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Add progress listener to none/VAD
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from abc import ABC, abstractmethod
from collections import Counter, deque
import time
from typing import Any, Deque, Iterator, List, Dict
from pprint import pprint
from src.hooks.whisperProgressHook import ProgressListener, SubTaskProgressListener, create_progress_listener_handle
from src.modelCache import GLOBAL_MODEL_CACHE, ModelCache
from src.segments import merge_timestamps
from src.whisperContainer import WhisperCallback
# 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 src.utils import format_timestamp
from enum import Enum
class NonSpeechStrategy(Enum):
"""
Ignore non-speech frames segments.
"""
SKIP = 1
"""
Just treat non-speech segments as speech.
"""
CREATE_SEGMENT = 2
"""
Expand speech segments into subsequent non-speech segments.
"""
EXPAND_SEGMENT = 3
# Defaults for Silero
SPEECH_TRESHOLD = 0.3
# Minimum size of segments to process
MIN_SEGMENT_DURATION = 1
# The maximum time for texts from old segments to be used in the next segment
MAX_PROMPT_WINDOW = 0 # seconds (0 = disabled)
PROMPT_NO_SPEECH_PROB = 0.1 # Do not pass the text from segments with a no speech probability higher than this
VAD_MAX_PROCESSING_CHUNK = 60 * 60 # 60 minutes of audio
class TranscriptionConfig(ABC):
def __init__(self, non_speech_strategy: NonSpeechStrategy = NonSpeechStrategy.SKIP,
segment_padding_left: float = None, segment_padding_right = None, max_silent_period: float = None,
max_merge_size: float = None, max_prompt_window: float = None, initial_segment_index = -1):
self.non_speech_strategy = non_speech_strategy
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.max_prompt_window = max_prompt_window
self.initial_segment_index = initial_segment_index
class PeriodicTranscriptionConfig(TranscriptionConfig):
def __init__(self, periodic_duration: float, non_speech_strategy: NonSpeechStrategy = NonSpeechStrategy.SKIP,
segment_padding_left: float = None, segment_padding_right = None, max_silent_period: float = None,
max_merge_size: float = None, max_prompt_window: float = None, initial_segment_index = -1):
super().__init__(non_speech_strategy, segment_padding_left, segment_padding_right, max_silent_period, max_merge_size, max_prompt_window, initial_segment_index)
self.periodic_duration = periodic_duration
class AbstractTranscription(ABC):
def __init__(self, sampling_rate: int = 16000):
self.sampling_rate = sampling_rate
def get_audio_segment(self, str, start_time: str = None, duration: str = None):
return load_audio(str, self.sampling_rate, start_time, duration)
def is_transcribe_timestamps_fast(self):
"""
Determine if get_transcribe_timestamps is fast enough to not need parallelization.
"""
return False
@abstractmethod
def get_transcribe_timestamps(self, audio: str, config: TranscriptionConfig, start_time: float, end_time: float):
"""
Get the start and end timestamps of the sections that should be transcribed by this VAD method.
Parameters
----------
audio: str
The audio file.
config: TranscriptionConfig
The transcription configuration.
Returns
-------
A list of start and end timestamps, in fractional seconds.
"""
return
def get_merged_timestamps(self, timestamps: List[Dict[str, Any]], config: TranscriptionConfig, total_duration: float):
"""
Get the start and end timestamps of the sections that should be transcribed by this VAD method,
after merging the given segments using the specified configuration.
Parameters
----------
audio: str
The audio file.
config: TranscriptionConfig
The transcription configuration.
Returns
-------
A list of start and end timestamps, in fractional seconds.
"""
merged = merge_timestamps(timestamps, config.max_silent_period, config.max_merge_size,
config.segment_padding_left, config.segment_padding_right)
if config.non_speech_strategy != NonSpeechStrategy.SKIP:
# Expand segments to include the gaps between them
if (config.non_speech_strategy == NonSpeechStrategy.CREATE_SEGMENT):
# When we have a prompt window, we create speech segments betwen each segment if we exceed the merge size
merged = self.fill_gaps(merged, total_duration=total_duration, max_expand_size=config.max_merge_size)
elif config.non_speech_strategy == NonSpeechStrategy.EXPAND_SEGMENT:
# With no prompt window, it is better to just expand the segments (this effectively passes the prompt to the next segment)
merged = self.expand_gaps(merged, total_duration=total_duration)
else:
raise Exception("Unknown non-speech strategy: " + str(config.non_speech_strategy))
print("Transcribing non-speech:")
pprint(merged)
return merged
def transcribe(self, audio: str, whisperCallable: WhisperCallback, config: TranscriptionConfig,
progressListener: ProgressListener = None):
"""
Transcribe the given audo file.
Parameters
----------
audio: str
The audio file.
whisperCallable: WhisperCallback
A callback object to call to transcribe each segment.
Returns
-------
A list of start and end timestamps, in fractional seconds.
"""
max_audio_duration = get_audio_duration(audio)
timestamp_segments = self.get_transcribe_timestamps(audio, config, 0, max_audio_duration)
# Get speech timestamps from full audio file
merged = self.get_merged_timestamps(timestamp_segments, config, max_audio_duration)
# A deque of transcribed segments that is passed to the next segment as a prompt
prompt_window = deque()
print("Processing timestamps:")
pprint(merged)
result = {
'text': "",
'segments': [],
'language': ""
}
languageCounter = Counter()
detected_language = None
segment_index = config.initial_segment_index
# For each time segment, run whisper
for segment in merged:
segment_index += 1
segment_start = segment['start']
segment_end = segment['end']
segment_expand_amount = segment.get('expand_amount', 0)
segment_gap = segment.get('gap', False)
segment_duration = segment_end - segment_start
if segment_duration < MIN_SEGMENT_DURATION:
continue
# Audio to run on Whisper
segment_audio = self.get_audio_segment(audio, start_time = str(segment_start), duration = str(segment_duration))
# Previous segments to use as a prompt
segment_prompt = ' '.join([segment['text'] for segment in prompt_window]) if len(prompt_window) > 0 else None
# Detected language
detected_language = languageCounter.most_common(1)[0][0] if len(languageCounter) > 0 else None
print("Running whisper from ", format_timestamp(segment_start), " to ", format_timestamp(segment_end), ", duration: ",
segment_duration, "expanded: ", segment_expand_amount, "prompt: ", segment_prompt, "language: ", detected_language)
scaled_progress_listener = SubTaskProgressListener(progressListener, base_task_total=max_audio_duration, sub_task_start=segment_start, sub_task_total=segment_duration)
segment_result = whisperCallable.invoke(segment_audio, segment_index, segment_prompt, detected_language, progress_listener=scaled_progress_listener)
adjusted_segments = self.adjust_timestamp(segment_result["segments"], adjust_seconds=segment_start, max_source_time=segment_duration)
# Propagate expand amount to the segments
if (segment_expand_amount > 0):
segment_without_expansion = segment_duration - segment_expand_amount
for adjusted_segment in adjusted_segments:
adjusted_segment_end = adjusted_segment['end']
# Add expand amount if the segment got expanded
if (adjusted_segment_end > segment_without_expansion):
adjusted_segment["expand_amount"] = adjusted_segment_end - segment_without_expansion
# Append to output
result['text'] += segment_result['text']
result['segments'].extend(adjusted_segments)
# Increment detected language
if not segment_gap:
languageCounter[segment_result['language']] += 1
# Update prompt window
self.__update_prompt_window(prompt_window, adjusted_segments, segment_end, segment_gap, config)
if detected_language is not None:
result['language'] = detected_language
return result
def __update_prompt_window(self, prompt_window: Deque, adjusted_segments: List, segment_end: float, segment_gap: bool, config: TranscriptionConfig):
if (config.max_prompt_window is not None and config.max_prompt_window > 0):
# Add segments to the current prompt window (unless it is a speech gap)
if not segment_gap:
for segment in adjusted_segments:
if segment.get('no_speech_prob', 0) <= PROMPT_NO_SPEECH_PROB:
prompt_window.append(segment)
while (len(prompt_window) > 0):
first_end_time = prompt_window[0].get('end', 0)
# Time expanded in the segments should be discounted from the prompt window
first_expand_time = prompt_window[0].get('expand_amount', 0)
if (first_end_time - first_expand_time < segment_end - config.max_prompt_window):
prompt_window.popleft()
else:
break
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
# Expand the end time of each segment to the start of the next segment
def expand_gaps(self, segments: List[Dict[str, Any]], total_duration: float):
result = []
if len(segments) == 0:
return result
# Add gap at the beginning if needed
if (segments[0]['start'] > 0):
result.append({ 'start': 0, 'end': segments[0]['start'], 'gap': True } )
for i in range(len(segments) - 1):
current_segment = segments[i]
next_segment = segments[i + 1]
delta = next_segment['start'] - current_segment['end']
# Expand if the gap actually exists
if (delta >= 0):
current_segment = current_segment.copy()
current_segment['expand_amount'] = delta
current_segment['end'] = next_segment['start']
result.append(current_segment)
# Add last segment
last_segment = segments[-1]
result.append(last_segment)
# Also include total duration if specified
if (total_duration is not None):
last_segment = result[-1]
if (last_segment['end'] < total_duration):
last_segment = last_segment.copy()
last_segment['end'] = total_duration
result[-1] = last_segment
return result
def fill_gaps(self, segments: List[Dict[str, Any]], total_duration: float, max_expand_size: float = None):
result = []
if len(segments) == 0:
return result
# Add gap at the beginning if needed
if (segments[0]['start'] > 0):
result.append({ 'start': 0, 'end': segments[0]['start'], 'gap': True } )
for i in range(len(segments) - 1):
expanded = False
current_segment = segments[i]
next_segment = segments[i + 1]
delta = next_segment['start'] - current_segment['end']
if (max_expand_size is not None and delta <= max_expand_size):
# Just expand the current segment
current_segment = current_segment.copy()
current_segment['expand_amount'] = delta
current_segment['end'] = next_segment['start']
expanded = True
result.append(current_segment)
# Add a gap to the next segment if needed
if (delta >= 0 and not expanded):
result.append({ 'start': current_segment['end'], 'end': next_segment['start'], 'gap': True } )
# Add last segment
last_segment = segments[-1]
result.append(last_segment)
# Also include total duration if specified
if (total_duration is not None):
last_segment = result[-1]
delta = total_duration - last_segment['end']
if (delta > 0):
if (max_expand_size is not None and delta <= max_expand_size):
# Expand the last segment
last_segment = last_segment.copy()
last_segment['expand_amount'] = delta
last_segment['end'] = total_duration
result[-1] = last_segment
else:
result.append({ 'start': last_segment['end'], 'end': total_duration, 'gap': True } )
return result
def adjust_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 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, sampling_rate: int = 16000, cache: ModelCache = None):
super().__init__(sampling_rate=sampling_rate)
self.model = None
self.cache = cache
self._initialize_model()
def _initialize_model(self):
if (self.cache is not None):
model_key = "VadSileroTranscription"
self.model, self.get_speech_timestamps = self.cache.get(model_key, self._create_model)
print("Loaded Silerio model from cache.")
else:
self.model, self.get_speech_timestamps = self._create_model()
print("Created Silerio model")
def _create_model(self):
model, utils = torch.hub.load(repo_or_dir='snakers4/silero-vad', model='silero_vad')
# Silero does not benefit from multi-threading
torch.set_num_threads(1) # JIT
(get_speech_timestamps, _, _, _, _) = utils
return model, get_speech_timestamps
def get_transcribe_timestamps(self, audio: str, config: TranscriptionConfig, start_time: float, end_time: float):
result = []
print("Getting timestamps from audio file: {}, start: {}, duration: {}".format(audio, start_time, end_time))
perf_start_time = time.perf_counter()
# Divide procesisng of audio into chunks
chunk_start = start_time
while (chunk_start < end_time):
chunk_duration = min(end_time - chunk_start, VAD_MAX_PROCESSING_CHUNK)
print("Processing VAD in chunk from {} to {}".format(format_timestamp(chunk_start), format_timestamp(chunk_start + chunk_duration)))
wav = self.get_audio_segment(audio, str(chunk_start), str(chunk_duration))
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)
adjusted = self.adjust_timestamp(seconds_timestamps, adjust_seconds=chunk_start, max_source_time=chunk_start + chunk_duration)
#pprint(adjusted)
result.extend(adjusted)
chunk_start += chunk_duration
perf_end_time = time.perf_counter()
print("VAD processing took {} seconds".format(perf_end_time - perf_start_time))
return result
def __getstate__(self):
# We only need the sampling rate
return { 'sampling_rate': self.sampling_rate }
def __setstate__(self, state):
self.sampling_rate = state['sampling_rate']
self.model = None
# Use the global cache
self.cache = GLOBAL_MODEL_CACHE
self._initialize_model()
# A very simple VAD that just marks every N seconds as speech
class VadPeriodicTranscription(AbstractTranscription):
def __init__(self, sampling_rate: int = 16000):
super().__init__(sampling_rate=sampling_rate)
def is_transcribe_timestamps_fast(self):
# This is a very fast VAD - no need to parallelize it
return True
def get_transcribe_timestamps(self, audio: str, config: PeriodicTranscriptionConfig, start_time: float, end_time: float):
result = []
# Generate a timestamp every N seconds
start_timestamp = start_time
while (start_timestamp < end_time):
end_timestamp = min(start_timestamp + config.periodic_duration, end_time)
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 get_audio_duration(file: str):
return float(ffmpeg.probe(file)["format"]["duration"])
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