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Running
on
Zero
import argparse | |
import os | |
import traceback | |
import warnings | |
from typing import TYPE_CHECKING, List, Optional, Tuple, Union | |
import numpy as np | |
import torch | |
import tqdm | |
from .audio import ( | |
FRAMES_PER_SECOND, | |
HOP_LENGTH, | |
N_FRAMES, | |
N_SAMPLES, | |
SAMPLE_RATE, | |
log_mel_spectrogram, | |
pad_or_trim, | |
) | |
from .decoding import DecodingOptions, DecodingResult | |
from .timing import add_word_timestamps | |
from .tokenizer import LANGUAGES, TO_LANGUAGE_CODE, get_tokenizer | |
from .utils import ( | |
exact_div, | |
format_timestamp, | |
get_end, | |
get_writer, | |
make_safe, | |
optional_float, | |
optional_int, | |
str2bool, | |
) | |
if TYPE_CHECKING: | |
from .model import Whisper | |
def transcribe( | |
model: "Whisper", | |
audio: Union[str, np.ndarray, torch.Tensor], | |
*, | |
verbose: Optional[bool] = None, | |
temperature: Union[float, Tuple[float, ...]] = (0.0, 0.2, 0.4, 0.6, 0.8, 1.0), | |
compression_ratio_threshold: Optional[float] = 2.4, | |
logprob_threshold: Optional[float] = -1.0, | |
no_speech_threshold: Optional[float] = 0.6, | |
condition_on_previous_text: bool = True, | |
initial_prompt: Optional[str] = None, | |
word_timestamps: bool = False, | |
prepend_punctuations: str = "\"'“¿([{-", | |
append_punctuations: str = "\"'.。,,!!??::”)]}、", | |
clip_timestamps: Union[str, List[float]] = "0", | |
hallucination_silence_threshold: Optional[float] = None, | |
**decode_options, | |
): | |
""" | |
Transcribe an audio file using Whisper | |
Parameters | |
---------- | |
model: Whisper | |
The Whisper model instance | |
audio: Union[str, np.ndarray, torch.Tensor] | |
The path to the audio file to open, or the audio waveform | |
verbose: bool | |
Whether to display the text being decoded to the console. If True, displays all the details, | |
If False, displays minimal details. If None, does not display anything | |
temperature: Union[float, Tuple[float, ...]] | |
Temperature for sampling. It can be a tuple of temperatures, which will be successively used | |
upon failures according to either `compression_ratio_threshold` or `logprob_threshold`. | |
compression_ratio_threshold: float | |
If the gzip compression ratio is above this value, treat as failed | |
logprob_threshold: float | |
If the average log probability over sampled tokens is below this value, treat as failed | |
no_speech_threshold: float | |
If the no_speech probability is higher than this value AND the average log probability | |
over sampled tokens is below `logprob_threshold`, consider the segment as silent | |
condition_on_previous_text: bool | |
if True, the previous output of the model is provided as a prompt for the next window; | |
disabling may make the text inconsistent across windows, but the model becomes less prone to | |
getting stuck in a failure loop, such as repetition looping or timestamps going out of sync. | |
word_timestamps: bool | |
Extract word-level timestamps using the cross-attention pattern and dynamic time warping, | |
and include the timestamps for each word in each segment. | |
prepend_punctuations: str | |
If word_timestamps is True, merge these punctuation symbols with the next word | |
append_punctuations: str | |
If word_timestamps is True, merge these punctuation symbols with the previous word | |
initial_prompt: Optional[str] | |
Optional text to provide as a prompt for the first window. This can be used to provide, or | |
"prompt-engineer" a context for transcription, e.g. custom vocabularies or proper nouns | |
to make it more likely to predict those word correctly. | |
decode_options: dict | |
Keyword arguments to construct `DecodingOptions` instances | |
clip_timestamps: Union[str, List[float]] | |
Comma-separated list start,end,start,end,... timestamps (in seconds) of clips to process. | |
The last end timestamp defaults to the end of the file. | |
hallucination_silence_threshold: Optional[float] | |
When word_timestamps is True, skip silent periods longer than this threshold (in seconds) | |
when a possible hallucination is detected | |
Returns | |
------- | |
A dictionary containing the resulting text ("text") and segment-level details ("segments"), and | |
the spoken language ("language"), which is detected when `decode_options["language"]` is None. | |
""" | |
dtype = torch.float16 if decode_options.get("fp16", True) else torch.float32 | |
if model.device == torch.device("cpu"): | |
if torch.cuda.is_available(): | |
warnings.warn("Performing inference on CPU when CUDA is available") | |
if dtype == torch.float16: | |
warnings.warn("FP16 is not supported on CPU; using FP32 instead") | |
dtype = torch.float32 | |
if dtype == torch.float32: | |
decode_options["fp16"] = False | |
# Pad 30-seconds of silence to the input audio, for slicing | |
mel = log_mel_spectrogram(audio, model.dims.n_mels, padding=N_SAMPLES) | |
content_frames = mel.shape[-1] - N_FRAMES | |
content_duration = float(content_frames * HOP_LENGTH / SAMPLE_RATE) | |
if decode_options.get("language", None) is None: | |
if not model.is_multilingual: | |
decode_options["language"] = "en" | |
else: | |
if verbose: | |
print( | |
"Detecting language using up to the first 30 seconds. Use `--language` to specify the language" | |
) | |
mel_segment = pad_or_trim(mel, N_FRAMES).to(model.device).to(dtype) | |
_, probs = model.detect_language(mel_segment) | |
decode_options["language"] = max(probs, key=probs.get) | |
if verbose is not None: | |
print( | |
f"Detected language: {LANGUAGES[decode_options['language']].title()}" | |
) | |
language: str = decode_options["language"] | |
task: str = decode_options.get("task", "transcribe") | |
tokenizer = get_tokenizer( | |
model.is_multilingual, | |
num_languages=model.num_languages, | |
language=language, | |
task=task, | |
) | |
if isinstance(clip_timestamps, str): | |
clip_timestamps = [ | |
float(ts) for ts in (clip_timestamps.split(",") if clip_timestamps else []) | |
] | |
seek_points: List[int] = [round(ts * FRAMES_PER_SECOND) for ts in clip_timestamps] | |
if len(seek_points) == 0: | |
seek_points.append(0) | |
if len(seek_points) % 2 == 1: | |
seek_points.append(content_frames) | |
seek_clips: List[Tuple[int, int]] = list(zip(seek_points[::2], seek_points[1::2])) | |
punctuation = "\"'“¿([{-\"'.。,,!!??::”)]}、" | |
if word_timestamps and task == "translate": | |
warnings.warn("Word-level timestamps on translations may not be reliable.") | |
def decode_with_fallback(segment: torch.Tensor) -> DecodingResult: | |
temperatures = ( | |
[temperature] if isinstance(temperature, (int, float)) else temperature | |
) | |
decode_result = None | |
for t in temperatures: | |
kwargs = {**decode_options} | |
if t > 0: | |
# disable beam_size and patience when t > 0 | |
kwargs.pop("beam_size", None) | |
kwargs.pop("patience", None) | |
else: | |
# disable best_of when t == 0 | |
kwargs.pop("best_of", None) | |
options = DecodingOptions(**kwargs, temperature=t) | |
decode_result = model.decode(segment, options) | |
needs_fallback = False | |
if ( | |
compression_ratio_threshold is not None | |
and decode_result.compression_ratio > compression_ratio_threshold | |
): | |
needs_fallback = True # too repetitive | |
if ( | |
logprob_threshold is not None | |
and decode_result.avg_logprob < logprob_threshold | |
): | |
needs_fallback = True # average log probability is too low | |
if ( | |
no_speech_threshold is not None | |
and decode_result.no_speech_prob > no_speech_threshold | |
): | |
needs_fallback = False # silence | |
if not needs_fallback: | |
break | |
return decode_result | |
clip_idx = 0 | |
seek = seek_clips[clip_idx][0] | |
input_stride = exact_div( | |
N_FRAMES, model.dims.n_audio_ctx | |
) # mel frames per output token: 2 | |
time_precision = ( | |
input_stride * HOP_LENGTH / SAMPLE_RATE | |
) # time per output token: 0.02 (seconds) | |
all_tokens = [] | |
all_segments = [] | |
prompt_reset_since = 0 | |
if initial_prompt is not None: | |
initial_prompt_tokens = tokenizer.encode(" " + initial_prompt.strip()) | |
all_tokens.extend(initial_prompt_tokens) | |
else: | |
initial_prompt_tokens = [] | |
def new_segment( | |
*, start: float, end: float, tokens: torch.Tensor, result: DecodingResult | |
): | |
tokens = tokens.tolist() | |
text_tokens = [token for token in tokens if token < tokenizer.eot] | |
return { | |
"seek": seek, | |
"start": start, | |
"end": end, | |
"text": tokenizer.decode(text_tokens), | |
"tokens": tokens, | |
"temperature": result.temperature, | |
"avg_logprob": result.avg_logprob, | |
"compression_ratio": result.compression_ratio, | |
"no_speech_prob": result.no_speech_prob, | |
} | |
# show the progress bar when verbose is False (if True, transcribed text will be printed) | |
with tqdm.tqdm( | |
total=content_frames, unit="frames", disable=verbose is not False | |
) as pbar: | |
last_speech_timestamp = 0.0 | |
# NOTE: This loop is obscurely flattened to make the diff readable. | |
# A later commit should turn this into a simpler nested loop. | |
# for seek_clip_start, seek_clip_end in seek_clips: | |
# while seek < seek_clip_end | |
while clip_idx < len(seek_clips): | |
seek_clip_start, seek_clip_end = seek_clips[clip_idx] | |
if seek < seek_clip_start: | |
seek = seek_clip_start | |
if seek >= seek_clip_end: | |
clip_idx += 1 | |
if clip_idx < len(seek_clips): | |
seek = seek_clips[clip_idx][0] | |
continue | |
time_offset = float(seek * HOP_LENGTH / SAMPLE_RATE) | |
window_end_time = float((seek + N_FRAMES) * HOP_LENGTH / SAMPLE_RATE) | |
segment_size = min(N_FRAMES, content_frames - seek, seek_clip_end - seek) | |
mel_segment = mel[:, seek : seek + segment_size] | |
segment_duration = segment_size * HOP_LENGTH / SAMPLE_RATE | |
mel_segment = pad_or_trim(mel_segment, N_FRAMES).to(model.device).to(dtype) | |
decode_options["prompt"] = all_tokens[prompt_reset_since:] | |
result: DecodingResult = decode_with_fallback(mel_segment) | |
tokens = torch.tensor(result.tokens) | |
if no_speech_threshold is not None: | |
# no voice activity check | |
should_skip = result.no_speech_prob > no_speech_threshold | |
if ( | |
logprob_threshold is not None | |
and result.avg_logprob > logprob_threshold | |
): | |
# don't skip if the logprob is high enough, despite the no_speech_prob | |
should_skip = False | |
if should_skip: | |
seek += segment_size # fast-forward to the next segment boundary | |
continue | |
previous_seek = seek | |
current_segments = [] | |
# anomalous words are very long/short/improbable | |
def word_anomaly_score(word: dict) -> float: | |
probability = word.get("probability", 0.0) | |
duration = word["end"] - word["start"] | |
score = 0.0 | |
if probability < 0.15: | |
score += 1.0 | |
if duration < 0.133: | |
score += (0.133 - duration) * 15 | |
if duration > 2.0: | |
score += duration - 2.0 | |
return score | |
def is_segment_anomaly(segment: Optional[dict]) -> bool: | |
if segment is None or not segment["words"]: | |
return False | |
words = [w for w in segment["words"] if w["word"] not in punctuation] | |
words = words[:8] | |
score = sum(word_anomaly_score(w) for w in words) | |
return score >= 3 or score + 0.01 >= len(words) | |
def next_words_segment(segments: List[dict]) -> Optional[dict]: | |
return next((s for s in segments if s["words"]), None) | |
timestamp_tokens: torch.Tensor = tokens.ge(tokenizer.timestamp_begin) | |
single_timestamp_ending = timestamp_tokens[-2:].tolist() == [False, True] | |
consecutive = torch.where(timestamp_tokens[:-1] & timestamp_tokens[1:])[0] | |
consecutive.add_(1) | |
if len(consecutive) > 0: | |
# if the output contains two consecutive timestamp tokens | |
slices = consecutive.tolist() | |
if single_timestamp_ending: | |
slices.append(len(tokens)) | |
last_slice = 0 | |
for current_slice in slices: | |
sliced_tokens = tokens[last_slice:current_slice] | |
start_timestamp_pos = ( | |
sliced_tokens[0].item() - tokenizer.timestamp_begin | |
) | |
end_timestamp_pos = ( | |
sliced_tokens[-1].item() - tokenizer.timestamp_begin | |
) | |
current_segments.append( | |
new_segment( | |
start=time_offset + start_timestamp_pos * time_precision, | |
end=time_offset + end_timestamp_pos * time_precision, | |
tokens=sliced_tokens, | |
result=result, | |
) | |
) | |
last_slice = current_slice | |
if single_timestamp_ending: | |
# single timestamp at the end means no speech after the last timestamp. | |
seek += segment_size | |
else: | |
# otherwise, ignore the unfinished segment and seek to the last timestamp | |
last_timestamp_pos = ( | |
tokens[last_slice - 1].item() - tokenizer.timestamp_begin | |
) | |
seek += last_timestamp_pos * input_stride | |
else: | |
duration = segment_duration | |
timestamps = tokens[timestamp_tokens.nonzero().flatten()] | |
if ( | |
len(timestamps) > 0 | |
and timestamps[-1].item() != tokenizer.timestamp_begin | |
): | |
# no consecutive timestamps but it has a timestamp; use the last one. | |
last_timestamp_pos = ( | |
timestamps[-1].item() - tokenizer.timestamp_begin | |
) | |
duration = last_timestamp_pos * time_precision | |
current_segments.append( | |
new_segment( | |
start=time_offset, | |
end=time_offset + duration, | |
tokens=tokens, | |
result=result, | |
) | |
) | |
seek += segment_size | |
if word_timestamps: | |
add_word_timestamps( | |
segments=current_segments, | |
model=model, | |
tokenizer=tokenizer, | |
mel=mel_segment, | |
num_frames=segment_size, | |
prepend_punctuations=prepend_punctuations, | |
append_punctuations=append_punctuations, | |
last_speech_timestamp=last_speech_timestamp, | |
) | |
if not single_timestamp_ending: | |
last_word_end = get_end(current_segments) | |
if last_word_end is not None and last_word_end > time_offset: | |
seek = round(last_word_end * FRAMES_PER_SECOND) | |
# skip silence before possible hallucinations | |
if hallucination_silence_threshold is not None: | |
threshold = hallucination_silence_threshold | |
if not single_timestamp_ending: | |
last_word_end = get_end(current_segments) | |
if last_word_end is not None and last_word_end > time_offset: | |
remaining_duration = window_end_time - last_word_end | |
if remaining_duration > threshold: | |
seek = round(last_word_end * FRAMES_PER_SECOND) | |
else: | |
seek = previous_seek + segment_size | |
# if first segment might be a hallucination, skip leading silence | |
first_segment = next_words_segment(current_segments) | |
if first_segment is not None and is_segment_anomaly(first_segment): | |
gap = first_segment["start"] - time_offset | |
if gap > threshold: | |
seek = previous_seek + round(gap * FRAMES_PER_SECOND) | |
continue | |
# skip silence before any possible hallucination that is surrounded | |
# by silence or more hallucinations | |
hal_last_end = last_speech_timestamp | |
for si in range(len(current_segments)): | |
segment = current_segments[si] | |
if not segment["words"]: | |
continue | |
if is_segment_anomaly(segment): | |
next_segment = next_words_segment( | |
current_segments[si + 1 :] | |
) | |
if next_segment is not None: | |
hal_next_start = next_segment["words"][0]["start"] | |
else: | |
hal_next_start = time_offset + segment_duration | |
silence_before = ( | |
segment["start"] - hal_last_end > threshold | |
or segment["start"] < threshold | |
or segment["start"] - time_offset < 2.0 | |
) | |
silence_after = ( | |
hal_next_start - segment["end"] > threshold | |
or is_segment_anomaly(next_segment) | |
or window_end_time - segment["end"] < 2.0 | |
) | |
if silence_before and silence_after: | |
seek = round( | |
max(time_offset + 1, segment["start"]) | |
* FRAMES_PER_SECOND | |
) | |
if content_duration - segment["end"] < threshold: | |
seek = content_frames | |
current_segments[si:] = [] | |
break | |
hal_last_end = segment["end"] | |
last_word_end = get_end(current_segments) | |
if last_word_end is not None: | |
last_speech_timestamp = last_word_end | |
if verbose: | |
for segment in current_segments: | |
start, end, text = segment["start"], segment["end"], segment["text"] | |
line = f"[{format_timestamp(start)} --> {format_timestamp(end)}] {text}" | |
print(make_safe(line)) | |
# if a segment is instantaneous or does not contain text, clear it | |
for i, segment in enumerate(current_segments): | |
if segment["start"] == segment["end"] or segment["text"].strip() == "": | |
segment["text"] = "" | |
segment["tokens"] = [] | |
segment["words"] = [] | |
all_segments.extend( | |
[ | |
{"id": i, **segment} | |
for i, segment in enumerate( | |
current_segments, start=len(all_segments) | |
) | |
] | |
) | |
all_tokens.extend( | |
[token for segment in current_segments for token in segment["tokens"]] | |
) | |
if not condition_on_previous_text or result.temperature > 0.5: | |
# do not feed the prompt tokens if a high temperature was used | |
prompt_reset_since = len(all_tokens) | |
# update progress bar | |
pbar.update(min(content_frames, seek) - previous_seek) | |
return dict( | |
text=tokenizer.decode(all_tokens[len(initial_prompt_tokens) :]), | |
segments=all_segments, | |
language=language, | |
) | |
def cli(): | |
from . import available_models | |
def valid_model_name(name): | |
if name in available_models() or os.path.exists(name): | |
return name | |
raise ValueError( | |
f"model should be one of {available_models()} or path to a model checkpoint" | |
) | |
# fmt: off | |
parser = argparse.ArgumentParser(formatter_class=argparse.ArgumentDefaultsHelpFormatter) | |
parser.add_argument("audio", nargs="+", type=str, help="audio file(s) to transcribe") | |
parser.add_argument("--model", default="turbo", type=valid_model_name, help="name of the Whisper model to use") | |
parser.add_argument("--model_dir", type=str, default=None, help="the path to save model files; uses ~/.cache/whisper by default") | |
parser.add_argument("--device", default="cuda" if torch.cuda.is_available() else "cpu", help="device to use for PyTorch inference") | |
parser.add_argument("--output_dir", "-o", type=str, default=".", help="directory to save the outputs") | |
parser.add_argument("--output_format", "-f", type=str, default="all", choices=["txt", "vtt", "srt", "tsv", "json", "all"], help="format of the output file; if not specified, all available formats will be produced") | |
parser.add_argument("--verbose", type=str2bool, default=True, help="whether to print out the progress and debug messages") | |
parser.add_argument("--task", type=str, default="transcribe", choices=["transcribe", "translate"], help="whether to perform X->X speech recognition ('transcribe') or X->English translation ('translate')") | |
parser.add_argument("--language", type=str, default=None, choices=sorted(LANGUAGES.keys()) + sorted([k.title() for k in TO_LANGUAGE_CODE.keys()]), help="language spoken in the audio, specify None to perform language detection") | |
parser.add_argument("--temperature", type=float, default=0, help="temperature to use for sampling") | |
parser.add_argument("--best_of", type=optional_int, default=5, help="number of candidates when sampling with non-zero temperature") | |
parser.add_argument("--beam_size", type=optional_int, default=5, help="number of beams in beam search, only applicable when temperature is zero") | |
parser.add_argument("--patience", type=float, default=None, help="optional patience value to use in beam decoding, as in https://arxiv.org/abs/2204.05424, the default (1.0) is equivalent to conventional beam search") | |
parser.add_argument("--length_penalty", type=float, default=None, help="optional token length penalty coefficient (alpha) as in https://arxiv.org/abs/1609.08144, uses simple length normalization by default") | |
parser.add_argument("--suppress_tokens", type=str, default="-1", help="comma-separated list of token ids to suppress during sampling; '-1' will suppress most special characters except common punctuations") | |
parser.add_argument("--initial_prompt", type=str, default=None, help="optional text to provide as a prompt for the first window.") | |
parser.add_argument("--condition_on_previous_text", type=str2bool, default=True, help="if True, provide the previous output of the model as a prompt for the next window; disabling may make the text inconsistent across windows, but the model becomes less prone to getting stuck in a failure loop") | |
parser.add_argument("--fp16", type=str2bool, default=True, help="whether to perform inference in fp16; True by default") | |
parser.add_argument("--temperature_increment_on_fallback", type=optional_float, default=0.2, help="temperature to increase when falling back when the decoding fails to meet either of the thresholds below") | |
parser.add_argument("--compression_ratio_threshold", type=optional_float, default=2.4, help="if the gzip compression ratio is higher than this value, treat the decoding as failed") | |
parser.add_argument("--logprob_threshold", type=optional_float, default=-1.0, help="if the average log probability is lower than this value, treat the decoding as failed") | |
parser.add_argument("--no_speech_threshold", type=optional_float, default=0.6, help="if the probability of the <|nospeech|> token is higher than this value AND the decoding has failed due to `logprob_threshold`, consider the segment as silence") | |
parser.add_argument("--word_timestamps", type=str2bool, default=False, help="(experimental) extract word-level timestamps and refine the results based on them") | |
parser.add_argument("--prepend_punctuations", type=str, default="\"\'“¿([{-", help="if word_timestamps is True, merge these punctuation symbols with the next word") | |
parser.add_argument("--append_punctuations", type=str, default="\"\'.。,,!!??::”)]}、", help="if word_timestamps is True, merge these punctuation symbols with the previous word") | |
parser.add_argument("--highlight_words", type=str2bool, default=False, help="(requires --word_timestamps True) underline each word as it is spoken in srt and vtt") | |
parser.add_argument("--max_line_width", type=optional_int, default=None, help="(requires --word_timestamps True) the maximum number of characters in a line before breaking the line") | |
parser.add_argument("--max_line_count", type=optional_int, default=None, help="(requires --word_timestamps True) the maximum number of lines in a segment") | |
parser.add_argument("--max_words_per_line", type=optional_int, default=None, help="(requires --word_timestamps True, no effect with --max_line_width) the maximum number of words in a segment") | |
parser.add_argument("--threads", type=optional_int, default=0, help="number of threads used by torch for CPU inference; supercedes MKL_NUM_THREADS/OMP_NUM_THREADS") | |
parser.add_argument("--clip_timestamps", type=str, default="0", help="comma-separated list start,end,start,end,... timestamps (in seconds) of clips to process, where the last end timestamp defaults to the end of the file") | |
parser.add_argument("--hallucination_silence_threshold", type=optional_float, help="(requires --word_timestamps True) skip silent periods longer than this threshold (in seconds) when a possible hallucination is detected") | |
# fmt: on | |
args = parser.parse_args().__dict__ | |
model_name: str = args.pop("model") | |
model_dir: str = args.pop("model_dir") | |
output_dir: str = args.pop("output_dir") | |
output_format: str = args.pop("output_format") | |
device: str = args.pop("device") | |
os.makedirs(output_dir, exist_ok=True) | |
if model_name.endswith(".en") and args["language"] not in {"en", "English"}: | |
if args["language"] is not None: | |
warnings.warn( | |
f"{model_name} is an English-only model but receipted '{args['language']}'; using English instead." | |
) | |
args["language"] = "en" | |
temperature = args.pop("temperature") | |
if (increment := args.pop("temperature_increment_on_fallback")) is not None: | |
temperature = tuple(np.arange(temperature, 1.0 + 1e-6, increment)) | |
else: | |
temperature = [temperature] | |
if (threads := args.pop("threads")) > 0: | |
torch.set_num_threads(threads) | |
from . import load_model | |
model = load_model(model_name, device=device, download_root=model_dir) | |
writer = get_writer(output_format, output_dir) | |
word_options = [ | |
"highlight_words", | |
"max_line_count", | |
"max_line_width", | |
"max_words_per_line", | |
] | |
if not args["word_timestamps"]: | |
for option in word_options: | |
if args[option]: | |
parser.error(f"--{option} requires --word_timestamps True") | |
if args["max_line_count"] and not args["max_line_width"]: | |
warnings.warn("--max_line_count has no effect without --max_line_width") | |
if args["max_words_per_line"] and args["max_line_width"]: | |
warnings.warn("--max_words_per_line has no effect with --max_line_width") | |
writer_args = {arg: args.pop(arg) for arg in word_options} | |
for audio_path in args.pop("audio"): | |
try: | |
result = transcribe(model, audio_path, temperature=temperature, **args) | |
writer(result, audio_path, **writer_args) | |
except Exception as e: | |
traceback.print_exc() | |
print(f"Skipping {audio_path} due to {type(e).__name__}: {str(e)}") | |
if __name__ == "__main__": | |
cli() | |