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# original https://github.com/guillaumekln/faster-whisper/blob/master/faster_whisper/transcribe.py | |
import itertools | |
import logging | |
import os | |
import zlib | |
from typing import BinaryIO, Iterable, List, NamedTuple, Optional, Tuple, Union | |
import ctranslate2 | |
import numpy as np | |
import tokenizers | |
from faster_whisper.audio import decode_audio | |
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_logger | |
from faster_whisper.vad import ( | |
SpeechTimestampsMap, | |
VadOptions, | |
collect_chunks, | |
get_speech_timestamps, | |
) | |
class Word(NamedTuple): | |
start: float | |
end: float | |
word: str | |
probability: float | |
class Segment(NamedTuple): | |
id: int | |
seek: int | |
start: float | |
end: float | |
text: str | |
tokens: List[int] | |
temperature: float | |
avg_logprob: float | |
compression_ratio: float | |
no_speech_prob: float | |
words: Optional[List[Word]] | |
class TranscriptionOptions(NamedTuple): | |
beam_size: int | |
best_of: int | |
patience: float | |
length_penalty: float | |
repetition_penalty: float | |
no_repeat_ngram_size: int | |
log_prob_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 | |
class TranscriptionInfo(NamedTuple): | |
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 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, | |
): | |
"""Initializes the Whisper model. | |
Args: | |
model_size_or_path: Size of the model to use (tiny, tiny.en, base, base.en, | |
small, small.en, medium, medium.en, large-v1, large-v2, or large), a path to a converted | |
model directory, or a CTranslate2-converted Whisper model ID from the Hugging Face 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. | |
""" | |
self.logger = get_logger() | |
if 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.model = ctranslate2.models.Whisper( | |
model_path, | |
device=device, | |
device_index=device_index, | |
compute_type=compute_type, | |
intra_threads=cpu_threads, | |
inter_threads=num_workers, | |
) | |
tokenizer_file = os.path.join(model_path, "tokenizer.json") | |
if 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.feature_extractor = FeatureExtractor() | |
self.num_samples_per_token = self.feature_extractor.hop_length * 2 | |
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.input_stride = 2 | |
self.time_precision = 0.02 | |
self.max_length = 448 | |
def supported_languages(self) -> List[str]: | |
"""The languages supported by the model.""" | |
return list(_LANGUAGE_CODES) if self.model.is_multilingual else ["en"] | |
def transcribe( | |
self, | |
audio: Union[str, BinaryIO, np.ndarray], | |
language: Optional[str] = None, | |
task: str = "transcribe", | |
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, | |
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 = "\"'.。,,!!??::”)]}、", | |
vad_filter: bool = False, | |
vad_parameters: Optional[Union[dict, VadOptions]] = None, | |
) -> 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). | |
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. | |
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 the model config.json file. | |
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 | |
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`). | |
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: | |
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 = collect_chunks(audio, speech_chunks) | |
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) | |
encoder_output = None | |
all_language_probs = None | |
if language is None: | |
if not self.model.is_multilingual: | |
language = "en" | |
language_probability = 1 | |
else: | |
segment = features[:, : self.feature_extractor.nb_max_frames] | |
encoder_output = self.encode(segment) | |
# results is a list of tuple[str, float] with language names and | |
# probabilities. | |
results = self.model.detect_language(encoder_output)[0] | |
# Parse language names to strip out markers | |
all_language_probs = [(token[2:-2], prob) for (token, prob) in results] | |
# Get top language token and probability | |
language, language_probability = all_language_probs[0] | |
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, | |
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), | |
without_timestamps=without_timestamps, | |
max_initial_timestamp=max_initial_timestamp, | |
word_timestamps=word_timestamps, | |
prepend_punctuations=prepend_punctuations, | |
append_punctuations=append_punctuations, | |
) | |
segments = self.generate_segments(features, tokenizer, options, 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 generate_segments( | |
self, | |
features: np.ndarray, | |
tokenizer: Tokenizer, | |
options: TranscriptionOptions, | |
encoder_output: Optional[ctranslate2.StorageView] = None, | |
) -> Iterable[Segment]: | |
content_frames = features.shape[-1] - self.feature_extractor.nb_max_frames | |
idx = 0 | |
seek = 0 | |
all_tokens = [] | |
prompt_reset_since = 0 | |
if options.initial_prompt is not None: | |
if isinstance(options.initial_prompt, str): | |
initial_prompt = " " + options.initial_prompt.strip() | |
initial_prompt_tokens = tokenizer.encode(initial_prompt) | |
all_tokens.extend(initial_prompt_tokens) | |
else: | |
all_tokens.extend(options.initial_prompt) | |
last_speech_timestamp = 0.0 | |
all_segments = [] | |
while seek < content_frames: | |
time_offset = seek * self.feature_extractor.time_per_frame | |
segment = features[:, seek : seek + self.feature_extractor.nb_max_frames] | |
segment_size = min( | |
self.feature_extractor.nb_max_frames, content_frames - seek | |
) | |
segment_duration = segment_size * self.feature_extractor.time_per_frame | |
if self.logger.isEnabledFor(logging.DEBUG): | |
self.logger.debug( | |
"Processing segment at %s", format_timestamp(time_offset) | |
) | |
previous_tokens = all_tokens[prompt_reset_since:] | |
prompt = self.get_prompt( | |
tokenizer, | |
previous_tokens, | |
without_timestamps=options.without_timestamps, | |
prefix=options.prefix if seek == 0 else None, | |
) | |
if seek > 0 or encoder_output is None: | |
encoder_output = self.encode(segment) | |
( | |
result, | |
avg_logprob, | |
temperature, | |
compression_ratio, | |
) = self.generate_with_fallback(encoder_output, prompt, tokenizer, options) | |
if options.no_speech_threshold is not None: | |
# no voice activity check | |
should_skip = result.no_speech_prob > options.no_speech_threshold | |
if ( | |
options.log_prob_threshold is not None | |
and avg_logprob > options.log_prob_threshold | |
): | |
# don't skip if the logprob is high enough, despite the no_speech_prob | |
should_skip = False | |
if should_skip: | |
self.logger.debug( | |
"No speech threshold is met (%f > %f)", | |
result.no_speech_prob, | |
options.no_speech_threshold, | |
) | |
# fast-forward to the next segment boundary | |
seek += segment_size | |
continue | |
tokens = result.sequences_ids[0] | |
previous_seek = seek | |
current_segments = [] | |
single_timestamp_ending = ( | |
len(tokens) >= 2 | |
and tokens[-2] < tokenizer.timestamp_begin | |
and tokens[-1] >= tokenizer.timestamp_begin | |
) | |
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) | |
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_position = ( | |
sliced_tokens[0] - tokenizer.timestamp_begin | |
) | |
end_timestamp_position = ( | |
sliced_tokens[-1] - tokenizer.timestamp_begin | |
) | |
start_time = ( | |
time_offset + start_timestamp_position * self.time_precision | |
) | |
end_time = ( | |
time_offset + end_timestamp_position * self.time_precision | |
) | |
current_segments.append( | |
dict( | |
seek=seek, | |
start=start_time, | |
end=end_time, | |
tokens=sliced_tokens, | |
) | |
) | |
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_position = ( | |
tokens[last_slice - 1] - tokenizer.timestamp_begin | |
) | |
seek += last_timestamp_position * self.input_stride | |
else: | |
duration = segment_duration | |
timestamps = [ | |
token for token in tokens if token >= tokenizer.timestamp_begin | |
] | |
if len(timestamps) > 0 and timestamps[-1] != tokenizer.timestamp_begin: | |
last_timestamp_position = timestamps[-1] - tokenizer.timestamp_begin | |
duration = last_timestamp_position * self.time_precision | |
current_segments.append( | |
dict( | |
seek=seek, | |
start=time_offset, | |
end=time_offset + duration, | |
tokens=tokens, | |
) | |
) | |
seek += segment_size | |
if options.word_timestamps: | |
self.add_word_timestamps( | |
current_segments, | |
tokenizer, | |
encoder_output, | |
segment_size, | |
options.prepend_punctuations, | |
options.append_punctuations, | |
last_speech_timestamp=last_speech_timestamp, | |
) | |
word_end_timestamps = [ | |
w["end"] for s in current_segments for w in s["words"] | |
] | |
if len(word_end_timestamps) > 0: | |
last_speech_timestamp = word_end_timestamps[-1] | |
if not single_timestamp_ending and len(word_end_timestamps) > 0: | |
seek_shift = round( | |
(word_end_timestamps[-1] - time_offset) * self.frames_per_second | |
) | |
if seek_shift > 0: | |
seek = previous_seek + seek_shift | |
for segment in current_segments: | |
tokens = segment["tokens"] | |
text = tokenizer.decode(tokens) | |
if segment["start"] == segment["end"] or not text.strip(): | |
continue | |
all_tokens.extend(tokens) | |
idx += 1 | |
all_segments.append(Segment( | |
id=idx, | |
seek=seek, | |
start=segment["start"], | |
end=segment["end"], | |
text=text, | |
tokens=tokens, | |
temperature=temperature, | |
avg_logprob=avg_logprob, | |
compression_ratio=compression_ratio, | |
no_speech_prob=result.no_speech_prob, | |
words=( | |
[Word(**word) for word in segment["words"]] | |
if options.word_timestamps | |
else None | |
), | |
)) | |
if ( | |
not options.condition_on_previous_text | |
or temperature > options.prompt_reset_on_temperature | |
): | |
if options.condition_on_previous_text: | |
self.logger.debug( | |
"Reset prompt. prompt_reset_on_temperature threshold is met %f > %f", | |
temperature, | |
options.prompt_reset_on_temperature, | |
) | |
prompt_reset_since = len(all_tokens) | |
return all_segments | |
def encode(self, features: np.ndarray) -> ctranslate2.StorageView: | |
# When the model is running on multiple GPUs, the encoder output should be moved | |
# to the CPU since we don't know which GPU will handle the next job. | |
to_cpu = self.model.device == "cuda" and len(self.model.device_index) > 1 | |
features = np.expand_dims(features, 0) | |
features = get_ctranslate2_storage(features) | |
return self.model.encode(features, to_cpu=to_cpu) | |
def generate_with_fallback( | |
self, | |
encoder_output: ctranslate2.StorageView, | |
prompt: List[int], | |
tokenizer: Tokenizer, | |
options: TranscriptionOptions, | |
) -> Tuple[ctranslate2.models.WhisperGenerationResult, float, float, float]: | |
decode_result = None | |
all_results = [] | |
below_cr_threshold_results = [] | |
max_initial_timestamp_index = int( | |
round(options.max_initial_timestamp / self.time_precision) | |
) | |
for temperature in options.temperatures: | |
if temperature > 0: | |
kwargs = { | |
"beam_size": 1, | |
"num_hypotheses": options.best_of, | |
"sampling_topk": 0, | |
"sampling_temperature": temperature, | |
} | |
else: | |
kwargs = { | |
"beam_size": options.beam_size, | |
"patience": options.patience, | |
} | |
result = self.model.generate( | |
encoder_output, | |
[prompt], | |
length_penalty=options.length_penalty, | |
repetition_penalty=options.repetition_penalty, | |
no_repeat_ngram_size=options.no_repeat_ngram_size, | |
max_length=self.max_length, | |
return_scores=True, | |
return_no_speech_prob=True, | |
suppress_blank=options.suppress_blank, | |
suppress_tokens=options.suppress_tokens, | |
max_initial_timestamp_index=max_initial_timestamp_index, | |
**kwargs, | |
)[0] | |
tokens = result.sequences_ids[0] | |
# Recover the average log prob from the returned score. | |
seq_len = len(tokens) | |
cum_logprob = result.scores[0] * (seq_len**options.length_penalty) | |
avg_logprob = cum_logprob / (seq_len + 1) | |
text = tokenizer.decode(tokens).strip() | |
compression_ratio = get_compression_ratio(text) | |
decode_result = ( | |
result, | |
avg_logprob, | |
temperature, | |
compression_ratio, | |
) | |
all_results.append(decode_result) | |
needs_fallback = False | |
if options.compression_ratio_threshold is not None: | |
if compression_ratio > options.compression_ratio_threshold: | |
needs_fallback = True # too repetitive | |
self.logger.debug( | |
"Compression ratio threshold is not met with temperature %.1f (%f > %f)", | |
temperature, | |
compression_ratio, | |
options.compression_ratio_threshold, | |
) | |
else: | |
below_cr_threshold_results.append(decode_result) | |
if ( | |
options.log_prob_threshold is not None | |
and avg_logprob < options.log_prob_threshold | |
): | |
needs_fallback = True # average log probability is too low | |
self.logger.debug( | |
"Log probability threshold is not met with temperature %.1f (%f < %f)", | |
temperature, | |
avg_logprob, | |
options.log_prob_threshold, | |
) | |
if ( | |
options.no_speech_threshold is not None | |
and result.no_speech_prob > options.no_speech_threshold | |
): | |
needs_fallback = False # silence | |
if not needs_fallback: | |
break | |
else: | |
# all failed, select the result with the highest average log probability | |
decode_result = max( | |
below_cr_threshold_results or all_results, key=lambda x: x[1] | |
) | |
return decode_result | |
def get_prompt( | |
self, | |
tokenizer: Tokenizer, | |
previous_tokens: List[int], | |
without_timestamps: bool = False, | |
prefix: Optional[str] = None, | |
) -> List[int]: | |
prompt = [] | |
if previous_tokens: | |
prompt.append(tokenizer.sot_prev) | |
prompt.extend(previous_tokens[-(self.max_length // 2 - 1) :]) | |
prompt.extend(tokenizer.sot_sequence) | |
if without_timestamps: | |
prompt.append(tokenizer.no_timestamps) | |
if prefix: | |
prefix_tokens = tokenizer.encode(" " + prefix.strip()) | |
if len(prefix_tokens) >= self.max_length // 2: | |
prefix_tokens = prefix_tokens[: self.max_length // 2 - 1] | |
if not without_timestamps: | |
prompt.append(tokenizer.timestamp_begin) | |
prompt.extend(prefix_tokens) | |
return prompt | |
def add_word_timestamps( | |
self, | |
segments: List[dict], | |
tokenizer: Tokenizer, | |
encoder_output: ctranslate2.StorageView, | |
num_frames: int, | |
prepend_punctuations: str, | |
append_punctuations: str, | |
last_speech_timestamp: float, | |
) -> None: | |
if len(segments) == 0: | |
return | |
text_tokens_per_segment = [ | |
[token for token in segment["tokens"] if token < tokenizer.eot] | |
for segment in segments | |
] | |
text_tokens = list(itertools.chain.from_iterable(text_tokens_per_segment)) | |
alignment = self.find_alignment( | |
tokenizer, text_tokens, encoder_output, num_frames | |
) | |
word_durations = np.array([word["end"] - word["start"] for word in alignment]) | |
word_durations = word_durations[word_durations.nonzero()] | |
median_duration = np.median(word_durations) if len(word_durations) > 0 else 0.0 | |
max_duration = median_duration * 2 | |
# hack: truncate long words at sentence boundaries. | |
# a better segmentation algorithm based on VAD should be able to replace this. | |
if len(word_durations) > 0: | |
sentence_end_marks = ".。!!??" | |
# ensure words at sentence boundaries | |
# are not longer than twice the median word duration. | |
for i in range(1, len(alignment)): | |
if alignment[i]["end"] - alignment[i]["start"] > max_duration: | |
if alignment[i]["word"] in sentence_end_marks: | |
alignment[i]["end"] = alignment[i]["start"] + max_duration | |
elif alignment[i - 1]["word"] in sentence_end_marks: | |
alignment[i]["start"] = alignment[i]["end"] - max_duration | |
merge_punctuations(alignment, prepend_punctuations, append_punctuations) | |
time_offset = ( | |
segments[0]["seek"] | |
* self.feature_extractor.hop_length | |
/ self.feature_extractor.sampling_rate | |
) | |
word_index = 0 | |
for segment, text_tokens in zip(segments, text_tokens_per_segment): | |
saved_tokens = 0 | |
words = [] | |
while word_index < len(alignment) and saved_tokens < len(text_tokens): | |
timing = alignment[word_index] | |
if timing["word"]: | |
words.append( | |
dict( | |
word=timing["word"], | |
start=round(time_offset + timing["start"], 2), | |
end=round(time_offset + timing["end"], 2), | |
probability=timing["probability"], | |
) | |
) | |
saved_tokens += len(timing["tokens"]) | |
word_index += 1 | |
# hack: truncate long words at segment boundaries. | |
# a better segmentation algorithm based on VAD should be able to replace this. | |
if len(words) > 0: | |
# ensure the first and second word after a pause is not longer than | |
# twice the median word duration. | |
if words[0]["end"] - last_speech_timestamp > median_duration * 4 and ( | |
words[0]["end"] - words[0]["start"] > max_duration | |
or ( | |
len(words) > 1 | |
and words[1]["end"] - words[0]["start"] > max_duration * 2 | |
) | |
): | |
if ( | |
len(words) > 1 | |
and words[1]["end"] - words[1]["start"] > max_duration | |
): | |
boundary = max( | |
words[1]["end"] / 2, words[1]["end"] - max_duration | |
) | |
words[0]["end"] = words[1]["start"] = boundary | |
words[0]["start"] = max(0, words[0]["end"] - max_duration) | |
# prefer the segment-level start timestamp if the first word is too long. | |
if ( | |
segment["start"] < words[0]["end"] | |
and segment["start"] - 0.5 > words[0]["start"] | |
): | |
words[0]["start"] = max( | |
0, min(words[0]["end"] - median_duration, segment["start"]) | |
) | |
else: | |
segment["start"] = words[0]["start"] | |
# prefer the segment-level end timestamp if the last word is too long. | |
if ( | |
segment["end"] > words[-1]["start"] | |
and segment["end"] + 0.5 < words[-1]["end"] | |
): | |
words[-1]["end"] = max( | |
words[-1]["start"] + median_duration, segment["end"] | |
) | |
else: | |
segment["end"] = words[-1]["end"] | |
last_speech_timestamp = segment["end"] | |
segment["words"] = words | |
def find_alignment( | |
self, | |
tokenizer: Tokenizer, | |
text_tokens: List[int], | |
encoder_output: ctranslate2.StorageView, | |
num_frames: int, | |
median_filter_width: int = 7, | |
) -> List[dict]: | |
if len(text_tokens) == 0: | |
return [] | |
result = self.model.align( | |
encoder_output, | |
tokenizer.sot_sequence, | |
[text_tokens], | |
num_frames, | |
median_filter_width=median_filter_width, | |
)[0] | |
text_token_probs = result.text_token_probs | |
alignments = result.alignments | |
text_indices = np.array([pair[0] for pair in alignments]) | |
time_indices = np.array([pair[1] for pair in alignments]) | |
words, word_tokens = tokenizer.split_to_word_tokens( | |
text_tokens + [tokenizer.eot] | |
) | |
word_boundaries = np.pad(np.cumsum([len(t) for t in word_tokens[:-1]]), (1, 0)) | |
if len(word_boundaries) <= 1: | |
return [] | |
jumps = np.pad(np.diff(text_indices), (1, 0), constant_values=1).astype(bool) | |
jump_times = time_indices[jumps] / self.tokens_per_second | |
start_times = jump_times[word_boundaries[:-1]] | |
end_times = jump_times[word_boundaries[1:]] | |
word_probabilities = [ | |
np.mean(text_token_probs[i:j]) | |
for i, j in zip(word_boundaries[:-1], word_boundaries[1:]) | |
] | |
return [ | |
dict( | |
word=word, tokens=tokens, start=start, end=end, probability=probability | |
) | |
for word, tokens, start, end, probability in zip( | |
words, word_tokens, start_times, end_times, word_probabilities | |
) | |
] | |
def destroy(self): | |
del self.model | |
def restore_speech_timestamps( | |
segments: Iterable[Segment], | |
speech_chunks: List[dict], | |
sampling_rate: int, | |
) -> Iterable[Segment]: | |
ts_map = SpeechTimestampsMap(speech_chunks, sampling_rate) | |
for segment in segments: | |
if segment.words: | |
words = [] | |
for word in segment.words: | |
# Ensure the word start and end times are resolved to the same chunk. | |
middle = (word.start + word.end) / 2 | |
chunk_index = ts_map.get_chunk_index(middle) | |
word = word._replace( | |
start=ts_map.get_original_time(word.start, chunk_index), | |
end=ts_map.get_original_time(word.end, chunk_index), | |
) | |
words.append(word) | |
segment = segment._replace( | |
start=words[0].start, | |
end=words[-1].end, | |
words=words, | |
) | |
else: | |
segment = segment._replace( | |
start=ts_map.get_original_time(segment.start), | |
end=ts_map.get_original_time(segment.end), | |
) | |
return segments | |
def get_ctranslate2_storage(segment: np.ndarray) -> ctranslate2.StorageView: | |
segment = np.ascontiguousarray(segment) | |
segment = ctranslate2.StorageView.from_array(segment) | |
return segment | |
def get_compression_ratio(text: str) -> float: | |
text_bytes = text.encode("utf-8") | |
return len(text_bytes) / len(zlib.compress(text_bytes)) | |
def get_suppressed_tokens( | |
tokenizer: Tokenizer, | |
suppress_tokens: Optional[List[int]], | |
) -> Optional[List[int]]: | |
if not suppress_tokens or -1 in suppress_tokens: | |
return suppress_tokens | |
suppress_tokens = list(suppress_tokens) | |
# Ensure the following special tokens are suppressed when the user does | |
# not use the default set (-1). | |
suppress_tokens.extend( | |
[ | |
tokenizer.transcribe, | |
tokenizer.translate, | |
tokenizer.sot, | |
tokenizer.sot_prev, | |
tokenizer.sot_lm, | |
] | |
) | |
return sorted(set(suppress_tokens)) | |
def merge_punctuations(alignment: List[dict], prepended: str, appended: str) -> None: | |
# merge prepended punctuations | |
i = len(alignment) - 2 | |
j = len(alignment) - 1 | |
while i >= 0: | |
previous = alignment[i] | |
following = alignment[j] | |
if previous["word"].startswith(" ") and previous["word"].strip() in prepended: | |
# prepend it to the following word | |
following["word"] = previous["word"] + following["word"] | |
following["tokens"] = previous["tokens"] + following["tokens"] | |
previous["word"] = "" | |
previous["tokens"] = [] | |
else: | |
j = i | |
i -= 1 | |
# merge appended punctuations | |
i = 0 | |
j = 1 | |
while j < len(alignment): | |
previous = alignment[i] | |
following = alignment[j] | |
if not previous["word"].endswith(" ") and following["word"] in appended: | |
# append it to the previous word | |
previous["word"] = previous["word"] + following["word"] | |
previous["tokens"] = previous["tokens"] + following["tokens"] | |
following["word"] = "" | |
following["tokens"] = [] | |
else: | |
i = j | |
j += 1 | |