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from abc import ABC, abstractmethod | |
from collections import Counter | |
from dis import dis | |
from typing import Any, Iterator, List, Dict | |
from pprint import pprint | |
# Workaround for https://github.com/tensorflow/tensorflow/issues/48797 | |
try: | |
import tensorflow as tf | |
except ModuleNotFoundError: | |
# Error handling | |
pass | |
import torch | |
import ffmpeg | |
import numpy as np | |
from utils import format_timestamp | |
# Defaults for Silero | |
# TODO: Make these configurable? | |
SPEECH_TRESHOLD = 0.3 | |
MAX_SILENT_PERIOD = 10 # seconds | |
MAX_MERGE_SIZE = 150 # Do not create segments larger than 2.5 minutes | |
SEGMENT_PADDING_LEFT = 1 # Start detected text segment early | |
SEGMENT_PADDING_RIGHT = 3 # End detected segments late | |
# Whether to attempt to transcribe non-speech | |
TRANSCRIBE_NON_SPEECH = False | |
# Minimum size of segments to process | |
MIN_SEGMENT_DURATION = 1 | |
class AbstractTranscription(ABC): | |
def __init__(self, segment_padding_left: int = None, segment_padding_right = None, max_silent_period: int = None, max_merge_size: int = None, transcribe_non_speech: bool = False): | |
self.sampling_rate = 16000 | |
self.segment_padding_left = segment_padding_left | |
self.segment_padding_right = segment_padding_right | |
self.max_silent_period = max_silent_period | |
self.max_merge_size = max_merge_size | |
self.transcribe_non_speech = transcribe_non_speech | |
def get_audio_segment(self, str, start_time: str = None, duration: str = None): | |
return load_audio(str, self.sampling_rate, start_time, duration) | |
def get_transcribe_timestamps(self, audio: str): | |
""" | |
Get the start and end timestamps of the sections that should be transcribed by this VAD method. | |
Parameters | |
---------- | |
audio: str | |
The audio file. | |
Returns | |
------- | |
A list of start and end timestamps, in fractional seconds. | |
""" | |
return | |
def transcribe(self, audio: str, whisperCallable): | |
""" | |
Transcribe the given audo file. | |
Parameters | |
---------- | |
audio: str | |
The audio file. | |
whisperCallable: Callable[[Union[str, np.ndarray, torch.Tensor]], dict[str, Union[dict, Any]]] | |
The callback that is used to invoke Whisper on an audio file/buffer. | |
Returns | |
------- | |
A list of start and end timestamps, in fractional seconds. | |
""" | |
# get speech timestamps from full audio file | |
seconds_timestamps = self.get_transcribe_timestamps(audio) | |
padded = self.pad_timestamps(seconds_timestamps, self.segment_padding_left, self.segment_padding_right) | |
merged = self.merge_timestamps(padded, self.max_silent_period, self.max_merge_size) | |
print("Timestamps:") | |
pprint(merged) | |
if self.transcribe_non_speech: | |
max_audio_duration = float(ffmpeg.probe(audio)["format"]["duration"]) | |
merged = self.include_gaps(merged, min_gap_length=5, total_duration=max_audio_duration) | |
print("Transcribing non-speech:") | |
pprint(merged) | |
result = { | |
'text': "", | |
'segments': [], | |
'language': "" | |
} | |
languageCounter = Counter() | |
# For each time segment, run whisper | |
for segment in merged: | |
segment_start = segment['start'] | |
segment_end = segment['end'] | |
segment_gap = segment.get('gap', False) | |
segment_duration = segment_end - segment_start | |
if segment_duration < MIN_SEGMENT_DURATION: | |
continue; | |
segment_audio = self.get_audio_segment(audio, start_time = str(segment_start), duration = str(segment_duration)) | |
print("Running whisper from ", format_timestamp(segment_start), " to ", format_timestamp(segment_end), ", duration: ", segment_duration, "gap: ", segment_gap) | |
if segment_gap: | |
# TODO: Use different parameters for these segments, as they are less likely to contain speech | |
segment_result = whisperCallable(segment_audio) | |
else: | |
segment_result = whisperCallable(segment_audio) | |
adjusted_segments = self.adjust_whisper_timestamp(segment_result["segments"], adjust_seconds=segment_start, max_source_time=segment_duration) | |
# Append to output | |
result['text'] += segment_result['text'] | |
result['segments'].extend(adjusted_segments) | |
# Increment detected language | |
languageCounter[segment_result['language']] += 1 | |
if len(languageCounter) > 0: | |
result['language'] = languageCounter.most_common(1)[0][0] | |
return result | |
def include_gaps(self, segments: Iterator[dict], min_gap_length: float, total_duration: float): | |
result = [] | |
last_end_time = 0 | |
for segment in segments: | |
segment_start = float(segment['start']) | |
segment_end = float(segment['end']) | |
if (last_end_time != segment_start): | |
delta = segment_start - last_end_time | |
if (min_gap_length is None or delta >= min_gap_length): | |
result.append( { 'start': last_end_time, 'end': segment_start, 'gap': True } ) | |
last_end_time = segment_end | |
result.append(segment) | |
# Also include total duration if specified | |
if (total_duration is not None and last_end_time < total_duration): | |
delta = total_duration - segment_start | |
if (min_gap_length is None or delta >= min_gap_length): | |
result.append( { 'start': last_end_time, 'end': total_duration, 'gap': True } ) | |
return result | |
def adjust_whisper_timestamp(self, segments: Iterator[dict], adjust_seconds: float, max_source_time: float = None): | |
result = [] | |
for segment in segments: | |
segment_start = float(segment['start']) | |
segment_end = float(segment['end']) | |
# Filter segments? | |
if (max_source_time is not None): | |
if (segment_start > max_source_time): | |
continue | |
segment_end = min(max_source_time, segment_end) | |
new_segment = segment.copy() | |
# Add to start and end | |
new_segment['start'] = segment_start + adjust_seconds | |
new_segment['end'] = segment_end + adjust_seconds | |
result.append(new_segment) | |
return result | |
def pad_timestamps(self, timestamps: List[Dict[str, Any]], padding_left: float, padding_right: float): | |
result = [] | |
for entry in timestamps: | |
segment_start = entry['start'] | |
segment_end = entry['end'] | |
if padding_left is not None: | |
segment_start = max(0, segment_start - padding_left) | |
if padding_right is not None: | |
segment_end = segment_end + padding_right | |
result.append({ 'start': segment_start, 'end': segment_end }) | |
return result | |
def merge_timestamps(self, timestamps: List[Dict[str, Any]], max_merge_gap: float, max_merge_size: float): | |
if max_merge_gap is None: | |
return timestamps | |
result = [] | |
current_entry = None | |
for entry in timestamps: | |
if current_entry is None: | |
current_entry = entry | |
continue | |
# Get distance to the previous entry | |
distance = entry['start'] - current_entry['end'] | |
current_entry_size = current_entry['end'] - current_entry['start'] | |
if distance <= max_merge_gap and (max_merge_size is None or current_entry_size <= max_merge_size): | |
# Merge | |
current_entry['end'] = entry['end'] | |
else: | |
# Output current entry | |
result.append(current_entry) | |
current_entry = entry | |
# Add final entry | |
if current_entry is not None: | |
result.append(current_entry) | |
return result | |
def multiply_timestamps(self, timestamps: List[Dict[str, Any]], factor: float): | |
result = [] | |
for entry in timestamps: | |
start = entry['start'] | |
end = entry['end'] | |
result.append({ | |
'start': start * factor, | |
'end': end * factor | |
}) | |
return result | |
class VadSileroTranscription(AbstractTranscription): | |
def __init__(self, segment_padding_left=SEGMENT_PADDING_LEFT, segment_padding_right=SEGMENT_PADDING_RIGHT, | |
max_silent_period=MAX_SILENT_PERIOD, max_merge_size=MAX_MERGE_SIZE, transcribe_non_speech: bool = False, | |
copy = None): | |
super().__init__(segment_padding_left=segment_padding_left, segment_padding_right=segment_padding_right, | |
max_silent_period=max_silent_period, max_merge_size=max_merge_size, transcribe_non_speech=transcribe_non_speech) | |
if copy: | |
self.model = copy.model | |
self.get_speech_timestamps = copy.get_speech_timestamps | |
else: | |
self.model, utils = torch.hub.load(repo_or_dir='snakers4/silero-vad', model='silero_vad') | |
(self.get_speech_timestamps, _, _, _, _) = utils | |
def get_transcribe_timestamps(self, audio: str): | |
wav = self.get_audio_segment(audio) | |
sample_timestamps = self.get_speech_timestamps(wav, self.model, sampling_rate=self.sampling_rate, threshold=SPEECH_TRESHOLD) | |
seconds_timestamps = self.multiply_timestamps(sample_timestamps, factor=1 / self.sampling_rate) | |
return seconds_timestamps | |
# A very simple VAD that just marks every N seconds as speech | |
class VadPeriodicTranscription(AbstractTranscription): | |
def __init__(self, periodic_duration: int): | |
super().__init__() | |
self.periodic_duration = periodic_duration | |
def get_transcribe_timestamps(self, audio: str): | |
# Get duration in seconds | |
audio_duration = float(ffmpeg.probe(audio)["format"]["duration"]) | |
result = [] | |
# Generate a timestamp every N seconds | |
start_timestamp = 0 | |
while (start_timestamp < audio_duration): | |
end_timestamp = min(start_timestamp + self.periodic_duration, audio_duration) | |
segment_duration = end_timestamp - start_timestamp | |
# Minimum duration is 1 second | |
if (segment_duration >= 1): | |
result.append( { 'start': start_timestamp, 'end': end_timestamp } ) | |
start_timestamp = end_timestamp | |
return result | |
def load_audio(file: str, sample_rate: int = 16000, | |
start_time: str = None, duration: str = None): | |
""" | |
Open an audio file and read as mono waveform, resampling as necessary | |
Parameters | |
---------- | |
file: str | |
The audio file to open | |
sr: int | |
The sample rate to resample the audio if necessary | |
start_time: str | |
The start time, using the standard FFMPEG time duration syntax, or None to disable. | |
duration: str | |
The duration, using the standard FFMPEG time duration syntax, or None to disable. | |
Returns | |
------- | |
A NumPy array containing the audio waveform, in float32 dtype. | |
""" | |
try: | |
inputArgs = {'threads': 0} | |
if (start_time is not None): | |
inputArgs['ss'] = start_time | |
if (duration is not None): | |
inputArgs['t'] = duration | |
# This launches a subprocess to decode audio while down-mixing and resampling as necessary. | |
# Requires the ffmpeg CLI and `ffmpeg-python` package to be installed. | |
out, _ = ( | |
ffmpeg.input(file, **inputArgs) | |
.output("-", format="s16le", acodec="pcm_s16le", ac=1, ar=sample_rate) | |
.run(cmd="ffmpeg", capture_stdout=True, capture_stderr=True) | |
) | |
except ffmpeg.Error as e: | |
raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") | |
return np.frombuffer(out, np.int16).flatten().astype(np.float32) / 32768.0 |