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whisperX/whisperx/asr.py

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import warnings
from typing import TYPE_CHECKING, Optional, Tuple, Union
import numpy as np
import torch
import tqdm
import ffmpeg
from whisper.audio import (
FRAMES_PER_SECOND,
HOP_LENGTH,
N_FRAMES,
N_SAMPLES,
SAMPLE_RATE,
CHUNK_LENGTH,
log_mel_spectrogram,
pad_or_trim,
load_audio
)
from whisper.decoding import DecodingOptions, DecodingResult
from whisper.timing import add_word_timestamps
from whisper.tokenizer import LANGUAGES, TO_LANGUAGE_CODE, get_tokenizer
from whisper.utils import (
exact_div,
format_timestamp,
make_safe,
)
if TYPE_CHECKING:
from whisper.model import Whisper
from .vad import merge_chunks
def transcribe(
model: "Whisper",
audio: Union[str, np.ndarray, torch.Tensor] = None,
mel: np.ndarray = None,
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 = "\"'.。,!?::”)]}、",
**decode_options,
):
"""
Transcribe an audio file using Whisper.
We redefine the Whisper transcribe function to allow mel input (for sequential slicing of audio)
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
mel: np.ndarray
Mel spectrogram of audio segment.
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
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
if mel is None:
if audio is None:
raise ValueError("Transcribe needs either audio or mel as input, currently both are none.")
mel = log_mel_spectrogram(audio, padding=N_SAMPLES)
content_frames = mel.shape[-1] - N_FRAMES
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, language=language, task=task)
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 not needs_fallback:
break
return decode_result
seek = 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:
while seek < content_frames:
time_offset = float(seek * HOP_LENGTH / SAMPLE_RATE)
mel_segment = mel[:, seek : seek + N_FRAMES]
segment_size = min(N_FRAMES, content_frames - seek)
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 = []
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 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)
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,
)
word_end_timestamps = [
w["end"] for s in current_segments for w in s["words"]
]
if not single_timestamp_ending and len(word_end_timestamps) > 0:
seek_shift = round(
(word_end_timestamps[-1] - time_offset) * FRAMES_PER_SECOND
)
if seek_shift > 0:
seek = previous_seek + seek_shift
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"]]
)
# 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 transcribe_with_vad(
model: "Whisper",
audio: str,
vad_pipeline,
mel = None,
verbose: Optional[bool] = None,
**kwargs
):
"""
Transcribe per VAD segment
"""
vad_segments = vad_pipeline(audio)
# if not torch.is_tensor(audio):
# if isinstance(audio, str):
audio = load_audio(audio)
audio = torch.from_numpy(audio)
prev = 0
output = {"segments": []}
# merge segments to approx 30s inputs to make whisper most appropraite
vad_segments = merge_chunks(vad_segments, chunk_size=CHUNK_LENGTH)
if len(vad_segments) == 0:
return output
print(">>Performing transcription...")
for sdx, seg_t in enumerate(vad_segments):
if verbose:
print(f"~~ Transcribing VAD chunk: ({format_timestamp(seg_t['start'])} --> {format_timestamp(seg_t['end'])}) ~~")
seg_f_start, seg_f_end = int(seg_t["start"] * SAMPLE_RATE), int(seg_t["end"] * SAMPLE_RATE)
local_f_start, local_f_end = seg_f_start - prev, seg_f_end - prev
audio = audio[local_f_start:] # seek forward
seg_audio = audio[:local_f_end-local_f_start] # seek forward
prev = seg_f_start
local_mel = log_mel_spectrogram(seg_audio, padding=N_SAMPLES)
# need to pad
result = transcribe(model, audio, mel=local_mel, verbose=verbose, **kwargs)
seg_t["text"] = result["text"]
output["segments"].append(
{
"start": seg_t["start"],
"end": seg_t["end"],
"language": result["language"],
"text": result["text"],
"seg-text": [x["text"] for x in result["segments"]],
"seg-start": [x["start"] for x in result["segments"]],
"seg-end": [x["end"] for x in result["segments"]],
}
)
output["language"] = output["segments"][0]["language"]
return output