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transcribe.py
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transcribe.py
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import itertools
import json
import logging
import os
import zlib
from dataclasses import asdict, dataclass
from inspect import signature
from math import ceil
from typing import BinaryIO, Iterable, List, Optional, Tuple, Union
from warnings import warn
import ctranslate2
import numpy as np
import tokenizers
from tqdm import tqdm
from faster_whisper.audio import decode_audio, pad_or_trim
from faster_whisper.feature_extractor import FeatureExtractor
from faster_whisper.tokenizer import _LANGUAGE_CODES, Tokenizer
from faster_whisper.utils import download_model, format_timestamp, get_end, get_logger
from faster_whisper.vad import (
SpeechTimestampsMap,
VadOptions,
collect_chunks,
get_speech_timestamps,
merge_segments,
)
@dataclass
class Word:
start: float
end: float
word: str
probability: float
def _asdict(self):
warn(
"Word._asdict() method is deprecated, use dataclasses.asdict(Word) instead",
DeprecationWarning,
2,
)
return asdict(self)
@dataclass
class Segment:
id: int
seek: int
start: float
end: float
text: str
tokens: List[int]
avg_logprob: float
compression_ratio: float
no_speech_prob: float
words: Optional[List[Word]]
temperature: Optional[float]
def _asdict(self):
warn(
"Segment._asdict() method is deprecated, use dataclasses.asdict(Segment) instead",
DeprecationWarning,
2,
)
return asdict(self)
@dataclass
class TranscriptionOptions:
beam_size: int
best_of: int
patience: float
length_penalty: float
repetition_penalty: float
no_repeat_ngram_size: int
log_prob_threshold: Optional[float]
log_prob_low_threshold: Optional[float]
no_speech_threshold: Optional[float]
compression_ratio_threshold: Optional[float]
condition_on_previous_text: bool
prompt_reset_on_temperature: float
temperatures: List[float]
initial_prompt: Optional[Union[str, Iterable[int]]]
prefix: Optional[str]
suppress_blank: bool
suppress_tokens: Optional[List[int]]
without_timestamps: bool
max_initial_timestamp: float
word_timestamps: bool
prepend_punctuations: str
append_punctuations: str
multilingual: bool
output_language: Optional[str]
max_new_tokens: Optional[int]
clip_timestamps: Union[str, List[float]]
hallucination_silence_threshold: Optional[float]
hotwords: Optional[str]
@dataclass
class TranscriptionInfo:
language: str
language_probability: float
duration: float
duration_after_vad: float
all_language_probs: Optional[List[Tuple[str, float]]]
transcription_options: TranscriptionOptions
vad_options: VadOptions
class BatchedInferencePipeline:
def __init__(
self,
model,
):
self.model: WhisperModel = model
self.last_speech_timestamp = 0.0
def forward(self, features, tokenizer, chunks_metadata, options):
encoder_output, outputs = self.generate_segment_batched(
features, tokenizer, options
)
segmented_outputs = []
segment_sizes = []
for chunk_metadata, output in zip(chunks_metadata, outputs):
duration = chunk_metadata["end_time"] - chunk_metadata["start_time"]
segment_size = int(ceil(duration) * self.model.frames_per_second)
segment_sizes.append(segment_size)
(
subsegments,
seek,
single_timestamp_ending,
) = self.model._split_segments_by_timestamps(
tokenizer=tokenizer,
tokens=output["tokens"],
time_offset=chunk_metadata["start_time"],
segment_size=segment_size,
segment_duration=duration,
seek=0,
)
segmented_outputs.append(
[
dict(
text=tokenizer.decode(subsegment["tokens"]),
avg_logprob=output["avg_logprob"],
no_speech_prob=output["no_speech_prob"],
tokens=subsegment["tokens"],
start=subsegment["start"],
end=subsegment["end"],
compression_ratio=get_compression_ratio(
tokenizer.decode(subsegment["tokens"])
),
seek=int(
chunk_metadata["start_time"] * self.model.frames_per_second
),
)
for subsegment in subsegments
]
)
if options.word_timestamps:
self.last_speech_timestamp = self.model.add_word_timestamps(
segmented_outputs,
tokenizer,
encoder_output,
segment_sizes,
options.prepend_punctuations,
options.append_punctuations,
self.last_speech_timestamp,
)
return segmented_outputs
def generate_segment_batched(
self,
features: np.ndarray,
tokenizer: Tokenizer,
options: TranscriptionOptions,
):
batch_size = features.shape[0]
prompt = self.model.get_prompt(
tokenizer,
previous_tokens=(
tokenizer.encode(options.initial_prompt)
if options.initial_prompt is not None
else []
),
without_timestamps=options.without_timestamps,
hotwords=options.hotwords,
)
if options.max_new_tokens is not None:
max_length = len(prompt) + options.max_new_tokens
else:
max_length = self.model.max_length
if max_length > self.model.max_length:
raise ValueError(
f"The length of the prompt is {len(prompt)}, and the `max_new_tokens` "
f"{max_length - len(prompt)}. Thus, the combined length of the prompt "
f"and `max_new_tokens` is: {max_length}. This exceeds the "
f"`max_length` of the Whisper model: {self.model.max_length}. "
"You should either reduce the length of your prompt, or "
"reduce the value of `max_new_tokens`, "
f"so that their combined length is less that {self.model.max_length}."
)
encoder_output = self.model.encode(features)
results = self.model.model.generate(
encoder_output,
[prompt] * batch_size,
beam_size=options.beam_size,
patience=options.patience,
length_penalty=options.length_penalty,
max_length=max_length,
suppress_blank=options.suppress_blank,
suppress_tokens=options.suppress_tokens,
return_scores=True,
return_no_speech_prob=True,
sampling_temperature=options.temperatures[0],
repetition_penalty=options.repetition_penalty,
no_repeat_ngram_size=options.no_repeat_ngram_size,
)
output = []
for result in results:
# return scores
seq_len = len(result.sequences_ids[0])
cum_logprob = result.scores[0] * (seq_len**options.length_penalty)
output.append(
dict(
avg_logprob=cum_logprob / (seq_len + 1),
no_speech_prob=result.no_speech_prob,
tokens=result.sequences_ids[0],
)
)
return encoder_output, output
def transcribe(
self,
audio: Union[str, BinaryIO, np.ndarray],
language: Optional[str] = None,
task: str = "transcribe",
log_progress: bool = False,
beam_size: int = 5,
best_of: int = 5,
patience: float = 1,
length_penalty: float = 1,
repetition_penalty: float = 1,
no_repeat_ngram_size: int = 0,
temperature: Union[float, List[float], Tuple[float, ...]] = [
0.0,
0.2,
0.4,
0.6,
0.8,
1.0,
],
compression_ratio_threshold: Optional[float] = 2.4,
log_prob_threshold: Optional[float] = -1.0,
log_prob_low_threshold: Optional[float] = None,
no_speech_threshold: Optional[float] = 0.6,
condition_on_previous_text: bool = True,
prompt_reset_on_temperature: float = 0.5,
initial_prompt: Optional[Union[str, Iterable[int]]] = None,
prefix: Optional[str] = None,
suppress_blank: bool = True,
suppress_tokens: Optional[List[int]] = [-1],
without_timestamps: bool = True,
max_initial_timestamp: float = 1.0,
word_timestamps: bool = False,
prepend_punctuations: str = "\"'“¿([{-",
append_punctuations: str = "\"'.。,,!!??::”)]}、",
multilingual: bool = False,
output_language: Optional[str] = None,
vad_filter: bool = True,
vad_parameters: Optional[Union[dict, VadOptions]] = None,
max_new_tokens: Optional[int] = None,
chunk_length: Optional[int] = None,
clip_timestamps: Optional[List[dict]] = None,
hallucination_silence_threshold: Optional[float] = None,
batch_size: int = 8,
hotwords: Optional[str] = None,
language_detection_threshold: Optional[float] = 0.5,
language_detection_segments: int = 1,
) -> Tuple[Iterable[Segment], TranscriptionInfo]:
"""transcribe audio in chunks in batched fashion and return with language info.
Arguments:
audio: Path to the input file (or a file-like object), or the audio waveform.
language: The language spoken in the audio. It should be a language code such
as "en" or "fr". If not set, the language will be detected in the first 30 seconds
of audio.
task: Task to execute (transcribe or translate).
log_progress: whether to show progress bar or not.
beam_size: Beam size to use for decoding.
best_of: Number of candidates when sampling with non-zero temperature.
patience: Beam search patience factor.
length_penalty: Exponential length penalty constant.
repetition_penalty: Penalty applied to the score of previously generated tokens
(set > 1 to penalize).
no_repeat_ngram_size: Prevent repetitions of ngrams with this size (set 0 to disable).
temperature: Temperature for sampling. If a list or tuple is passed,
only the first value is used.
initial_prompt: Optional text string or iterable of token ids to provide as a
prompt for the each window.
suppress_blank: Suppress blank outputs at the beginning of the sampling.
suppress_tokens: List of token IDs to suppress. -1 will suppress a default set
of symbols as defined in `tokenizer.non_speech_tokens()`.
without_timestamps: Only sample text tokens.
word_timestamps: Extract word-level timestamps using the cross-attention pattern
and dynamic time warping, and include the timestamps for each word in each segment.
Set as False.
prepend_punctuations: If word_timestamps is True, merge these punctuation symbols
with the next word
append_punctuations: If word_timestamps is True, merge these punctuation symbols
with the previous word
vad_filter: Enable the voice activity detection (VAD) to filter out parts of the audio
without speech. This step is using the Silero VAD model
https://github.com/snakers4/silero-vad.
vad_parameters: Dictionary of Silero VAD parameters or VadOptions class (see available
parameters and default values in the class `VadOptions`).
max_new_tokens: Maximum number of new tokens to generate per-chunk. If not set,
the maximum will be set by the default max_length.
chunk_length: The length of audio segments. If it is not None, it will overwrite the
default chunk_length of the FeatureExtractor.
clip_timestamps: Optionally provide list of dictionaries each containing "start" and
"end" keys that specify the start and end of the voiced region within
`chunk_length` boundary. vad_filter will be ignored if clip_timestamps is used.
batch_size: the maximum number of parallel requests to model for decoding.
hotwords:
Hotwords/hint phrases to the model. Has no effect if prefix is not None.
language_detection_threshold: If the maximum probability of the language tokens is
higher than this value, the language is detected.
language_detection_segments: Number of segments to consider for the language detection.
Unused Arguments
compression_ratio_threshold: If the gzip compression ratio is above this value,
treat as failed.
log_prob_threshold: If the average log probability over sampled tokens is
below this value, treat as failed.
log_prob_low_threshold: This parameter alone is sufficient to skip an output text,
whereas log_prob_threshold also looks for appropriate no_speech_threshold value.
This value should be less than log_prob_threshold.
no_speech_threshold: If the no_speech probability is higher than this value AND
the average log probability over sampled tokens is below `log_prob_threshold`,
consider the segment as silent.
condition_on_previous_text: 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. Set as False
prompt_reset_on_temperature: Resets prompt if temperature is above this value.
Arg has effect only if condition_on_previous_text is True. Set at 0.5
prefix: Optional text to provide as a prefix at the beginning of each window.
max_initial_timestamp: The initial timestamp cannot be later than this, set at 0.0.
multilingual: If True, perform transcription on multilingual videos. Set as False.
output_language: Valid only if multilingual is set to True.
Specifies the string representing the output language. One of
'en' (English) or 'hybrid' (code-switched transcription). set as None.
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. set as None.
Returns:
A tuple with:
- a generator over transcribed segments
- an instance of TranscriptionInfo
"""
sampling_rate = self.model.feature_extractor.sampling_rate
if not isinstance(audio, np.ndarray):
audio = decode_audio(audio, sampling_rate=sampling_rate)
duration = audio.shape[0] / sampling_rate
chunk_length = chunk_length or self.model.feature_extractor.chunk_length
# if no segment split is provided, use vad_model and generate segments
if not clip_timestamps:
if vad_filter:
if vad_parameters is None:
vad_parameters = VadOptions(
max_speech_duration_s=chunk_length,
min_silence_duration_ms=160,
)
elif isinstance(vad_parameters, dict):
if "max_speech_duration_s" in vad_parameters.keys():
vad_parameters.pop("max_speech_duration_s")
vad_parameters = VadOptions(
**vad_parameters, max_speech_duration_s=chunk_length
)
active_segments = get_speech_timestamps(audio, vad_parameters)
clip_timestamps = merge_segments(active_segments, vad_parameters)
# run the audio if it is less than 30 sec even without clip_timestamps
elif duration < chunk_length:
clip_timestamps = [{"start": 0, "end": audio.shape[0]}]
else:
raise RuntimeError(
"No clip timestamps found. "
"Set 'vad_filter' to True or provide 'clip_timestamps'."
)
duration_after_vad = (
sum((segment["end"] - segment["start"]) for segment in clip_timestamps)
/ sampling_rate
)
audio_chunks, chunks_metadata = collect_chunks(audio, clip_timestamps)
features = (
[self.model.feature_extractor(chunk)[..., :-1] for chunk in audio_chunks]
if duration_after_vad
else []
)
all_language_probs = None
# detecting the language if not provided
if language is None:
if not self.model.model.is_multilingual:
language = "en"
language_probability = 1
else:
(
language,
language_probability,
all_language_probs,
) = self.model.detect_language(
features=np.concatenate(
features
+ [
np.full((self.model.model.n_mels, 1), -1.5, dtype="float32")
],
axis=1,
), # add a dummy feature to account for empty audio
language_detection_segments=language_detection_segments,
language_detection_threshold=language_detection_threshold,
)
self.model.logger.info(
"Detected language '%s' with probability %.2f",
language,
language_probability,
)
else:
if not self.model.model.is_multilingual and language != "en":
self.model.logger.warning(
"The current model is English-only but the language parameter is set to '%s'; "
"using 'en' instead." % language
)
language = "en"
language_probability = 1
tokenizer = Tokenizer(
self.model.hf_tokenizer,
self.model.model.is_multilingual,
task=task,
language=language,
)
features = (
np.stack([pad_or_trim(feature) for feature in features]) if features else []
)
options = TranscriptionOptions(
beam_size=beam_size,
best_of=best_of,
patience=patience,
length_penalty=length_penalty,
repetition_penalty=repetition_penalty,
no_repeat_ngram_size=no_repeat_ngram_size,
log_prob_threshold=log_prob_threshold,
log_prob_low_threshold=log_prob_low_threshold,
no_speech_threshold=no_speech_threshold,
compression_ratio_threshold=compression_ratio_threshold,
temperatures=(
temperature[:1]
if isinstance(temperature, (list, tuple))
else [temperature]
),
initial_prompt=initial_prompt,
prefix=prefix,
suppress_blank=suppress_blank,
suppress_tokens=get_suppressed_tokens(tokenizer, suppress_tokens),
prepend_punctuations=prepend_punctuations,
append_punctuations=append_punctuations,
max_new_tokens=max_new_tokens,
hotwords=hotwords,
word_timestamps=word_timestamps,
hallucination_silence_threshold=None,
condition_on_previous_text=False,
clip_timestamps=clip_timestamps,
prompt_reset_on_temperature=0.5,
multilingual=False,
output_language=None,
without_timestamps=without_timestamps,
max_initial_timestamp=0.0,
)
info = TranscriptionInfo(
language=language,
language_probability=language_probability,
duration=duration,
duration_after_vad=duration_after_vad,
transcription_options=options,
vad_options=vad_parameters,
all_language_probs=all_language_probs,
)
segments = self._batched_segments_generator(
features,
tokenizer,
chunks_metadata,
batch_size,
options,
log_progress,
)
return segments, info
def _batched_segments_generator(
self, features, tokenizer, chunks_metadata, batch_size, options, log_progress
):
pbar = tqdm(total=len(features), disable=not log_progress, position=0)
seg_idx = 0
for i in range(0, len(features), batch_size):
results = self.forward(
features[i : i + batch_size],
tokenizer,
chunks_metadata[i : i + batch_size],
options,
)
for result in results:
for segment in result:
seg_idx += 1
yield Segment(
seek=segment["seek"],
id=seg_idx,
text=segment["text"],
start=round(segment["start"], 3),
end=round(segment["end"], 3),
words=(
None
if not options.word_timestamps
else [Word(**word) for word in segment["words"]]
),
tokens=segment["tokens"],
avg_logprob=segment["avg_logprob"],
no_speech_prob=segment["no_speech_prob"],
compression_ratio=segment["compression_ratio"],
temperature=options.temperatures[0],
)
pbar.update(1)
pbar.close()
self.last_speech_timestamp = 0.0
class WhisperModel:
def __init__(
self,
model_size_or_path: str,
device: str = "auto",
device_index: Union[int, List[int]] = 0,
compute_type: str = "default",
cpu_threads: int = 0,
num_workers: int = 1,
download_root: Optional[str] = None,
local_files_only: bool = False,
files: dict = None,
**model_kwargs,
):
"""Initializes the Whisper model.
Args:
model_size_or_path: Size of the model to use (tiny, tiny.en, base, base.en,
small, small.en, distil-small.en, medium, medium.en, distil-medium.en, large-v1,
large-v2, large-v3, large, distil-large-v2, distil-large-v3, large-v3-turbo, or turbo),
a path to a converted model directory, or a CTranslate2-converted Whisper model ID from
the HF Hub. When a size or a model ID is configured, the converted model is downloaded
from the Hugging Face Hub.
device: Device to use for computation ("cpu", "cuda", "auto").
device_index: Device ID to use.
The model can also be loaded on multiple GPUs by passing a list of IDs
(e.g. [0, 1, 2, 3]). In that case, multiple transcriptions can run in parallel
when transcribe() is called from multiple Python threads (see also num_workers).
compute_type: Type to use for computation.
See https://opennmt.net/CTranslate2/quantization.html.
cpu_threads: Number of threads to use when running on CPU (4 by default).
A non zero value overrides the OMP_NUM_THREADS environment variable.
num_workers: When transcribe() is called from multiple Python threads,
having multiple workers enables true parallelism when running the model
(concurrent calls to self.model.generate() will run in parallel).
This can improve the global throughput at the cost of increased memory usage.
download_root: Directory where the models should be saved. If not set, the models
are saved in the standard Hugging Face cache directory.
local_files_only: If True, avoid downloading the file and return the path to the
local cached file if it exists.
files: Load model files from the memory. This argument is a dictionary mapping file names
to file contents as file-like or bytes objects. If this is set, model_path acts as an
identifier for this model.
"""
self.logger = get_logger()
tokenizer_bytes, preprocessor_bytes = None, None
if files:
model_path = model_size_or_path
tokenizer_bytes = files.pop("tokenizer.json", None)
preprocessor_bytes = files.pop("preprocessor_config.json", None)
elif os.path.isdir(model_size_or_path):
model_path = model_size_or_path
else:
model_path = download_model(
model_size_or_path,
local_files_only=local_files_only,
cache_dir=download_root,
)
self.device = device
# set the random seed to make sure consistency across runs
ctranslate2.set_random_seed(42)
self.model = ctranslate2.models.Whisper(
model_path,
device=self.device,
device_index=device_index,
compute_type=compute_type,
intra_threads=cpu_threads,
inter_threads=num_workers,
files=files,
**model_kwargs,
)
tokenizer_file = os.path.join(model_path, "tokenizer.json")
if tokenizer_bytes:
self.hf_tokenizer = tokenizers.Tokenizer.from_buffer(tokenizer_bytes)
elif os.path.isfile(tokenizer_file):
self.hf_tokenizer = tokenizers.Tokenizer.from_file(tokenizer_file)
else:
self.hf_tokenizer = tokenizers.Tokenizer.from_pretrained(
"openai/whisper-tiny" + ("" if self.model.is_multilingual else ".en")
)
self.feat_kwargs = self._get_feature_kwargs(model_path, preprocessor_bytes)
self.feature_extractor = FeatureExtractor(**self.feat_kwargs)
self.input_stride = 2
self.num_samples_per_token = (
self.feature_extractor.hop_length * self.input_stride
)
self.frames_per_second = (
self.feature_extractor.sampling_rate // self.feature_extractor.hop_length
)
self.tokens_per_second = (
self.feature_extractor.sampling_rate // self.num_samples_per_token
)
self.time_precision = 0.02
self.max_length = 448
@property
def supported_languages(self) -> List[str]:
"""The languages supported by the model."""
return list(_LANGUAGE_CODES) if self.model.is_multilingual else ["en"]
def _get_feature_kwargs(self, model_path, preprocessor_bytes=None) -> dict:
config = {}
try:
config_path = os.path.join(model_path, "preprocessor_config.json")
if preprocessor_bytes:
config = json.loads(preprocessor_bytes)
elif os.path.isfile(config_path):
with open(config_path, "r", encoding="utf-8") as file:
config = json.load(file)
else:
return config
valid_keys = signature(FeatureExtractor.__init__).parameters.keys()
return {k: v for k, v in config.items() if k in valid_keys}
except json.JSONDecodeError as e:
self.logger.warning("Could not load preprocessor config: %s", e)
return config
def transcribe(
self,
audio: Union[str, BinaryIO, np.ndarray],
language: Optional[str] = None,
task: str = "transcribe",
log_progress: bool = False,
beam_size: int = 5,
best_of: int = 5,
patience: float = 1,
length_penalty: float = 1,
repetition_penalty: float = 1,
no_repeat_ngram_size: int = 0,
temperature: Union[float, List[float], Tuple[float, ...]] = [
0.0,
0.2,
0.4,
0.6,
0.8,
1.0,
],
compression_ratio_threshold: Optional[float] = 2.4,
log_prob_threshold: Optional[float] = -1.0,
log_prob_low_threshold: Optional[float] = None,
no_speech_threshold: Optional[float] = 0.6,
condition_on_previous_text: bool = True,
prompt_reset_on_temperature: float = 0.5,
initial_prompt: Optional[Union[str, Iterable[int]]] = None,
prefix: Optional[str] = None,
suppress_blank: bool = True,
suppress_tokens: Optional[List[int]] = [-1],
without_timestamps: bool = False,
max_initial_timestamp: float = 1.0,
word_timestamps: bool = False,
prepend_punctuations: str = "\"'“¿([{-",
append_punctuations: str = "\"'.。,,!!??::”)]}、",
multilingual: bool = False,
output_language: Optional[str] = None,
vad_filter: bool = False,
vad_parameters: Optional[Union[dict, VadOptions]] = None,
max_new_tokens: Optional[int] = None,
chunk_length: Optional[int] = None,
clip_timestamps: Union[str, List[float]] = "0",
hallucination_silence_threshold: Optional[float] = None,
hotwords: Optional[str] = None,
language_detection_threshold: Optional[float] = 0.5,
language_detection_segments: int = 1,
) -> Tuple[Iterable[Segment], TranscriptionInfo]:
"""Transcribes an input file.
Arguments:
audio: Path to the input file (or a file-like object), or the audio waveform.
language: The language spoken in the audio. It should be a language code such
as "en" or "fr". If not set, the language will be detected in the first 30 seconds
of audio.
task: Task to execute (transcribe or translate).
log_progress: whether to show progress bar or not.
beam_size: Beam size to use for decoding.
best_of: Number of candidates when sampling with non-zero temperature.
patience: Beam search patience factor.
length_penalty: Exponential length penalty constant.
repetition_penalty: Penalty applied to the score of previously generated tokens
(set > 1 to penalize).
no_repeat_ngram_size: Prevent repetitions of ngrams with this size (set 0 to disable).
temperature: Temperature for sampling. It can be a tuple of temperatures,
which will be successively used upon failures according to either
`compression_ratio_threshold` or `log_prob_threshold`.
compression_ratio_threshold: If the gzip compression ratio is above this value,
treat as failed.
log_prob_threshold: If the average log probability over sampled tokens is
below this value, treat as failed.
log_prob_low_threshold: This parameter alone is sufficient to skip an output text,
wheras log_prob_threshold also looks for appropriate no_speech_threshold value.
This value should be less than log_prob_threshold.
no_speech_threshold: If the no_speech probability is higher than this value AND
the average log probability over sampled tokens is below `log_prob_threshold`,
consider the segment as silent.
condition_on_previous_text: 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.
prompt_reset_on_temperature: Resets prompt if temperature is above this value.
Arg has effect only if condition_on_previous_text is True.
initial_prompt: Optional text string or iterable of token ids to provide as a
prompt for the first window.
prefix: Optional text to provide as a prefix for the first window.
suppress_blank: Suppress blank outputs at the beginning of the sampling.
suppress_tokens: List of token IDs to suppress. -1 will suppress a default set
of symbols as defined in `tokenizer.non_speech_tokens()`.
without_timestamps: Only sample text tokens.
max_initial_timestamp: The initial timestamp cannot be later than this.
word_timestamps: 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: If word_timestamps is True, merge these punctuation symbols
with the next word
append_punctuations: If word_timestamps is True, merge these punctuation symbols
with the previous word
multilingual: If True, perform transcription on multilingual videos
and return the transcript based
on the 'output_language' flag.
output_language: Valid only if multilingual is set to True.
Specifies the string representing the output language. One of
'en' (English) or 'hybrid' (code-switched transcription).
vad_filter: Enable the voice activity detection (VAD) to filter out parts of the audio
without speech. This step is using the Silero VAD model
https://github.com/snakers4/silero-vad.
vad_parameters: Dictionary of Silero VAD parameters or VadOptions class (see available
parameters and default values in the class `VadOptions`).
max_new_tokens: Maximum number of new tokens to generate per-chunk. If not set,
the maximum will be set by the default max_length.
chunk_length: The length of audio segments. If it is not None, it will overwrite the
default chunk_length of the FeatureExtractor.
clip_timestamps:
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.
vad_filter will be ignored if clip_timestamps is used.
hallucination_silence_threshold:
When word_timestamps is True, skip silent periods longer than this threshold
(in seconds) when a possible hallucination is detected
hotwords:
Hotwords/hint phrases to provide the model with. Has no effect if prefix is not None.
language_detection_threshold: If the maximum probability of the language tokens is higher
than this value, the language is detected.
language_detection_segments: Number of segments to consider for the language detection.
Returns:
A tuple with:
- a generator over transcribed segments
- an instance of TranscriptionInfo
"""
sampling_rate = self.feature_extractor.sampling_rate
if not isinstance(audio, np.ndarray):
audio = decode_audio(audio, sampling_rate=sampling_rate)
duration = audio.shape[0] / sampling_rate
duration_after_vad = duration
self.logger.info(
"Processing audio with duration %s", format_timestamp(duration)
)
if vad_filter and clip_timestamps == "0":
if vad_parameters is None:
vad_parameters = VadOptions()
elif isinstance(vad_parameters, dict):
vad_parameters = VadOptions(**vad_parameters)
speech_chunks = get_speech_timestamps(audio, vad_parameters)
audio_chunks, chunks_metadata = collect_chunks(audio, speech_chunks)
audio = np.concatenate(audio_chunks, axis=0)
duration_after_vad = audio.shape[0] / sampling_rate
self.logger.info(
"VAD filter removed %s of audio",
format_timestamp(duration - duration_after_vad),
)
if self.logger.isEnabledFor(logging.DEBUG):
self.logger.debug(
"VAD filter kept the following audio segments: %s",
", ".join(
"[%s -> %s]"
% (
format_timestamp(chunk["start"] / sampling_rate),
format_timestamp(chunk["end"] / sampling_rate),
)
for chunk in speech_chunks
),
)
else:
speech_chunks = None
features = self.feature_extractor(audio, chunk_length=chunk_length)
encoder_output = None
all_language_probs = None
# setting output_language for multilingual videos
if multilingual:
if output_language is None:
output_language = "en"
elif output_language not in ["en", "hybrid"]:
raise ValueError("Output language needs to be one of 'en'/'hybrid'.")
# detecting the language if not provided
if language is None:
if not self.model.is_multilingual:
language = "en"
language_probability = 1
else:
start_timestamp = (
float(clip_timestamps.split(",")[0])
if isinstance(clip_timestamps, str)
else clip_timestamps[0]
)
content_frames = features.shape[-1] - 1
seek = (
int(start_timestamp * self.frames_per_second)
if start_timestamp * self.frames_per_second < content_frames
else 0
)
(
language,
language_probability,
all_language_probs,
) = self.detect_language(
features=features[..., seek:],
language_detection_segments=language_detection_segments,
language_detection_threshold=language_detection_threshold,
)
self.logger.info(
"Detected language '%s' with probability %.2f",
language,
language_probability,
)
else:
if not self.model.is_multilingual and language != "en":
self.logger.warning(
"The current model is English-only but the language parameter is set to '%s'; "
"using 'en' instead." % language
)
language = "en"
language_probability = 1
tokenizer = Tokenizer(
self.hf_tokenizer,
self.model.is_multilingual,
task=task,
language=language,
)
options = TranscriptionOptions(
beam_size=beam_size,
best_of=best_of,
patience=patience,
length_penalty=length_penalty,
repetition_penalty=repetition_penalty,
no_repeat_ngram_size=no_repeat_ngram_size,
log_prob_threshold=log_prob_threshold,
log_prob_low_threshold=log_prob_low_threshold,
no_speech_threshold=no_speech_threshold,
compression_ratio_threshold=compression_ratio_threshold,
condition_on_previous_text=condition_on_previous_text,
prompt_reset_on_temperature=prompt_reset_on_temperature,
temperatures=(
temperature if isinstance(temperature, (list, tuple)) else [temperature]
),
initial_prompt=initial_prompt,
prefix=prefix,
suppress_blank=suppress_blank,
suppress_tokens=(
get_suppressed_tokens(tokenizer, suppress_tokens)
if suppress_tokens
else suppress_tokens
),
without_timestamps=without_timestamps,
max_initial_timestamp=max_initial_timestamp,
word_timestamps=word_timestamps,
prepend_punctuations=prepend_punctuations,
append_punctuations=append_punctuations,
multilingual=multilingual,
output_language=output_language,
max_new_tokens=max_new_tokens,
clip_timestamps=clip_timestamps,
hallucination_silence_threshold=hallucination_silence_threshold,
hotwords=hotwords,
)
segments = self.generate_segments(
features, tokenizer, options, log_progress, encoder_output
)
if speech_chunks:
segments = restore_speech_timestamps(segments, speech_chunks, sampling_rate)
info = TranscriptionInfo(
language=language,
language_probability=language_probability,
duration=duration,
duration_after_vad=duration_after_vad,
transcription_options=options,
vad_options=vad_parameters,
all_language_probs=all_language_probs,
)
return segments, info
def _split_segments_by_timestamps(
self,
tokenizer: Tokenizer,
tokens: List[int],
time_offset: float,
segment_size: int,
segment_duration: float,
seek: int,
) -> List[List[int]]:
current_segments = []
single_timestamp_ending = (
len(tokens) >= 2 and tokens[-2] < tokenizer.timestamp_begin <= tokens[-1]
)
consecutive_timestamps = [
i
for i in range(len(tokens))
if i > 0
and tokens[i] >= tokenizer.timestamp_begin
and tokens[i - 1] >= tokenizer.timestamp_begin
]
if len(consecutive_timestamps) > 0:
slices = list(consecutive_timestamps)