Merge 8d6e0e5e1af80e81f3ef1d0122a58473afdabdcd into 173ff7dd1d9fb1c4fddea0d41d704cfefeb8908c

whisper/audio.py

@@ -20,39 +20,48 @@ N_FRAMES = exact_div(N_SAMPLES, HOP_LENGTH)  # 3000 frames in a mel spectrogram
 N_SAMPLES_PER_TOKEN = HOP_LENGTH * 2  # the initial convolutions has stride 2
 FRAMES_PER_SECOND = exact_div(SAMPLE_RATE, HOP_LENGTH)  # 10ms per audio frame
 TOKENS_PER_SECOND = exact_div(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
+    Loads an audio file as a mono waveform, resampling to the specified sample rate.
 
     Parameters
     ----------
     file : str
-        The audio file to open
+        Path to the audio file.
-
+    sr : int, optional
-    sr: int
+        Target sample rate for resampling, defaults to SAMPLE_RATE.
-        The sample rate to resample the audio if necessary
 
     Returns
     -------
-    A NumPy array containing the audio waveform, in float32 dtype.
+    np.ndarray
+        1D NumPy array of the audio waveform, normalized between -1 and 1.
+
+    Raises
+    ------
+    RuntimeError
+        If the audio cannot be loaded.
+
+    Notes
+    -----
+    Requires ffmpeg installed and accessible in the system's PATH.
     """
 
     # This launches a subprocess to decode audio while down-mixing
     # and resampling as necessary.  Requires the ffmpeg CLI in PATH.
     # fmt: off
+    
     cmd = [
-        "ffmpeg",
+    "ffmpeg",           # Command to run the ffmpeg tool.
-        "-nostdin",
+    "-nostdin",         # Prevents ffmpeg from reading from stdin.
-        "-threads", "0",
+    "-threads", "0",    # Uses all available CPU cores for processing.
-        "-i", file,
+    "-i", file,         # Specifies the input file path.
-        "-f", "s16le",
+    "-f", "s16le",      # Sets the output format to 16-bit PCM.
-        "-ac", "1",
+    "-ac", "1",         # Converts audio to mono (1 channel).
-        "-acodec", "pcm_s16le",
+    "-acodec", "pcm_s16le",  # Specifies the audio codec as PCM signed 16-bit little-endian.
-        "-ar", str(sr),
+    "-ar", str(sr),     # Resamples the audio to the specified sample rate.
-        "-"
+    "-"                 # Outputs the processed audio to stdout.
 ]
+
     # fmt: on
     try:
         out = run(cmd, capture_output=True, check=True).stdout
@@ -63,8 +72,22 @@ def load_audio(file: str, sr: int = SAMPLE_RATE):
 
 
 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.
+    Pads or trims the input array to a specified length along the given axis.
+
+    Parameters:
+    - array: Input array (torch.Tensor or np.ndarray).
+    - length: Target length along the specified axis (default is N_SAMPLES).
+    - axis: Axis to pad or trim (default is -1 for the last axis).
+
+    Returns:
+    - The modified array, either padded with zeros or trimmed to the target length.
+    
+     Note:
+    - The function handles both PyTorch tensors and NumPy arrays, applying appropriate methods 
+      for padding and trimming depending on the array type.
+    
     """
     if torch.is_tensor(array):
         if array.shape[axis] > length:
@@ -91,14 +114,30 @@ def pad_or_trim(array, length: int = N_SAMPLES, *, axis: int = -1):
 @lru_cache(maxsize=None)
 def mel_filters(device, n_mels: int) -> torch.Tensor:
     """
-    load the mel filterbank matrix for projecting STFT into a Mel spectrogram.
+    Loads a precomputed Mel filterbank matrix for converting STFT to a Mel spectrogram.
-    Allows decoupling librosa dependency; saved using:
+
+    Parameters
+    ----------
+    device : torch.device
+        The device (CPU or GPU) to load the tensor onto.
+    n_mels : int
+        The number of Mel bands, must be either 80 or 128.
+
+    Returns
+    -------
+    torch.Tensor
+        A tensor containing the Mel filterbank matrix.
+
+    Raises
+    ------
+    AssertionError
+        If `n_mels` is not supported.
+
+    Notes
+    -----
+    The Mel filterbank matrices are saved in a compressed npz file, which decouples 
+    the dependency on librosa for generating these filters.
     
-        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}"
 
@@ -114,27 +153,32 @@ def log_mel_spectrogram(
     device: Optional[Union[str, torch.device]] = None,
 ):
     """
-    Compute the log-Mel spectrogram of
+    Computes the log-Mel spectrogram of an audio waveform.
 
     Parameters
     ----------
-    audio: Union[str, np.ndarray, torch.Tensor], shape = (*)
+    audio : Union[str, np.ndarray, torch.Tensor]
-        The path to audio or either a NumPy array or Tensor containing the audio waveform in 16 kHz
+        The audio input, either as a file path, NumPy array, or Torch tensor. 
-
+        The waveform should be in 16 kHz.
-    n_mels: int
+    n_mels : int, optional
-        The number of Mel-frequency filters, only 80 is supported
+        The number of Mel-frequency filters, only 80 is supported. Defaults to 80.
-
+    padding : int, optional
-    padding: int
+        Number of zero samples to pad at the end of the audio. Defaults to 0.
-        Number of zero samples to pad to the right
+    device : Optional[Union[str, torch.device]], optional
-
+        The device to perform computations on. If provided, the audio tensor is moved 
-    device: Optional[Union[str, torch.device]]
+        to this device. Defaults to None.
-        If given, the audio tensor is moved to this device before STFT
 
     Returns
     -------
-    torch.Tensor, shape = (80, n_frames)
+    torch.Tensor
-        A Tensor that contains the Mel spectrogram
+        A tensor containing the Mel spectrogram with shape (80, n_frames).
+
+    Notes
+    -----
+    The function expects a 16 kHz sampling rate for the input audio and uses a Hann 
+    window for the Short-Time Fourier Transform (STFT).
     """
+    
     if not torch.is_tensor(audio):
         if isinstance(audio, str):
             audio = load_audio(audio)