Merge pull request #108 from kadirnar/add_lightning_mlx

🍎 Add Lightning Mlx library

whisperplus/pipelines/lightning_whisper_mlx/__init__.py

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+from .lightning import LightningWhisperMLX

whisperplus/pipelines/lightning_whisper_mlx/audio.py

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