audioData, i / (outSamples - 1)) * maxSampleValue;
offset += format.channels;
}
}
return data;
};
/**
* getUserMedia() callback which handles successful retrieval of an
* audio stream (successful start of recording).
*
* @private
* @param {MediaStream} stream
* A MediaStream which provides access to audio data read from the
* user's local audio input device.
*/
var streamReceived = function streamReceived(stream) {
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