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torchaudio.functional

Functions to perform common audio operations.

Utility

amplitude_to_DB

torchaudio.functional.amplitude_to_DB(x: torch.Tensor, multiplier: float, amin: float, db_multiplier: float, top_db: Optional[float] = None)torch.Tensor[source]

Turn a spectrogram from the power/amplitude scale to the decibel scale.

The output of each tensor in a batch depends on the maximum value of that tensor, and so may return different values for an audio clip split into snippets vs. a full clip.

Parameters
  • x (Tensor) – Input spectrogram(s) before being converted to decibel scale. Input should take the form (…, freq, time). Batched inputs should include a channel dimension and have the form (batch, channel, freq, time).

  • multiplier (float) – Use 10. for power and 20. for amplitude

  • amin (float) – Number to clamp x

  • db_multiplier (float) – Log10(max(reference value and amin))

  • top_db (float or None, optional) – Minimum negative cut-off in decibels. A reasonable number is 80. (Default: None)

Returns

Output tensor in decibel scale

Return type

Tensor

DB_to_amplitude

torchaudio.functional.DB_to_amplitude(x: torch.Tensor, ref: float, power: float)torch.Tensor[source]

Turn a tensor from the decibel scale to the power/amplitude scale.

Parameters
  • x (Tensor) – Input tensor before being converted to power/amplitude scale.

  • ref (float) – Reference which the output will be scaled by.

  • power (float) – If power equals 1, will compute DB to power. If 0.5, will compute DB to amplitude.

Returns

Output tensor in power/amplitude scale.

Return type

Tensor

melscale_fbanks

torchaudio.functional.melscale_fbanks(n_freqs: int, f_min: float, f_max: float, n_mels: int, sample_rate: int, norm: Optional[str] = None, mel_scale: str = 'htk')torch.Tensor[source]

Create a frequency bin conversion matrix.

Note

For the sake of the numerical compatibility with librosa, not all the coefficients in the resulting filter bank has magnitude of 1.

Visualization of generated filter bank
Parameters
  • n_freqs (int) – Number of frequencies to highlight/apply

  • f_min (float) – Minimum frequency (Hz)

  • f_max (float) – Maximum frequency (Hz)

  • n_mels (int) – Number of mel filterbanks

  • sample_rate (int) – Sample rate of the audio waveform

  • norm (str or None, optional) – If ‘slaney’, divide the triangular mel weights by the width of the mel band (area normalization). (Default: None)

  • mel_scale (str, optional) – Scale to use: htk or slaney. (Default: htk)

Returns

Triangular filter banks (fb matrix) of size (n_freqs, n_mels) meaning number of frequencies to highlight/apply to x the number of filterbanks. Each column is a filterbank so that assuming there is a matrix A of size (…, n_freqs), the applied result would be A * melscale_fbanks(A.size(-1), ...).

Return type

Tensor

Tutorials using melscale_fbanks:

linear_fbanks

torchaudio.functional.linear_fbanks(n_freqs: int, f_min: float, f_max: float, n_filter: int, sample_rate: int)torch.Tensor[source]

Creates a linear triangular filterbank.

Note

For the sake of the numerical compatibility with librosa, not all the coefficients in the resulting filter bank has magnitude of 1.

Visualization of generated filter bank
Parameters
  • n_freqs (int) – Number of frequencies to highlight/apply

  • f_min (float) – Minimum frequency (Hz)

  • f_max (float) – Maximum frequency (Hz)

  • n_filter (int) – Number of (linear) triangular filter

  • sample_rate (int) – Sample rate of the audio waveform

Returns

Triangular filter banks (fb matrix) of size (n_freqs, n_filter) meaning number of frequencies to highlight/apply to x the number of filterbanks. Each column is a filterbank so that assuming there is a matrix A of size (…, n_freqs), the applied result would be A * linear_fbanks(A.size(-1), ...).

Return type

Tensor

create_dct

torchaudio.functional.create_dct(n_mfcc: int, n_mels: int, norm: Optional[str])torch.Tensor[source]

Create a DCT transformation matrix with shape (n_mels, n_mfcc), normalized depending on norm.

Parameters
  • n_mfcc (int) – Number of mfc coefficients to retain

  • n_mels (int) – Number of mel filterbanks

  • norm (str or None) – Norm to use (either ‘ortho’ or None)

Returns

The transformation matrix, to be right-multiplied to row-wise data of size (n_mels, n_mfcc).

Return type

Tensor

mask_along_axis

torchaudio.functional.mask_along_axis(specgram: torch.Tensor, mask_param: int, mask_value: float, axis: int, p: float = 1.0)torch.Tensor[source]

Apply a mask along axis. Mask will be applied from indices [v_0, v_0 + v), where v is sampled from uniform(0, max_v) and v_0 from uniform(0, specgrams.size(axis) - v), with max_v = mask_param when p = 1.0 and max_v = min(mask_param, floor(specgrams.size(axis) * p)) otherwise. All examples will have the same mask interval.

Parameters
  • specgram (Tensor) – Real spectrogram (channel, freq, time)

  • mask_param (int) – Number of columns to be masked will be uniformly sampled from [0, mask_param]

  • mask_value (float) – Value to assign to the masked columns

  • axis (int) – Axis to apply masking on (1 -> frequency, 2 -> time)

  • p (float, optional) – maximum proportion of columns that can be masked. (Default: 1.0)

Returns

Masked spectrogram of dimensions (channel, freq, time)

Return type

Tensor

mask_along_axis_iid

torchaudio.functional.mask_along_axis_iid(specgrams: torch.Tensor, mask_param: int, mask_value: float, axis: int, p: float = 1.0)torch.Tensor[source]

Apply a mask along axis. Mask will be applied from indices [v_0, v_0 + v), where v is sampled from uniform(0, max_v) and v_0 from uniform(0, specgrams.size(axis) - v), with max_v = mask_param when p = 1.0 and max_v = min(mask_param, floor(specgrams.size(axis) * p)) otherwise.

Parameters
  • specgrams (Tensor) – Real spectrograms (batch, channel, freq, time)

  • mask_param (int) – Number of columns to be masked will be uniformly sampled from [0, mask_param]

  • mask_value (float) – Value to assign to the masked columns

  • axis (int) – Axis to apply masking on (2 -> frequency, 3 -> time)

  • p (float, optional) – maximum proportion of columns that can be masked. (Default: 1.0)

Returns

Masked spectrograms of dimensions (batch, channel, freq, time)

Return type

Tensor

mu_law_encoding

torchaudio.functional.mu_law_encoding(x: torch.Tensor, quantization_channels: int)torch.Tensor[source]

Encode signal based on mu-law companding. For more info see the Wikipedia Entry

This algorithm expects the signal has been scaled to between -1 and 1 and returns a signal encoded with values from 0 to quantization_channels - 1.

Parameters
  • x (Tensor) – Input tensor

  • quantization_channels (int) – Number of channels

Returns

Input after mu-law encoding

Return type

Tensor

mu_law_decoding

torchaudio.functional.mu_law_decoding(x_mu: torch.Tensor, quantization_channels: int)torch.Tensor[source]

Decode mu-law encoded signal. For more info see the Wikipedia Entry

This expects an input with values between 0 and quantization_channels - 1 and returns a signal scaled between -1 and 1.

Parameters
  • x_mu (Tensor) – Input tensor

  • quantization_channels (int) – Number of channels

Returns

Input after mu-law decoding

Return type

Tensor

apply_codec

torchaudio.functional.apply_codec(waveform: torch.Tensor, sample_rate: int, format: str, channels_first: bool = True, compression: Optional[float] = None, encoding: Optional[str] = None, bits_per_sample: Optional[int] = None)torch.Tensor[source]

Apply codecs as a form of augmentation.

Parameters
  • waveform (Tensor) – Audio data. Must be 2 dimensional. See also `channels_first`.

  • sample_rate (int) – Sample rate of the audio waveform.

  • format (str) – File format.

  • channels_first (bool, optional) – When True, both the input and output Tensor have dimension (channel, time). Otherwise, they have dimension (time, channel).

  • compression (float or None, optional) – Used for formats other than WAV. For more details see torchaudio.backend.sox_io_backend.save().

  • encoding (str or None, optional) – Changes the encoding for the supported formats. For more details see torchaudio.backend.sox_io_backend.save().

  • bits_per_sample (int or None, optional) – Changes the bit depth for the supported formats. For more details see torchaudio.backend.sox_io_backend.save().

Returns

Resulting Tensor. If channels_first=True, it has (channel, time) else (time, channel).

Return type

Tensor

Tutorials using apply_codec:

resample

torchaudio.functional.resample(waveform: torch.Tensor, orig_freq: int, new_freq: int, lowpass_filter_width: int = 6, rolloff: float = 0.99, resampling_method: str = 'sinc_interpolation', beta: Optional[float] = None)torch.Tensor[source]

Resamples the waveform at the new frequency using bandlimited interpolation.

https://ccrma.stanford.edu/~jos/resample/Theory_Ideal_Bandlimited_Interpolation.html

Note

transforms.Resample precomputes and reuses the resampling kernel, so using it will result in more efficient computation if resampling multiple waveforms with the same resampling parameters.

Parameters
  • waveform (Tensor) – The input signal of dimension (…, time)

  • orig_freq (int) – The original frequency of the signal

  • new_freq (int) – The desired frequency

  • lowpass_filter_width (int, optional) – Controls the sharpness of the filter, more == sharper but less efficient. (Default: 6)

  • rolloff (float, optional) – The roll-off frequency of the filter, as a fraction of the Nyquist. Lower values reduce anti-aliasing, but also reduce some of the highest frequencies. (Default: 0.99)

  • resampling_method (str, optional) – The resampling method to use. Options: [sinc_interpolation, kaiser_window] (Default: 'sinc_interpolation')

  • beta (float or None, optional) – The shape parameter used for kaiser window.

Returns

The waveform at the new frequency of dimension (…, time).

Return type

Tensor

Tutorials using resample:

Filtering

allpass_biquad

torchaudio.functional.allpass_biquad(waveform: torch.Tensor, sample_rate: int, central_freq: float, Q: float = 0.707)torch.Tensor[source]

Design two-pole all-pass filter. Similar to SoX implementation.

Parameters
Returns

Waveform of dimension of (…, time)

Return type

Tensor

Reference:

band_biquad

torchaudio.functional.band_biquad(waveform: torch.Tensor, sample_rate: int, central_freq: float, Q: float = 0.707, noise: bool = False)torch.Tensor[source]

Design two-pole band filter. Similar to SoX implementation.

Parameters
  • waveform (Tensor) – audio waveform of dimension of (…, time)

  • sample_rate (int) – sampling rate of the waveform, e.g. 44100 (Hz)

  • central_freq (float or torch.Tensor) – central frequency (in Hz)

  • Q (float or torch.Tensor, optional) – https://en.wikipedia.org/wiki/Q_factor (Default: 0.707).

  • noise (bool, optional) – If True, uses the alternate mode for un-pitched audio (e.g. percussion). If False, uses mode oriented to pitched audio, i.e. voice, singing, or instrumental music (Default: False).

Returns

Waveform of dimension of (…, time)

Return type

Tensor

Reference:

bandpass_biquad

torchaudio.functional.bandpass_biquad(waveform: torch.Tensor, sample_rate: int, central_freq: float, Q: float = 0.707, const_skirt_gain: bool = False)torch.Tensor[source]

Design two-pole band-pass filter. Similar to SoX implementation.

Parameters
  • waveform (Tensor) – audio waveform of dimension of (…, time)

  • sample_rate (int) – sampling rate of the waveform, e.g. 44100 (Hz)

  • central_freq (float or torch.Tensor) – central frequency (in Hz)

  • Q (float or torch.Tensor, optional) – https://en.wikipedia.org/wiki/Q_factor (Default: 0.707)

  • const_skirt_gain (bool, optional) – If True, uses a constant skirt gain (peak gain = Q). If False, uses a constant 0dB peak gain. (Default: False)

Returns

Waveform of dimension of (…, time)

Return type

Tensor

Reference:

bandreject_biquad

torchaudio.functional.bandreject_biquad(waveform: torch.Tensor, sample_rate: int, central_freq: float, Q: float = 0.707)torch.Tensor[source]

Design two-pole band-reject filter. Similar to SoX implementation.

Parameters
Returns

Waveform of dimension of (…, time)

Return type

Tensor

Reference:

bass_biquad

torchaudio.functional.bass_biquad(waveform: torch.Tensor, sample_rate: int, gain: float, central_freq: float = 100, Q: float = 0.707)torch.Tensor[source]

Design a bass tone-control effect. Similar to SoX implementation.

Parameters
Returns

Waveform of dimension of (…, time)

Return type

Tensor

Reference:

biquad

torchaudio.functional.biquad(waveform: torch.Tensor, b0: float, b1: float, b2: float, a0: float, a1: float, a2: float)torch.Tensor[source]

Perform a biquad filter of input tensor. Initial conditions set to 0. https://en.wikipedia.org/wiki/Digital_biquad_filter

Parameters
  • waveform (Tensor) – audio waveform of dimension of (…, time)

  • b0 (float or torch.Tensor) – numerator coefficient of current input, x[n]

  • b1 (float or torch.Tensor) – numerator coefficient of input one time step ago x[n-1]

  • b2 (float or torch.Tensor) – numerator coefficient of input two time steps ago x[n-2]

  • a0 (float or torch.Tensor) – denominator coefficient of current output y[n], typically 1

  • a1 (float or torch.Tensor) – denominator coefficient of current output y[n-1]

  • a2 (float or torch.Tensor) – denominator coefficient of current output y[n-2]

Returns

Waveform with dimension of (…, time)

Return type

Tensor

contrast

torchaudio.functional.contrast(waveform: torch.Tensor, enhancement_amount: float = 75.0)torch.Tensor[source]

Apply contrast effect. Similar to SoX implementation. Comparable with compression, this effect modifies an audio signal to make it sound louder

Parameters
  • waveform (Tensor) – audio waveform of dimension of (…, time)

  • enhancement_amount (float, optional) – controls the amount of the enhancement Allowed range of values for enhancement_amount : 0-100 Note that enhancement_amount = 0 still gives a significant contrast enhancement

Returns

Waveform of dimension of (…, time)

Return type

Tensor

Reference:

dcshift

torchaudio.functional.dcshift(waveform: torch.Tensor, shift: float, limiter_gain: Optional[float] = None)torch.Tensor[source]

Apply a DC shift to the audio. Similar to SoX implementation. This can be useful to remove a DC offset (caused perhaps by a hardware problem in the recording chain) from the audio

Parameters
  • waveform (Tensor) – audio waveform of dimension of (…, time)

  • shift (float) – indicates the amount to shift the audio Allowed range of values for shift : -2.0 to +2.0

  • limiter_gain (float of None, optional) – It is used only on peaks to prevent clipping It should have a value much less than 1 (e.g. 0.05 or 0.02)

Returns

Waveform of dimension of (…, time)

Return type

Tensor

Reference:

deemph_biquad

torchaudio.functional.deemph_biquad(waveform: torch.Tensor, sample_rate: int)torch.Tensor[source]

Apply ISO 908 CD de-emphasis (shelving) IIR filter. Similar to SoX implementation.

Parameters
  • waveform (Tensor) – audio waveform of dimension of (…, time)

  • sample_rate (int) – sampling rate of the waveform, Allowed sample rate 44100 or 48000

Returns

Waveform of dimension of (…, time)

Return type

Tensor

Reference:

dither

torchaudio.functional.dither(waveform: torch.Tensor, density_function: str = 'TPDF', noise_shaping: bool = False)torch.Tensor[source]

Dither increases the perceived dynamic range of audio stored at a particular bit-depth by eliminating nonlinear truncation distortion (i.e. adding minimally perceived noise to mask distortion caused by quantization).

Parameters
  • waveform (Tensor) – Tensor of audio of dimension (…, time)

  • density_function (str, optional) – The density function of a continuous random variable. One of "TPDF" (Triangular Probability Density Function), "RPDF" (Rectangular Probability Density Function) or "GPDF" (Gaussian Probability Density Function) (Default: "TPDF").

  • noise_shaping (bool, optional) – a filtering process that shapes the spectral energy of quantisation error (Default: False)

Returns

waveform dithered

Return type

Tensor

equalizer_biquad

torchaudio.functional.equalizer_biquad(waveform: torch.Tensor, sample_rate: int, center_freq: float, gain: float, Q: float = 0.707)torch.Tensor[source]

Design biquad peaking equalizer filter and perform filtering. Similar to SoX implementation.

Parameters
Returns

Waveform of dimension of (…, time)

Return type

Tensor

filtfilt

torchaudio.functional.filtfilt(waveform: torch.Tensor, a_coeffs: torch.Tensor, b_coeffs: torch.Tensor, clamp: bool = True)torch.Tensor[source]

Apply an IIR filter forward and backward to a waveform.

Inspired by https://docs.scipy.org/doc/scipy/reference/generated/scipy.signal.filtfilt.html

Parameters
  • waveform (Tensor) – audio waveform of dimension of (…, time). Must be normalized to -1 to 1.

  • a_coeffs (Tensor) – denominator coefficients of difference equation of dimension of either 1D with shape (num_order + 1) or 2D with shape (num_filters, num_order + 1). Lower delay coefficients are first, e.g. [a0, a1, a2, ...]. Must be same size as b_coeffs (pad with 0’s as necessary).

  • b_coeffs (Tensor) – numerator coefficients of difference equation of dimension of either 1D with shape (num_order + 1) or 2D with shape (num_filters, num_order + 1). Lower delay coefficients are first, e.g. [b0, b1, b2, ...]. Must be same size as a_coeffs (pad with 0’s as necessary).

  • clamp (bool, optional) – If True, clamp the output signal to be in the range [-1, 1] (Default: True)

Returns

Waveform with dimension of either (…, num_filters, time) if a_coeffs and b_coeffs are 2D Tensors, or (…, time) otherwise.

Return type

Tensor

flanger

torchaudio.functional.flanger(waveform: torch.Tensor, sample_rate: int, delay: float = 0.0, depth: float = 2.0, regen: float = 0.0, width: float = 71.0, speed: float = 0.5, phase: float = 25.0, modulation: str = 'sinusoidal', interpolation: str = 'linear')torch.Tensor[source]

Apply a flanger effect to the audio. Similar to SoX implementation.

Parameters
  • waveform (Tensor) – audio waveform of dimension of (…, channel, time) . Max 4 channels allowed

  • sample_rate (int) – sampling rate of the waveform, e.g. 44100 (Hz)

  • delay (float, optional) – desired delay in milliseconds(ms) Allowed range of values are 0 to 30

  • depth (float, optional) – desired delay depth in milliseconds(ms) Allowed range of values are 0 to 10

  • regen (float, optional) – desired regen(feedback gain) in dB Allowed range of values are -95 to 95

  • width (float, optional) – desired width(delay gain) in dB Allowed range of values are 0 to 100

  • speed (float, optional) – modulation speed in Hz Allowed range of values are 0.1 to 10

  • phase (float, optional) – percentage phase-shift for multi-channel Allowed range of values are 0 to 100

  • modulation (str, optional) – Use either “sinusoidal” or “triangular” modulation. (Default: sinusoidal)

  • interpolation (str, optional) – Use either “linear” or “quadratic” for delay-line interpolation. (Default: linear)

Returns

Waveform of dimension of (…, channel, time)

Return type

Tensor

Reference:

gain

torchaudio.functional.gain(waveform: torch.Tensor, gain_db: float = 1.0)torch.Tensor[source]

Apply amplification or attenuation to the whole waveform.

Parameters
  • waveform (Tensor) – Tensor of audio of dimension (…, time).

  • gain_db (float, optional) Gain adjustment in decibels (dB) – 1.0).

Returns

the whole waveform amplified by gain_db.

Return type

Tensor

highpass_biquad

torchaudio.functional.highpass_biquad(waveform: torch.Tensor, sample_rate: int, cutoff_freq: float, Q: float = 0.707)torch.Tensor[source]

Design biquad highpass filter and perform filtering. Similar to SoX implementation.

Parameters
Returns

Waveform dimension of (…, time)

Return type

Tensor

lfilter

torchaudio.functional.lfilter(waveform: torch.Tensor, a_coeffs: torch.Tensor, b_coeffs: torch.Tensor, clamp: bool = True, batching: bool = True)torch.Tensor[source]

Perform an IIR filter by evaluating difference equation.

Note

To avoid numerical problems, small filter order is preferred. Using double precision could also minimize numerical precision errors.

Parameters
  • waveform (Tensor) – audio waveform of dimension of (…, time). Must be normalized to -1 to 1.

  • a_coeffs (Tensor) – denominator coefficients of difference equation of dimension of either 1D with shape (num_order + 1) or 2D with shape (num_filters, num_order + 1). Lower delays coefficients are first, e.g. [a0, a1, a2, ...]. Must be same size as b_coeffs (pad with 0’s as necessary).

  • b_coeffs (Tensor) – numerator coefficients of difference equation of dimension of either 1D with shape (num_order + 1) or 2D with shape (num_filters, num_order + 1). Lower delays coefficients are first, e.g. [b0, b1, b2, ...]. Must be same size as a_coeffs (pad with 0’s as necessary).

  • clamp (bool, optional) – If True, clamp the output signal to be in the range [-1, 1] (Default: True)

  • batching (bool, optional) – Effective only when coefficients are 2D. If True, then waveform should be at least 2D, and the size of second axis from last should equals to num_filters. The output can be expressed as output[..., i, :] = lfilter(waveform[..., i, :], a_coeffs[i], b_coeffs[i], clamp=clamp, batching=False). (Default: True)

Returns

Waveform with dimension of either (…, num_filters, time) if a_coeffs and b_coeffs are 2D Tensors, or (…, time) otherwise.

Return type

Tensor

lowpass_biquad

torchaudio.functional.lowpass_biquad(waveform: torch.Tensor, sample_rate: int, cutoff_freq: float, Q: float = 0.707)torch.Tensor[source]

Design biquad lowpass filter and perform filtering. Similar to SoX implementation.

Parameters
Returns

Waveform of dimension of (…, time)

Return type

Tensor

overdrive

torchaudio.functional.overdrive(waveform: torch.Tensor, gain: float = 20, colour: float = 20)torch.Tensor[source]

Apply a overdrive effect to the audio. Similar to SoX implementation. This effect applies a non linear distortion to the audio signal.

Parameters
  • waveform (Tensor) – audio waveform of dimension of (…, time)

  • gain (float, optional) – desired gain at the boost (or attenuation) in dB Allowed range of values are 0 to 100

  • colour (float, optional) – controls the amount of even harmonic content in the over-driven output Allowed range of values are 0 to 100

Returns

Waveform of dimension of (…, time)

Return type

Tensor

Reference:

phaser

torchaudio.functional.phaser(waveform: torch.Tensor, sample_rate: int, gain_in: float = 0.4, gain_out: float = 0.74, delay_ms: float = 3.0, decay: float = 0.4, mod_speed: float = 0.5, sinusoidal: bool = True)torch.Tensor[source]

Apply a phasing effect to the audio. Similar to SoX implementation.

Parameters
  • waveform (Tensor) – audio waveform of dimension of (…, time)

  • sample_rate (int) – sampling rate of the waveform, e.g. 44100 (Hz)

  • gain_in (float, optional) – desired input gain at the boost (or attenuation) in dB Allowed range of values are 0 to 1

  • gain_out (float, optional) – desired output gain at the boost (or attenuation) in dB Allowed range of values are 0 to 1e9

  • delay_ms (float, optional) – desired delay in milliseconds Allowed range of values are 0 to 5.0

  • decay (float, optional) – desired decay relative to gain-in Allowed range of values are 0 to 0.99

  • mod_speed (float, optional) – modulation speed in Hz Allowed range of values are 0.1 to 2

  • sinusoidal (bool, optional) – If True, uses sinusoidal modulation (preferable for multiple instruments) If False, uses triangular modulation (gives single instruments a sharper phasing effect) (Default: True)

Returns

Waveform of dimension of (…, time)

Return type

Tensor

Reference:

riaa_biquad

torchaudio.functional.riaa_biquad(waveform: torch.Tensor, sample_rate: int)torch.Tensor[source]

Apply RIAA vinyl playback equalization. Similar to SoX implementation.

Parameters
  • waveform (Tensor) – audio waveform of dimension of (…, time)

  • sample_rate (int) – sampling rate of the waveform, e.g. 44100 (Hz). Allowed sample rates in Hz : 44100,``48000``,``88200``,``96000``

Returns

Waveform of dimension of (…, time)

Return type

Tensor

Reference:

treble_biquad

torchaudio.functional.treble_biquad(waveform: torch.Tensor, sample_rate: int, gain: float, central_freq: float = 3000, Q: float = 0.707)torch.Tensor[source]

Design a treble tone-control effect. Similar to SoX implementation.

Parameters
Returns

Waveform of dimension of (…, time)

Return type

Tensor

Reference:

Feature Extractions

vad

torchaudio.functional.vad(waveform: torch.Tensor, sample_rate: int, trigger_level: float = 7.0, trigger_time: float = 0.25, search_time: float = 1.0, allowed_gap: float = 0.25, pre_trigger_time: float = 0.0, boot_time: float = 0.35, noise_up_time: float = 0.1, noise_down_time: float = 0.01, noise_reduction_amount: float = 1.35, measure_freq: float = 20.0, measure_duration: Optional[float] = None, measure_smooth_time: float = 0.4, hp_filter_freq: float = 50.0, lp_filter_freq: float = 6000.0, hp_lifter_freq: float = 150.0, lp_lifter_freq: float = 2000.0)torch.Tensor[source]

Voice Activity Detector. Similar to SoX implementation. Attempts to trim silence and quiet background sounds from the ends of recordings of speech. The algorithm currently uses a simple cepstral power measurement to detect voice, so may be fooled by other things, especially music.

The effect can trim only from the front of the audio, so in order to trim from the back, the reverse effect must also be used.

Parameters
  • waveform (Tensor) – Tensor of audio of dimension (channels, time) or (time) Tensor of shape (channels, time) is treated as a multi-channel recording of the same event and the resulting output will be trimmed to the earliest voice activity in any channel.

  • sample_rate (int) – Sample rate of audio signal.

  • trigger_level (float, optional) – The measurement level used to trigger activity detection. This may need to be cahnged depending on the noise level, signal level, and other characteristics of the input audio. (Default: 7.0)

  • trigger_time (float, optional) – The time constant (in seconds) used to help ignore short bursts of sound. (Default: 0.25)

  • search_time (float, optional) – The amount of audio (in seconds) to search for quieter/shorter bursts of audio to include prior to the detected trigger point. (Default: 1.0)

  • allowed_gap (float, optional) – The allowed gap (in seconds) between quieter/shorter bursts of audio to include prior to the detected trigger point. (Default: 0.25)

  • pre_trigger_time (float, optional) – The amount of audio (in seconds) to preserve before the trigger point and any found quieter/shorter bursts. (Default: 0.0)

  • boot_time (float, optional) The algorithm (internally) – estimation/reduction in order to detect the start of the wanted audio. This option sets the time for the initial noise estimate. (Default: 0.35)

  • noise_up_time (float, optional) – for when the noise level is increasing. (Default: 0.1)

  • noise_down_time (float, optional) – for when the noise level is decreasing. (Default: 0.01)

  • noise_reduction_amount (float, optional) – the detection algorithm (e.g. 0, 0.5, …). (Default: 1.35)

  • measure_freq (float, optional) – processing/measurements. (Default: 20.0)

  • measure_duration – (float, optional) Measurement duration. (Default: Twice the measurement period; i.e. with overlap.)

  • measure_smooth_time (float, optional) – spectral measurements. (Default: 0.4)

  • hp_filter_freq (float, optional) – at the input to the detector algorithm. (Default: 50.0)

  • lp_filter_freq (float, optional) – at the input to the detector algorithm. (Default: 6000.0)

  • hp_lifter_freq (float, optional) – in the detector algorithm. (Default: 150.0)

  • lp_lifter_freq (float, optional) – in the detector algorithm. (Default: 2000.0)

Returns

Tensor of audio of dimension (…, time).

Return type

Tensor

Reference:

spectrogram

torchaudio.functional.spectrogram(waveform: torch.Tensor, pad: int, window: torch.Tensor, n_fft: int, hop_length: int, win_length: int, power: Optional[float], normalized: bool, center: bool = True, pad_mode: str = 'reflect', onesided: bool = True, return_complex: Optional[bool] = None)torch.Tensor[source]

Create a spectrogram or a batch of spectrograms from a raw audio signal. The spectrogram can be either magnitude-only or complex.

Parameters
  • waveform (Tensor) – Tensor of audio of dimension (…, time)

  • pad (int) – Two sided padding of signal

  • window (Tensor) – Window tensor that is applied/multiplied to each frame/window

  • n_fft (int) – Size of FFT

  • hop_length (int) – Length of hop between STFT windows

  • win_length (int) – Window size

  • power (float or None) – Exponent for the magnitude spectrogram, (must be > 0) e.g., 1 for energy, 2 for power, etc. If None, then the complex spectrum is returned instead.

  • normalized (bool) – Whether to normalize by magnitude after stft

  • center (bool, optional) – whether to pad waveform on both sides so that the \(t\)-th frame is centered at time \(t \times \text{hop\_length}\). Default: True

  • pad_mode (string, optional) – controls the padding method used when center is True. Default: "reflect"

  • onesided (bool, optional) – controls whether to return half of results to avoid redundancy. Default: True

  • return_complex (bool, optional) – Deprecated and not used.

Returns

Dimension (…, freq, time), freq is n_fft // 2 + 1 and n_fft is the number of Fourier bins, and time is the number of window hops (n_frame).

Return type

Tensor

inverse_spectrogram

torchaudio.functional.inverse_spectrogram(spectrogram: torch.Tensor, length: Optional[int], pad: int, window: torch.Tensor, n_fft: int, hop_length: int, win_length: int, normalized: bool, center: bool = True, pad_mode: str = 'reflect', onesided: bool = True)torch.Tensor[source]

Create an inverse spectrogram or a batch of inverse spectrograms from the provided complex-valued spectrogram.

Parameters
  • spectrogram (Tensor) – Complex tensor of audio of dimension (…, freq, time).

  • length (int or None) – The output length of the waveform.

  • pad (int) – Two sided padding of signal. It is only effective when length is provided.

  • window (Tensor) – Window tensor that is applied/multiplied to each frame/window

  • n_fft (int) – Size of FFT

  • hop_length (int) – Length of hop between STFT windows

  • win_length (int) – Window size

  • normalized (bool) – Whether the stft output was normalized by magnitude

  • center (bool, optional) – whether the waveform was padded on both sides so that the \(t\)-th frame is centered at time \(t \times \text{hop\_length}\). Default: True

  • pad_mode (string, optional) – controls the padding method used when center is True. This parameter is provided for compatibility with the spectrogram function and is not used. Default: "reflect"

  • onesided (bool, optional) – controls whether spectrogram was done in onesided mode. Default: True

Returns

Dimension (…, time). Least squares estimation of the original signal.

Return type

Tensor

griffinlim

torchaudio.functional.griffinlim(specgram: torch.Tensor, window: torch.Tensor, n_fft: int, hop_length: int, win_length: int, power: float, n_iter: int, momentum: float, length: Optional[int], rand_init: bool)torch.Tensor[source]

Compute waveform from a linear scale magnitude spectrogram using the Griffin-Lim transformation.

Implementation ported from librosa [1], A fast Griffin-Lim algorithm [2] and Signal estimation from modified short-time Fourier transform [3].

Parameters
  • specgram (Tensor) – A magnitude-only STFT spectrogram of dimension (…, freq, frames) where freq is n_fft // 2 + 1.

  • window (Tensor) – Window tensor that is applied/multiplied to each frame/window

  • n_fft (int) – Size of FFT, creates n_fft // 2 + 1 bins

  • hop_length (int) – Length of hop between STFT windows. ( Default: win_length // 2)

  • win_length (int) – Window size. (Default: n_fft)

  • power (float) – Exponent for the magnitude spectrogram, (must be > 0) e.g., 1 for energy, 2 for power, etc.

  • n_iter (int) – Number of iteration for phase recovery process.

  • momentum (float) – The momentum parameter for fast Griffin-Lim. Setting this to 0 recovers the original Griffin-Lim method. Values near 1 can lead to faster convergence, but above 1 may not converge.

  • length (int or None) – Array length of the expected output.

  • rand_init (bool) – Initializes phase randomly if True, to zero otherwise.

Returns

waveform of (…, time), where time equals the length parameter if given.

Return type

Tensor

phase_vocoder

torchaudio.functional.phase_vocoder(complex_specgrams: torch.Tensor, rate: float, phase_advance: torch.Tensor)torch.Tensor[source]

Given a STFT tensor, speed up in time without modifying pitch by a factor of rate.

Parameters
  • complex_specgrams (Tensor) – A tensor of dimension (…, freq, num_frame) with complex dtype.

  • rate (float) – Speed-up factor

  • phase_advance (Tensor) – Expected phase advance in each bin. Dimension of (freq, 1)

Returns

Stretched spectrogram. The resulting tensor is of the same dtype as the input spectrogram, but the number of frames is changed to ceil(num_frame / rate).

Return type

Tensor

Example
>>> freq, hop_length = 1025, 512
>>> # (channel, freq, time)
>>> complex_specgrams = torch.randn(2, freq, 300, dtype=torch.cfloat)
>>> rate = 1.3 # Speed up by 30%
>>> phase_advance = torch.linspace(
>>>    0, math.pi * hop_length, freq)[..., None]
>>> x = phase_vocoder(complex_specgrams, rate, phase_advance)
>>> x.shape # with 231 == ceil(300 / 1.3)
torch.Size([2, 1025, 231])

pitch_shift

torchaudio.functional.pitch_shift(waveform: torch.Tensor, sample_rate: int, n_steps: int, bins_per_octave: int = 12, n_fft: int = 512, win_length: Optional[int] = None, hop_length: Optional[int] = None, window: Optional[torch.Tensor] = None)torch.Tensor[source]

Shift the pitch of a waveform by n_steps steps.

Parameters
  • waveform (Tensor) – The input waveform of shape (…, time).

  • sample_rate (int) – Sample rate of waveform.

  • n_steps (int) – The (fractional) steps to shift waveform.

  • bins_per_octave (int, optional) – The number of steps per octave (Default: 12).

  • n_fft (int, optional) – Size of FFT, creates n_fft // 2 + 1 bins (Default: 512).

  • win_length (int or None, optional) – Window size. If None, then n_fft is used. (Default: None).

  • hop_length (int or None, optional) – Length of hop between STFT windows. If None, then win_length // 4 is used (Default: None).

  • window (Tensor or None, optional) – Window tensor that is applied/multiplied to each frame/window. If None, then torch.hann_window(win_length) is used (Default: None).

Returns

The pitch-shifted audio waveform of shape (…, time).

Return type

Tensor

compute_deltas

torchaudio.functional.compute_deltas(specgram: torch.Tensor, win_length: int = 5, mode: str = 'replicate')torch.Tensor[source]

Compute delta coefficients of a tensor, usually a spectrogram:

\[d_t = \frac{\sum_{n=1}^{\text{N}} n (c_{t+n} - c_{t-n})}{2 \sum_{n=1}^{\text{N}} n^2} \]

where \(d_t\) is the deltas at time \(t\), \(c_t\) is the spectrogram coeffcients at time \(t\), \(N\) is (win_length-1)//2.

Parameters
  • specgram (Tensor) – Tensor of audio of dimension (…, freq, time)

  • win_length (int, optional) – The window length used for computing delta (Default: 5)

  • mode (str, optional) – Mode parameter passed to padding (Default: "replicate")

Returns

Tensor of deltas of dimension (…, freq, time)

Return type

Tensor

Example
>>> specgram = torch.randn(1, 40, 1000)
>>> delta = compute_deltas(specgram)
>>> delta2 = compute_deltas(delta)

detect_pitch_frequency

torchaudio.functional.detect_pitch_frequency(waveform: torch.Tensor, sample_rate: int, frame_time: float = 0.01, win_length: int = 30, freq_low: int = 85, freq_high: int = 3400)torch.Tensor[source]

Detect pitch frequency.

It is implemented using normalized cross-correlation function and median smoothing.

Parameters
  • waveform (Tensor) – Tensor of audio of dimension (…, freq, time)

  • sample_rate (int) – The sample rate of the waveform (Hz)

  • frame_time (float, optional) – Duration of a frame (Default: 10 ** (-2)).

  • win_length (int, optional) – The window length for median smoothing (in number of frames) (Default: 30).

  • freq_low (int, optional) – Lowest frequency that can be detected (Hz) (Default: 85).

  • freq_high (int, optional) – Highest frequency that can be detected (Hz) (Default: 3400).

Returns

Tensor of freq of dimension (…, frame)

Return type

Tensor

Tutorials using detect_pitch_frequency:

sliding_window_cmn

torchaudio.functional.sliding_window_cmn(specgram: torch.Tensor, cmn_window: int = 600, min_cmn_window: int = 100, center: bool = False, norm_vars: bool = False)torch.Tensor[source]

Apply sliding-window cepstral mean (and optionally variance) normalization per utterance.

Parameters
  • specgram (Tensor) – Tensor of spectrogram of dimension (…, time, freq)

  • cmn_window (int, optional) – Window in frames for running average CMN computation (int, default = 600)

  • min_cmn_window (int, optional) – Minimum CMN window used at start of decoding (adds latency only at start). Only applicable if center == false, ignored if center==true (int, default = 100)

  • center (bool, optional) – If true, use a window centered on the current frame (to the extent possible, modulo end effects). If false, window is to the left. (bool, default = false)

  • norm_vars (bool, optional) – If true, normalize variance to one. (bool, default = false)

Returns

Tensor matching input shape (…, freq, time)

Return type

Tensor

compute_kaldi_pitch

torchaudio.functional.compute_kaldi_pitch(waveform: torch.Tensor, sample_rate: float, frame_length: float = 25.0, frame_shift: float = 10.0, min_f0: float = 50, max_f0: float = 400, soft_min_f0: float = 10.0, penalty_factor: float = 0.1, lowpass_cutoff: float = 1000, resample_frequency: float = 4000, delta_pitch: float = 0.005, nccf_ballast: float = 7000, lowpass_filter_width: int = 1, upsample_filter_width: int = 5, max_frames_latency: int = 0, frames_per_chunk: int = 0, simulate_first_pass_online: bool = False, recompute_frame: int = 500, snip_edges: bool = True)torch.Tensor[source]

Extract pitch based on method described in A pitch extraction algorithm tuned for automatic speech recognition [4].

This function computes the equivalent of compute-kaldi-pitch-feats from Kaldi.

Parameters
  • waveform (Tensor) – The input waveform of shape (…, time).

  • sample_rate (float) – Sample rate of waveform.

  • frame_length (float, optional) – Frame length in milliseconds. (default: 25.0)

  • frame_shift (float, optional) – Frame shift in milliseconds. (default: 10.0)

  • min_f0 (float, optional) – Minimum F0 to search for (Hz) (default: 50.0)

  • max_f0 (float, optional) – Maximum F0 to search for (Hz) (default: 400.0)

  • soft_min_f0 (float, optional) – Minimum f0, applied in soft way, must not exceed min-f0 (default: 10.0)

  • penalty_factor (float, optional) – Cost factor for FO change. (default: 0.1)

  • lowpass_cutoff (float, optional) – Cutoff frequency for LowPass filter (Hz) (default: 1000)

  • resample_frequency (float, optional) – Frequency that we down-sample the signal to. Must be more than twice lowpass-cutoff. (default: 4000)

  • delta_pitch (float, optional) – Smallest relative change in pitch that our algorithm measures. (default: 0.005)

  • nccf_ballast (float, optional) – Increasing this factor reduces NCCF for quiet frames (default: 7000)

  • lowpass_filter_width (int, optional) – Integer that determines filter width of lowpass filter, more gives sharper filter. (default: 1)

  • upsample_filter_width (int, optional) – Integer that determines filter width when upsampling NCCF. (default: 5)

  • max_frames_latency (int, optional) – Maximum number of frames of latency that we allow pitch tracking to introduce into the feature processing (affects output only if frames_per_chunk > 0 and simulate_first_pass_online=True) (default: 0)

  • frames_per_chunk (int, optional) – The number of frames used for energy normalization. (default: 0)

  • simulate_first_pass_online (bool, optional) – If true, the function will output features that correspond to what an online decoder would see in the first pass of decoding – not the final version of the features, which is the default. (default: False) Relevant if frames_per_chunk > 0.

  • recompute_frame (int, optional) – Only relevant for compatibility with online pitch extraction. A non-critical parameter; the frame at which we recompute some of the forward pointers, after revising our estimate of the signal energy. Relevant if frames_per_chunk > 0. (default: 500)

  • snip_edges (bool, optional) – If this is set to false, the incomplete frames near the ending edge won’t be snipped, so that the number of frames is the file size divided by the frame-shift. This makes different types of features give the same number of frames. (default: True)

Returns

Pitch feature. Shape: (batch, frames 2) where the last dimension corresponds to pitch and NCCF.

Return type

Tensor

Tutorials using compute_kaldi_pitch:

spectral_centroid

torchaudio.functional.spectral_centroid(waveform: torch.Tensor, sample_rate: int, pad: int, window: torch.Tensor, n_fft: int, hop_length: int, win_length: int)torch.Tensor[source]

Compute the spectral centroid for each channel along the time axis.

The spectral centroid is defined as the weighted average of the frequency values, weighted by their magnitude.

Parameters
  • waveform (Tensor) – Tensor of audio of dimension (…, time)

  • sample_rate (int) – Sample rate of the audio waveform

  • pad (int) – Two sided padding of signal

  • window (Tensor) – Window tensor that is applied/multiplied to each frame/window

  • n_fft (int) – Size of FFT

  • hop_length (int) – Length of hop between STFT windows

  • win_length (int) – Window size

Returns

Dimension (…, time)

Return type

Tensor

Loss

rnnt_loss

torchaudio.functional.rnnt_loss(logits: torch.Tensor, targets: torch.Tensor, logit_lengths: torch.Tensor, target_lengths: torch.Tensor, blank: int = - 1, clamp: float = - 1, reduction: str = 'mean')[source]

Compute the RNN Transducer loss from Sequence Transduction with Recurrent Neural Networks [5]. The RNN Transducer loss extends the CTC loss by defining a distribution over output sequences of all lengths, and by jointly modelling both input-output and output-output dependencies.

Parameters
  • logits (Tensor) – Tensor of dimension (batch, max seq length, max target length + 1, class) containing output from joiner

  • targets (Tensor) – Tensor of dimension (batch, max target length) containing targets with zero padded

  • logit_lengths (Tensor) – Tensor of dimension (batch) containing lengths of each sequence from encoder

  • target_lengths (Tensor) – Tensor of dimension (batch) containing lengths of targets for each sequence

  • blank (int, optional) – blank label (Default: -1)

  • clamp (float, optional) – clamp for gradients (Default: -1)

  • reduction (string, optional) – Specifies the reduction to apply to the output: 'none' | 'mean' | 'sum'. (Default: 'mean')

Returns

Loss with the reduction option applied. If reduction is 'none', then size (batch), otherwise scalar.

Return type

Tensor

Metric

edit_distance

torchaudio.functional.edit_distance(seq1: collections.abc.Sequence, seq2: collections.abc.Sequence)int[source]

Calculate the word level edit (Levenshtein) distance between two sequences.

The function computes an edit distance allowing deletion, insertion and substitution. The result is an integer.

For most applications, the two input sequences should be the same type. If two strings are given, the output is the edit distance between the two strings (character edit distance). If two lists of strings are given, the output is the edit distance between sentences (word edit distance). Users may want to normalize the output by the length of the reference sequence.

torchscipt is not supported for this function.

Parameters
  • seq1 (Sequence) – the first sequence to compare.

  • seq2 (Sequence) – the second sequence to compare.

Returns

The distance between the first and second sequences.

Return type

int

Tutorials using edit_distance:

References

1

Brian McFee, Colin Raffel, Dawen Liang, Daniel P.W. Ellis, Matt McVicar, Eric Battenberg, and Oriol Nieto. Librosa: Audio and Music Signal Analysis in Python. In Kathryn Huff and James Bergstra, editors, Proceedings of the 14th Python in Science Conference, 18 – 24. 2015. doi:10.25080/Majora-7b98e3ed-003.

2

Nathanaël Perraudin, Peter Balazs, and Peter L. Søndergaard. A fast griffin-lim algorithm. In 2013 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, volume, 1–4. 2013. doi:10.1109/WASPAA.2013.6701851.

3

D. Griffin and Jae Lim. Signal estimation from modified short-time fourier transform. In ICASSP ‘83. IEEE International Conference on Acoustics, Speech, and Signal Processing, volume 8, 804–807. 1983. doi:10.1109/ICASSP.1983.1172092.

4

Pegah Ghahremani, Bagher BabaAli, Daniel Povey, Korbinian Riedhammer, Jan Trmal, and Sanjeev Khudanpur. A pitch extraction algorithm tuned for automatic speech recognition. In 2014 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), volume, 2494–2498. 2014. doi:10.1109/ICASSP.2014.6854049.

5

Alex Graves. Sequence transduction with recurrent neural networks. 2012. arXiv:1211.3711.

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