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https://github.com/ml-explore/mlx-examples.git
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Whisper: Add pip distribution configuration to support pip installations. (#739)
* Whisper: rename whisper to mlx_whisper * Whisper: add setup.py config for publish * Whisper: add assets data to setup config * Whisper: pre-commit for setup.py * Whisper: Update README.md * Whisper: Update README.md * nits * fix package data * nit in readme --------- Co-authored-by: Awni Hannun <awni@apple.com>
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174
whisper/mlx_whisper/audio.py
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174
whisper/mlx_whisper/audio.py
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# Copyright © 2023 Apple Inc.
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import os
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from functools import lru_cache
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from subprocess import CalledProcessError, run
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from typing import Union
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import mlx.core as mx
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import numpy as np
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# hard-coded audio hyperparameters
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SAMPLE_RATE = 16000
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N_FFT = 400
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HOP_LENGTH = 160
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CHUNK_LENGTH = 30
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N_SAMPLES = CHUNK_LENGTH * SAMPLE_RATE # 480000 samples in a 30-second chunk
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N_FRAMES = N_SAMPLES // HOP_LENGTH # 3000 frames in a mel spectrogram input
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N_SAMPLES_PER_TOKEN = HOP_LENGTH * 2 # the initial convolutions has stride 2
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FRAMES_PER_SECOND = SAMPLE_RATE // HOP_LENGTH # 10ms per audio frame
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TOKENS_PER_SECOND = SAMPLE_RATE // N_SAMPLES_PER_TOKEN # 20ms per audio token
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def load_audio(file: str, sr: int = SAMPLE_RATE):
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"""
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Open an audio file and read as mono waveform, resampling as necessary
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Parameters
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----------
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file: str
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The audio file to open
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sr: int
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The sample rate to resample the audio if necessary
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Returns
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-------
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A NumPy array containing the audio waveform, in float32 dtype.
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"""
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# This launches a subprocess to decode audio while down-mixing
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# and resampling as necessary. Requires the ffmpeg CLI in PATH.
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# fmt: off
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cmd = [
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"ffmpeg",
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"-nostdin",
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"-threads", "0",
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"-i", file,
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"-f", "s16le",
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"-ac", "1",
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"-acodec", "pcm_s16le",
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"-ar", str(sr),
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"-"
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]
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# fmt: on
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try:
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out = run(cmd, capture_output=True, check=True).stdout
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except CalledProcessError as e:
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raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e
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return mx.array(np.frombuffer(out, np.int16)).flatten().astype(mx.float32) / 32768.0
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def pad_or_trim(array, length: int = N_SAMPLES, *, axis: int = -1):
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"""
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Pad or trim the audio array to N_SAMPLES, as expected by the encoder.
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"""
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if array.shape[axis] > length:
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sl = [slice(None)] * array.ndim
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sl[axis] = slice(0, length)
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array = array[tuple(sl)]
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if array.shape[axis] < length:
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pad_widths = [(0, 0)] * array.ndim
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pad_widths[axis] = (0, length - array.shape[axis])
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array = mx.pad(array, pad_widths)
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return array
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@lru_cache(maxsize=None)
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def mel_filters(n_mels: int) -> mx.array:
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"""
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load the mel filterbank matrix for projecting STFT into a Mel spectrogram.
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Allows decoupling librosa dependency; saved using:
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np.savez_compressed(
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"mel_filters.npz",
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mel_80=librosa.filters.mel(sr=16000, n_fft=400, n_mels=80),
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mel_128=librosa.filters.mel(sr=16000, n_fft=400, n_mels=128),
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)
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"""
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assert n_mels in {80, 128}, f"Unsupported n_mels: {n_mels}"
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filename = os.path.join(os.path.dirname(__file__), "assets", "mel_filters.npz")
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return mx.load(filename)[f"mel_{n_mels}"]
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@lru_cache(maxsize=None)
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def hanning(size):
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return mx.array(np.hanning(size + 1)[:-1])
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def stft(x, window, nperseg=256, noverlap=None, nfft=None, axis=-1, pad_mode="reflect"):
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if nfft is None:
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nfft = nperseg
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if noverlap is None:
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noverlap = nfft // 4
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def _pad(x, padding, pad_mode="constant"):
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if pad_mode == "constant":
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return mx.pad(x, [(padding, padding)])
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elif pad_mode == "reflect":
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prefix = x[1 : padding + 1][::-1]
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suffix = x[-(padding + 1) : -1][::-1]
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return mx.concatenate([prefix, x, suffix])
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else:
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raise ValueError(f"Invalid pad_mode {pad_mode}")
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padding = nperseg // 2
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x = _pad(x, padding, pad_mode)
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strides = [noverlap, 1]
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t = (x.size - nperseg + noverlap) // noverlap
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shape = [t, nfft]
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x = mx.as_strided(x, shape=shape, strides=strides)
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return mx.fft.rfft(x * window)
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def log_mel_spectrogram(
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audio: Union[str, np.ndarray],
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n_mels: int = 80,
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padding: int = 0,
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):
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"""
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Compute the log-Mel spectrogram of
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Parameters
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----------
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audio: Union[str, np.ndarray, mx.array], shape = (*)
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The path to audio or either a NumPy or mlx array containing the audio waveform in 16 kHz
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n_mels: int
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The number of Mel-frequency filters, only 80 is supported
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padding: int
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Number of zero samples to pad to the right
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Returns
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-------
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mx.array, shape = (80, n_frames)
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An array that contains the Mel spectrogram
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"""
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device = mx.default_device()
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mx.set_default_device(mx.cpu)
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if isinstance(audio, str):
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audio = load_audio(audio)
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elif not isinstance(audio, mx.array):
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audio = mx.array(audio)
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if padding > 0:
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audio = mx.pad(audio, (0, padding))
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window = hanning(N_FFT)
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freqs = stft(audio, window, nperseg=N_FFT, noverlap=HOP_LENGTH)
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magnitudes = freqs[:-1, :].abs().square()
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filters = mel_filters(n_mels)
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mel_spec = magnitudes @ filters.T
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log_spec = mx.maximum(mel_spec, 1e-10).log10()
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log_spec = mx.maximum(log_spec, log_spec.max() - 8.0)
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log_spec = (log_spec + 4.0) / 4.0
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mx.set_default_device(device)
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return log_spec
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