spec-cat 0.1.0 → 0.1.2

package/.output/server/node_modules/@xenova/transformers/src/transformers.js

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+/**
+ * @file Entry point for the Transformers.js library. Only the exports from this file
+ * are available to the end user, and are grouped as follows:
+ * 
+ * 1. [Pipelines](./pipelines)
+ * 2. [Environment variables](./env)
+ * 3. [Models](./models)
+ * 4. [Tokenizers](./tokenizers)
+ * 5. [Processors](./processors)
+ * 
+ * @module transformers
+ */
+
+export * from './pipelines.js';
+export * from './env.js';
+export * from './models.js';
+export * from './tokenizers.js';
+export * from './processors.js';
+export * from './configs.js';
+
+export * from './utils/audio.js';
+export * from './utils/image.js';
+export * from './utils/tensor.js';
+export * from './utils/maths.js';

package/.output/server/node_modules/@xenova/transformers/src/utils/audio.js

@@ -0,0 +1,672 @@
+/**
+ * @file Helper module for audio processing. 
+ * 
+ * These functions and classes are only used internally, 
+ * meaning an end-user shouldn't need to access anything here.
+ * 
+ * @module utils/audio
+ */
+
+import {
+    getFile,
+} from './hub.js';
+import { FFT, max } from './maths.js';
+import {
+    calculateReflectOffset,
+} from './core.js';
+
+
+/**
+ * Helper function to read audio from a path/URL.
+ * @param {string|URL} url The path/URL to load the audio from.
+ * @param {number} sampling_rate The sampling rate to use when decoding the audio.
+ * @returns {Promise<Float32Array>} The decoded audio as a `Float32Array`.
+ */
+export async function read_audio(url, sampling_rate) {
+    if (typeof AudioContext === 'undefined') {
+        // Running in node or an environment without AudioContext
+        throw Error(
+            "Unable to load audio from path/URL since `AudioContext` is not available in your environment. " +
+            "Instead, audio data should be passed directly to the pipeline/processor. " +
+            "For more information and some example code, see https://huggingface.co/docs/transformers.js/guides/node-audio-processing."
+        )
+    }
+
+    const response = await (await getFile(url)).arrayBuffer();
+    const audioCTX = new AudioContext({ sampleRate: sampling_rate });
+    if (typeof sampling_rate === 'undefined') {
+        console.warn(`No sampling rate provided, using default of ${audioCTX.sampleRate}Hz.`)
+    }
+    const decoded = await audioCTX.decodeAudioData(response);
+
+    /** @type {Float32Array} */
+    let audio;
+
+    // We now replicate HuggingFace's `ffmpeg_read` method:
+    if (decoded.numberOfChannels === 2) {
+        // When downmixing a stereo audio file to mono using the -ac 1 option in FFmpeg,
+        // the audio signal is summed across both channels to create a single mono channel.
+        // However, if the audio is at full scale (i.e. the highest possible volume level),
+        // the summing of the two channels can cause the audio signal to clip or distort.
+
+        // To prevent this clipping, FFmpeg applies a scaling factor of 1/sqrt(2) (~ 0.707)
+        // to the audio signal before summing the two channels. This scaling factor ensures
+        // that the combined audio signal will not exceed the maximum possible level, even
+        // if both channels are at full scale.
+
+        // After applying this scaling factor, the audio signal from both channels is summed
+        // to create a single mono channel. It's worth noting that this scaling factor is
+        // only applied when downmixing stereo audio to mono using the -ac 1 option in FFmpeg.
+        // If you're using a different downmixing method, or if you're not downmixing the
+        // audio at all, this scaling factor may not be needed.
+        const SCALING_FACTOR = Math.sqrt(2);
+
+        const left = decoded.getChannelData(0);
+        const right = decoded.getChannelData(1);
+
+        audio = new Float32Array(left.length);
+        for (let i = 0; i < decoded.length; ++i) {
+            audio[i] = SCALING_FACTOR * (left[i] + right[i]) / 2;
+        }
+
+    } else {
+        // If the audio is not stereo, we can just use the first channel:
+        audio = decoded.getChannelData(0);
+    }
+
+    return audio;
+}
+
+/**
+ * Generates a Hanning window of length M.
+ *
+ * @param {number} M The length of the Hanning window to generate.
+ * @returns {Float64Array} The generated Hanning window.
+ */
+export function hanning(M) {
+    if (M < 1) {
+        return new Float64Array();
+    }
+    if (M === 1) {
+        return new Float64Array([1]);
+    }
+    const denom = M - 1;
+    const factor = Math.PI / denom;
+    const cos_vals = new Float64Array(M);
+    for (let i = 0; i < M; ++i) {
+        const n = 2 * i - denom;
+        cos_vals[i] = 0.5 + 0.5 * Math.cos(factor * n);
+    }
+    return cos_vals;
+}
+
+const HERTZ_TO_MEL_MAPPING = {
+    "htk": (/** @type {number} */ freq) => 2595.0 * Math.log10(1.0 + (freq / 700.0)),
+    "kaldi": (/** @type {number} */ freq) => 1127.0 * Math.log(1.0 + (freq / 700.0)),
+    "slaney": (/** @type {number} */ freq, min_log_hertz = 1000.0, min_log_mel = 15.0, logstep = 27.0 / Math.log(6.4)) =>
+        freq >= min_log_hertz
+            ? min_log_mel + Math.log(freq / min_log_hertz) * logstep
+            : 3.0 * freq / 200.0,
+}
+
+/**
+ * @template {Float32Array|Float64Array|number} T 
+ * @param {T} freq 
+ * @param {string} [mel_scale]
+ * @returns {T}
+ */
+function hertz_to_mel(freq, mel_scale = "htk") {
+    const fn = HERTZ_TO_MEL_MAPPING[mel_scale];
+    if (!fn) {
+        throw new Error('mel_scale should be one of "htk", "slaney" or "kaldi".');
+    }
+
+    return typeof freq === 'number' ? fn(freq) : freq.map(x => fn(x));
+}
+
+const MEL_TO_HERTZ_MAPPING = {
+    "htk": (/** @type {number} */ mels) => 700.0 * (10.0 ** (mels / 2595.0) - 1.0),
+    "kaldi": (/** @type {number} */ mels) => 700.0 * (Math.exp(mels / 1127.0) - 1.0),
+    "slaney": (/** @type {number} */ mels, min_log_hertz = 1000.0, min_log_mel = 15.0, logstep = Math.log(6.4) / 27.0) => mels >= min_log_mel
+        ? min_log_hertz * Math.exp(logstep * (mels - min_log_mel))
+        : 200.0 * mels / 3.0,
+}
+
+/**
+ * @template {Float32Array|Float64Array|number} T 
+ * @param {T} mels 
+ * @param {string} [mel_scale]
+ * @returns {T}
+ */
+function mel_to_hertz(mels, mel_scale = "htk") {
+    const fn = MEL_TO_HERTZ_MAPPING[mel_scale];
+    if (!fn) {
+        throw new Error('mel_scale should be one of "htk", "slaney" or "kaldi".');
+    }
+
+    return typeof mels === 'number' ? fn(mels) : mels.map(x => fn(x));
+}
+
+/**
+* Creates a triangular filter bank.
+*
+* Adapted from torchaudio and librosa.
+*
+* @param {Float64Array} fft_freqs Discrete frequencies of the FFT bins in Hz, of shape `(num_frequency_bins,)`.
+* @param {Float64Array} filter_freqs Center frequencies of the triangular filters to create, in Hz, of shape `(num_mel_filters,)`.
+* @returns {number[][]} of shape `(num_frequency_bins, num_mel_filters)`.
+*/
+function _create_triangular_filter_bank(fft_freqs, filter_freqs) {
+    const filter_diff = Float64Array.from(
+        { length: filter_freqs.length - 1 },
+        (_, i) => filter_freqs[i + 1] - filter_freqs[i]
+    );
+
+    const slopes = Array.from({
+        length: fft_freqs.length
+    }, () => new Array(filter_freqs.length));
+
+    for (let j = 0; j < fft_freqs.length; ++j) {
+        const slope = slopes[j];
+        for (let i = 0; i < filter_freqs.length; ++i) {
+            slope[i] = filter_freqs[i] - fft_freqs[j];
+        }
+    }
+
+    const numFreqs = filter_freqs.length - 2;
+    const ret = Array.from({ length: numFreqs }, () => new Array(fft_freqs.length));
+
+    for (let j = 0; j < fft_freqs.length; ++j) { // 201
+        const slope = slopes[j];
+        for (let i = 0; i < numFreqs; ++i) { // 80
+            const down = -slope[i] / filter_diff[i];
+            const up = slope[i + 2] / filter_diff[i + 1];
+            ret[i][j] = Math.max(0, Math.min(down, up));
+        }
+    }
+    return ret;
+}
+
+/**
+ * Return evenly spaced numbers over a specified interval.
+ * @param {number} start The starting value of the sequence.
+ * @param {number} end The end value of the sequence.
+ * @param {number} num Number of samples to generate.
+ * @returns `num` evenly spaced samples, calculated over the interval `[start, stop]`.
+ */
+function linspace(start, end, num) {
+    const step = (end - start) / (num - 1);
+    return Float64Array.from({ length: num }, (_, i) => start + step * i);
+}
+
+/**
+ * Creates a frequency bin conversion matrix used to obtain a mel spectrogram. This is called a *mel filter bank*, and
+ * various implementation exist, which differ in the number of filters, the shape of the filters, the way the filters
+ * are spaced, the bandwidth of the filters, and the manner in which the spectrum is warped. The goal of these
+ * features is to approximate the non-linear human perception of the variation in pitch with respect to the frequency.
+ * @param {number} num_frequency_bins Number of frequencies used to compute the spectrogram (should be the same as in `stft`).
+ * @param {number} num_mel_filters Number of mel filters to generate.
+ * @param {number} min_frequency Lowest frequency of interest in Hz.
+ * @param {number} max_frequency Highest frequency of interest in Hz. This should not exceed `sampling_rate / 2`.
+ * @param {number} sampling_rate Sample rate of the audio waveform.
+ * @param {string} [norm] If `"slaney"`, divide the triangular mel weights by the width of the mel band (area normalization).
+ * @param {string} [mel_scale] The mel frequency scale to use, `"htk"` or `"slaney"`.
+ * @param {boolean} [triangularize_in_mel_space] If this option is enabled, the triangular filter is applied in mel space rather than frequency space.
+ * This should be set to `true` in order to get the same results as `torchaudio` when computing mel filters.
+ * @returns {number[][]} Triangular filter bank matrix, which is a 2D array of shape (`num_frequency_bins`, `num_mel_filters`).
+ * This is a projection matrix to go from a spectrogram to a mel spectrogram.
+ */
+export function mel_filter_bank(
+    num_frequency_bins,
+    num_mel_filters,
+    min_frequency,
+    max_frequency,
+    sampling_rate,
+    norm = null,
+    mel_scale = "htk",
+    triangularize_in_mel_space = false,
+) {
+    if (norm !== null && norm !== "slaney") {
+        throw new Error('norm must be one of null or "slaney"');
+    }
+
+    const mel_min = hertz_to_mel(min_frequency, mel_scale);
+    const mel_max = hertz_to_mel(max_frequency, mel_scale);
+    const mel_freqs = linspace(mel_min, mel_max, num_mel_filters + 2);
+
+    let filter_freqs = mel_to_hertz(mel_freqs, mel_scale);
+    let fft_freqs; // frequencies of FFT bins in Hz
+
+    if (triangularize_in_mel_space) {
+        const fft_bin_width = sampling_rate / (num_frequency_bins * 2);
+        fft_freqs = hertz_to_mel(Float64Array.from({ length: num_frequency_bins }, (_, i) => i * fft_bin_width), mel_scale);
+        filter_freqs = mel_freqs;
+    } else {
+        fft_freqs = linspace(0, Math.floor(sampling_rate / 2), num_frequency_bins);
+    }
+
+    const mel_filters = _create_triangular_filter_bank(fft_freqs, filter_freqs);
+
+    if (norm !== null && norm === "slaney") {
+        // Slaney-style mel is scaled to be approx constant energy per channel
+        for (let i = 0; i < num_mel_filters; ++i) {
+            const filter = mel_filters[i];
+            const enorm = 2.0 / (filter_freqs[i + 2] - filter_freqs[i]);
+            for (let j = 0; j < num_frequency_bins; ++j) {
+                // Apply this enorm to all frequency bins
+                filter[j] *= enorm;
+            }
+        }
+    }
+
+    // TODO warn if there is a zero row
+
+    return mel_filters;
+
+}
+
+/**
+ * @template {Float32Array|Float64Array} T
+ * Pads an array with a reflected version of itself on both ends.
+ * @param {T} array The array to pad.
+ * @param {number} left The amount of padding to add to the left.
+ * @param {number} right The amount of padding to add to the right.
+ * @returns {T} The padded array.
+ */
+function padReflect(array, left, right) {
+    // @ts-ignore
+    const padded = new array.constructor(array.length + left + right);
+    const w = array.length - 1;
+
+    for (let i = 0; i < array.length; ++i) {
+        padded[left + i] = array[i];
+    }
+
+    for (let i = 1; i <= left; ++i) {
+        padded[left - i] = array[calculateReflectOffset(i, w)];
+    }
+
+    for (let i = 1; i <= right; ++i) {
+        padded[w + left + i] = array[calculateReflectOffset(w - i, w)];
+    }
+
+    return padded;
+}
+
+/**
+ * Helper function to compute `amplitude_to_db` and `power_to_db`.
+ * @template {Float32Array|Float64Array} T
+ * @param {T} spectrogram 
+ * @param {number} factor 
+ * @param {number} reference 
+ * @param {number} min_value 
+ * @param {number} db_range 
+ * @returns {T}
+ */
+function _db_conversion_helper(spectrogram, factor, reference, min_value, db_range) {
+    if (reference <= 0) {
+        throw new Error('reference must be greater than zero');
+    }
+
+    if (min_value <= 0) {
+        throw new Error('min_value must be greater than zero');
+    }
+
+    reference = Math.max(min_value, reference);
+
+    const logReference = Math.log10(reference);
+    for (let i = 0; i < spectrogram.length; ++i) {
+        spectrogram[i] = factor * Math.log10(Math.max(min_value, spectrogram[i]) - logReference)
+    }
+
+    if (db_range !== null) {
+        if (db_range <= 0) {
+            throw new Error('db_range must be greater than zero');
+        }
+        const maxValue = max(spectrogram)[0] - db_range;
+        for (let i = 0; i < spectrogram.length; ++i) {
+            spectrogram[i] = Math.max(spectrogram[i], maxValue);
+        }
+    }
+
+    return spectrogram;
+}
+
+/**
+ * Converts an amplitude spectrogram to the decibel scale. This computes `20 * log10(spectrogram / reference)`,
+ * using basic logarithm properties for numerical stability. NOTE: Operates in-place.
+ * 
+ * The motivation behind applying the log function on the (mel) spectrogram is that humans do not hear loudness on a
+ * linear scale. Generally to double the perceived volume of a sound we need to put 8 times as much energy into it.
+ * This means that large variations in energy may not sound all that different if the sound is loud to begin with.
+ * This compression operation makes the (mel) spectrogram features match more closely what humans actually hear.
+ * 
+ * @template {Float32Array|Float64Array} T
+ * @param {T} spectrogram The input amplitude (mel) spectrogram.
+ * @param {number} [reference=1.0] Sets the input spectrogram value that corresponds to 0 dB.
+ * For example, use `np.max(spectrogram)` to set the loudest part to 0 dB. Must be greater than zero.
+ * @param {number} [min_value=1e-5] The spectrogram will be clipped to this minimum value before conversion to decibels,
+ * to avoid taking `log(0)`. The default of `1e-5` corresponds to a minimum of -100 dB. Must be greater than zero.
+ * @param {number} [db_range=null] Sets the maximum dynamic range in decibels. For example, if `db_range = 80`, the
+ * difference between the peak value and the smallest value will never be more than 80 dB. Must be greater than zero.
+ * @returns {T} The modified spectrogram in decibels.
+ */
+function amplitude_to_db(spectrogram, reference = 1.0, min_value = 1e-5, db_range = null) {
+    return _db_conversion_helper(spectrogram, 20.0, reference, min_value, db_range);
+}
+
+/**
+ * Converts a power spectrogram to the decibel scale. This computes `10 * log10(spectrogram / reference)`,
+ * using basic logarithm properties for numerical stability. NOTE: Operates in-place.
+ * 
+ * The motivation behind applying the log function on the (mel) spectrogram is that humans do not hear loudness on a
+ * linear scale. Generally to double the perceived volume of a sound we need to put 8 times as much energy into it.
+ * This means that large variations in energy may not sound all that different if the sound is loud to begin with.
+ * This compression operation makes the (mel) spectrogram features match more closely what humans actually hear.
+ * 
+ * Based on the implementation of `librosa.power_to_db`.
+ * 
+ * @template {Float32Array|Float64Array} T
+ * @param {T} spectrogram The input power (mel) spectrogram. Note that a power spectrogram has the amplitudes squared!
+ * @param {number} [reference=1.0] Sets the input spectrogram value that corresponds to 0 dB.
+ * For example, use `np.max(spectrogram)` to set the loudest part to 0 dB. Must be greater than zero.
+ * @param {number} [min_value=1e-10] The spectrogram will be clipped to this minimum value before conversion to decibels,
+ * to avoid taking `log(0)`. The default of `1e-10` corresponds to a minimum of -100 dB. Must be greater than zero.
+ * @param {number} [db_range=null] Sets the maximum dynamic range in decibels. For example, if `db_range = 80`, the
+ * difference between the peak value and the smallest value will never be more than 80 dB. Must be greater than zero.
+ * @returns {T} The modified spectrogram in decibels.
+ */
+function power_to_db(spectrogram, reference = 1.0, min_value = 1e-10, db_range = null) {
+    return _db_conversion_helper(spectrogram, 10.0, reference, min_value, db_range);
+}
+
+/**
+ * Calculates a spectrogram over one waveform using the Short-Time Fourier Transform.
+ * 
+ * This function can create the following kinds of spectrograms:
+ *   - amplitude spectrogram (`power = 1.0`)
+ *   - power spectrogram (`power = 2.0`)
+ *   - complex-valued spectrogram (`power = None`)
+ *   - log spectrogram (use `log_mel` argument)
+ *   - mel spectrogram (provide `mel_filters`)
+ *   - log-mel spectrogram (provide `mel_filters` and `log_mel`)
+ *
+ * In this implementation, the window is assumed to be zero-padded to have the same size as the analysis frame.
+ * A padded window can be obtained from `window_function()`. The FFT input buffer may be larger than the analysis frame, 
+ * typically the next power of two.
+ * 
+ * @param {Float32Array|Float64Array} waveform The input waveform of shape `(length,)`. This must be a single real-valued, mono waveform.
+ * @param {Float32Array|Float64Array} window The windowing function to apply of shape `(frame_length,)`, including zero-padding if necessary. The actual window length may be
+ * shorter than `frame_length`, but we're assuming the array has already been zero-padded.
+ * @param {number} frame_length The length of the analysis frames in samples (a.k.a., `fft_length`).
+ * @param {number} hop_length The stride between successive analysis frames in samples.
+ * @param {Object} options
+ * @param {number} [options.fft_length=null] The size of the FFT buffer in samples. This determines how many frequency bins the spectrogram will have.
+ * For optimal speed, this should be a power of two. If `null`, uses `frame_length`.
+ * @param {number} [options.power=1.0] If 1.0, returns the amplitude spectrogram. If 2.0, returns the power spectrogram. If `null`, returns complex numbers.
+ * @param {boolean} [options.center=true] Whether to pad the waveform so that frame `t` is centered around time `t * hop_length`. If `false`, frame
+ * `t` will start at time `t * hop_length`.
+ * @param {string} [options.pad_mode="reflect"] Padding mode used when `center` is `true`. Possible values are: `"constant"` (pad with zeros),
+ * `"edge"` (pad with edge values), `"reflect"` (pads with mirrored values).
+ * @param {boolean} [options.onesided=true] If `true`, only computes the positive frequencies and returns a spectrogram containing `fft_length // 2 + 1`
+ * frequency bins. If `false`, also computes the negative frequencies and returns `fft_length` frequency bins.
+ * @param {number} [options.preemphasis=null] Coefficient for a low-pass filter that applies pre-emphasis before the DFT.
+ * @param {number[][]} [options.mel_filters=null] The mel filter bank of shape `(num_freq_bins, num_mel_filters)`.
+ * If supplied, applies this filter bank to create a mel spectrogram.
+ * @param {number} [options.mel_floor=1e-10] Minimum value of mel frequency banks.
+ * @param {string} [options.log_mel=null] How to convert the spectrogram to log scale. Possible options are:
+ * `null` (don't convert), `"log"` (take the natural logarithm) `"log10"` (take the base-10 logarithm), `"dB"` (convert to decibels).
+ * Can only be used when `power` is not `null`.
+ * @param {number} [options.reference=1.0] Sets the input spectrogram value that corresponds to 0 dB. For example, use `max(spectrogram)[0]` to set
+ * the loudest part to 0 dB. Must be greater than zero.
+ * @param {number} [options.min_value=1e-10] The spectrogram will be clipped to this minimum value before conversion to decibels, to avoid taking `log(0)`.
+ * For a power spectrogram, the default of `1e-10` corresponds to a minimum of -100 dB. For an amplitude spectrogram, the value `1e-5` corresponds to -100 dB.
+ * Must be greater than zero.
+ * @param {number} [options.db_range=null] Sets the maximum dynamic range in decibels. For example, if `db_range = 80`, the difference between the
+ * peak value and the smallest value will never be more than 80 dB. Must be greater than zero.
+ * @param {boolean} [options.remove_dc_offset=null] Subtract mean from waveform on each frame, applied before pre-emphasis. This should be set to `true` in
+ * order to get the same results as `torchaudio.compliance.kaldi.fbank` when computing mel filters.
+ * @param {number} [options.max_num_frames=null] If provided, limits the number of frames to compute to this value.
+ * @param {boolean} [options.do_pad=true] If `true`, pads the output spectrogram to have `max_num_frames` frames.
+ * @param {boolean} [options.transpose=false] If `true`, the returned spectrogram will have shape `(num_frames, num_frequency_bins/num_mel_filters)`. If `false`, the returned spectrogram will have shape `(num_frequency_bins/num_mel_filters, num_frames)`.
+ * @returns {{data: Float32Array, dims: number[]}} Spectrogram of shape `(num_frequency_bins, length)` (regular spectrogram) or shape `(num_mel_filters, length)` (mel spectrogram).
+ */
+export function spectrogram(
+    waveform,
+    window,
+    frame_length,
+    hop_length,
+    {
+        fft_length = null,
+        power = 1.0,
+        center = true,
+        pad_mode = "reflect",
+        onesided = true,
+        preemphasis = null,
+        mel_filters = null,
+        mel_floor = 1e-10,
+        log_mel = null,
+        reference = 1.0,
+        min_value = 1e-10,
+        db_range = null,
+        remove_dc_offset = null,
+
+        // Custom parameters for efficiency reasons
+        max_num_frames = null,
+        do_pad = true,
+        transpose = false,
+    } = {}
+) {
+    const window_length = window.length;
+    if (fft_length === null) {
+        fft_length = frame_length;
+    }
+    if (frame_length > fft_length) {
+        throw Error(`frame_length (${frame_length}) may not be larger than fft_length (${fft_length})`)
+    }
+
+    if (window_length !== frame_length) {
+        throw new Error(`Length of the window (${window_length}) must equal frame_length (${frame_length})`);
+    }
+
+    if (hop_length <= 0) {
+        throw new Error("hop_length must be greater than zero");
+    }
+
+    if (power === null && mel_filters !== null) {
+        throw new Error(
+            "You have provided `mel_filters` but `power` is `None`. Mel spectrogram computation is not yet supported for complex-valued spectrogram. " +
+            "Specify `power` to fix this issue."
+        );
+    }
+
+    if (center) {
+        if (pad_mode !== 'reflect') {
+            throw new Error(`pad_mode="${pad_mode}" not implemented yet.`)
+        }
+        const half_window = Math.floor((fft_length - 1) / 2) + 1;
+        waveform = padReflect(waveform, half_window, half_window);
+    }
+
+    // split waveform into frames of frame_length size
+    const num_frames = Math.floor(1 + Math.floor((waveform.length - frame_length) / hop_length))
+
+    const num_frequency_bins = onesided ? Math.floor(fft_length / 2) + 1 : fft_length
+
+    let d1 = num_frames;
+    let d1Max = num_frames;
+
+    // If maximum number of frames is provided, we must either pad or truncate
+    if (max_num_frames !== null) {
+        if (max_num_frames > num_frames) { // input is too short, so we pad
+            if (do_pad) {
+                d1Max = max_num_frames;
+            }
+        } else { // input is too long, so we truncate
+            d1Max = d1 = max_num_frames;
+        }
+    }
+
+    // Preallocate arrays to store output.
+    const fft = new FFT(fft_length);
+    const inputBuffer = new Float64Array(fft_length);
+    const outputBuffer = new Float64Array(fft.outputBufferSize);
+    const magnitudes = new Array(d1);
+
+    for (let i = 0; i < d1; ++i) {
+        // Populate buffer with waveform data
+        const offset = i * hop_length;
+        for (let j = 0; j < frame_length; ++j) {
+            inputBuffer[j] = waveform[offset + j];
+        }
+
+        if (remove_dc_offset) {
+            let sum = 0;
+            for (let j = 0; j < frame_length; ++j) {
+                sum += inputBuffer[j];
+            }
+            const mean = sum / frame_length;
+            for (let j = 0; j < frame_length; ++j) {
+                inputBuffer[j] -= mean;
+            }
+        }
+
+        if (preemphasis !== null) {
+            // Done in reverse to avoid copies and distructive modification
+            for (let j = frame_length - 1; j >= 1; --j) {
+                inputBuffer[j] -= preemphasis * inputBuffer[j - 1];
+            }
+            inputBuffer[0] *= 1 - preemphasis;
+        }
+
+        for (let j = 0; j < window.length; ++j) {
+            inputBuffer[j] *= window[j];
+        }
+
+        fft.realTransform(outputBuffer, inputBuffer);
+
+        // compute magnitudes
+        const row = new Array(num_frequency_bins);
+        for (let j = 0; j < row.length; ++j) {
+            const j2 = j << 1;
+            row[j] = outputBuffer[j2] ** 2 + outputBuffer[j2 + 1] ** 2;
+        }
+        magnitudes[i] = row;
+    }
+
+    if (power !== null && power !== 2) {
+        // slight optimization to not sqrt
+        const pow = 2 / power; // we use 2 since we already squared
+        for (let i = 0; i < magnitudes.length; ++i) {
+            const magnitude = magnitudes[i];
+            for (let j = 0; j < magnitude.length; ++j) {
+                magnitude[j] **= pow;
+            }
+        }
+    }
+
+    // TODO: What if `mel_filters` is null?
+    const num_mel_filters = mel_filters.length;
+
+    // Only here do we create Float32Array
+    const mel_spec = new Float32Array(num_mel_filters * d1Max);
+
+    // Perform matrix muliplication:
+    // mel_spec = mel_filters @ magnitudes.T
+    //  - mel_filters.shape=(80, 201)
+    //  - magnitudes.shape=(3000, 201) => - magnitudes.T.shape=(201, 3000)
+    //  - mel_spec.shape=(80, 3000)
+    const dims = transpose ? [d1Max, num_mel_filters] : [num_mel_filters, d1Max];
+    for (let i = 0; i < num_mel_filters; ++i) { // num melfilters (e.g., 80)
+        const filter = mel_filters[i];
+        for (let j = 0; j < d1; ++j) { // num frames (e.g., 3000)
+            const magnitude = magnitudes[j];
+
+            let sum = 0;
+            for (let k = 0; k < num_frequency_bins; ++k) { // num frequency bins (e.g., 201)
+                sum += filter[k] * magnitude[k];
+            }
+
+            mel_spec[
+                transpose
+                    ? j * num_mel_filters + i
+                    : i * d1 + j
+            ] = Math.max(mel_floor, sum);
+        }
+    }
+
+    if (power !== null && log_mel !== null) {
+        const o = Math.min(mel_spec.length, d1 * num_mel_filters);
+        switch (log_mel) {
+            case 'log':
+                for (let i = 0; i < o; ++i) {
+                    mel_spec[i] = Math.log(mel_spec[i]);
+                }
+                break;
+            case 'log10':
+                for (let i = 0; i < o; ++i) {
+                    mel_spec[i] = Math.log10(mel_spec[i]);
+                }
+                break;
+            case 'dB':
+                if (power === 1.0) {
+                    // NOTE: operates in-place
+                    amplitude_to_db(mel_spec, reference, min_value, db_range);
+                } else if (power === 2.0) {
+                    power_to_db(mel_spec, reference, min_value, db_range);
+                } else {
+                    throw new Error(`Cannot use log_mel option '${log_mel}' with power ${power}`)
+                }
+                break;
+            default:
+                throw new Error(`log_mel must be one of null, 'log', 'log10' or 'dB'. Got '${log_mel}'`);
+        }
+    }
+
+    return { data: mel_spec, dims };
+}
+
+/**
+ * Returns an array containing the specified window.
+ * @param {number} window_length The length of the window in samples.
+ * @param {string} name The name of the window function.
+ * @param {Object} options Additional options.
+ * @param {boolean} [options.periodic=true] Whether the window is periodic or symmetric.
+ * @param {number} [options.frame_length=null] The length of the analysis frames in samples.
+ * Provide a value for `frame_length` if the window is smaller than the frame length, so that it will be zero-padded.
+ * @param {boolean} [options.center=true] Whether to center the window inside the FFT buffer. Only used when `frame_length` is provided.
+ * @returns {Float64Array} The window of shape `(window_length,)` or `(frame_length,)`.
+ */
+export function window_function(window_length, name, {
+    periodic = true,
+    frame_length = null,
+    center = true,
+} = {}) {
+    const length = periodic ? window_length + 1 : window_length;
+    let window;
+    switch (name) {
+        case 'boxcar':
+            window = new Float64Array(length).fill(1.0);
+            break;
+        case 'hann':
+        case 'hann_window':
+            window = hanning(length);
+            break;
+        case 'povey':
+            window = hanning(length).map(x => Math.pow(x, 0.85));
+            break;
+        default:
+            throw new Error(`Unknown window type ${name}.`);
+    }
+    if (periodic) {
+        window = window.subarray(0, window_length);
+    }
+    if (frame_length === null) {
+        return window;
+    }
+    if (window_length > frame_length) {
+        throw new Error(`Length of the window (${window_length}) may not be larger than frame_length (${frame_length})`);
+    }
+
+    return window;
+}