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import spaces | |
from pydub import AudioSegment | |
import os | |
import torchaudio | |
import torch | |
import re | |
import whisper_timestamped as whisper_ts | |
from typing import Dict | |
from faster_whisper import WhisperModel | |
device = 0 if torch.cuda.is_available() else "cpu" | |
torch_dtype = torch.float32 | |
DEBUG_MODE = True | |
MODEL_PATH_V2 = "langtech-veu/whisper-timestamped-cs" | |
MODEL_PATH_V2_FAST = "langtech-veu/faster-whisper-timestamped-cs" | |
DEVICE = "cuda" if torch.cuda.is_available() else "cpu" | |
print("[INFO] CUDA available:", torch.cuda.is_available()) | |
def clean_text(input_text): | |
remove_chars = ['.', ',', ';', ':', '¿', '?', '«', '»', '-', '¡', '!', '@', | |
'*', '{', '}', '[', ']', '=', '/', '\\', '&', '#', '…'] | |
output_text = ''.join(char if char not in remove_chars else ' ' for char in input_text) | |
return ' '.join(output_text.split()).lower() | |
def split_stereo_channels(audio_path): | |
ext = os.path.splitext(audio_path)[1].lower() | |
if ext == ".wav": | |
audio = AudioSegment.from_wav(audio_path) | |
elif ext == ".mp3": | |
audio = AudioSegment.from_file(audio_path, format="mp3") | |
else: | |
raise ValueError(f"Unsupported file format: {audio_path}") | |
channels = audio.split_to_mono() | |
if len(channels) != 2: | |
raise ValueError(f"Audio {audio_path} does not have 2 channels.") | |
channels[0].export(f"temp_mono_speaker1.wav", format="wav") # Right | |
channels[1].export(f"temp_mono_speaker2.wav", format="wav") # Left | |
def format_audio(audio_path): | |
input_audio, sample_rate = torchaudio.load(audio_path) | |
if input_audio.shape[0] == 2: | |
input_audio = torch.mean(input_audio, dim=0, keepdim=True) | |
resampler = torchaudio.transforms.Resample(orig_freq=sample_rate, new_freq=16000) | |
input_audio = resampler(input_audio) | |
return input_audio.squeeze(), 16000 | |
def post_process_transcription(transcription, max_repeats=2): | |
tokens = re.findall(r'\b\w+\'?\w*\b[.,!?]?', transcription) | |
cleaned_tokens = [] | |
repetition_count = 0 | |
previous_token = None | |
for token in tokens: | |
reduced_token = re.sub(r"(\w{1,3})(\1{2,})", "", token) | |
if reduced_token == previous_token: | |
repetition_count += 1 | |
if repetition_count <= max_repeats: | |
cleaned_tokens.append(reduced_token) | |
else: | |
repetition_count = 1 | |
cleaned_tokens.append(reduced_token) | |
previous_token = reduced_token | |
cleaned_transcription = " ".join(cleaned_tokens) | |
cleaned_transcription = re.sub(r'\s+', ' ', cleaned_transcription).strip() | |
return cleaned_transcription | |
def post_merge_consecutive_segments_from_text(transcription_text: str) -> str: | |
segments = re.split(r'(\[SPEAKER_\d{2}\])', transcription_text) | |
merged_transcription = '' | |
current_speaker = None | |
current_segment = [] | |
for i in range(1, len(segments) - 1, 2): | |
speaker_tag = segments[i] | |
text = segments[i + 1].strip() | |
speaker = re.search(r'\d{2}', speaker_tag).group() | |
if speaker == current_speaker: | |
current_segment.append(text) | |
else: | |
if current_speaker is not None: | |
merged_transcription += f'[SPEAKER_{current_speaker}] {" ".join(current_segment)}\n' | |
current_speaker = speaker | |
current_segment = [text] | |
if current_speaker is not None: | |
merged_transcription += f'[SPEAKER_{current_speaker}] {" ".join(current_segment)}\n' | |
return merged_transcription.strip() | |
def cleanup_temp_files(*file_paths): | |
if DEBUG_MODE: print(f"Entered cleanup_temp_files function...") | |
if DEBUG_MODE: print(f"file_paths: {file_paths}") | |
for path in file_paths: | |
if path and os.path.exists(path): | |
if DEBUG_MODE: print(f"Removing path: {path}") | |
os.remove(path) | |
if DEBUG_MODE: print(f"Exited cleanup_temp_files function.") | |
''' | |
try: | |
faster_model = WhisperModel( | |
MODEL_PATH_V2_FAST, | |
device="cuda" if torch.cuda.is_available() else "cpu", | |
compute_type="float16" if torch.cuda.is_available() else "int8" | |
) | |
except RuntimeError as e: | |
print(f"[WARNING] Failed to load model on GPU: {e}") | |
faster_model = WhisperModel( | |
MODEL_PATH_V2_FAST, | |
device="cpu", | |
compute_type="int8" | |
) | |
''' | |
#faster_model = WhisperModel(MODEL_PATH_V2_FAST, device=DEVICE, compute_type="int8") | |
def load_whisper_model(model_path: str): | |
device = "cuda" if torch.cuda.is_available() else "cpu" | |
model = whisper_ts.load_model(model_path, device=device) | |
return model | |
def transcribe_audio(model, audio_path: str) -> Dict: | |
try: | |
result = whisper_ts.transcribe( | |
model, | |
audio_path, | |
beam_size=5, | |
best_of=5, | |
temperature=(0.0, 0.2, 0.4, 0.6, 0.8, 1.0), | |
vad=False, | |
detect_disfluencies=True, | |
) | |
words = [] | |
for segment in result.get('segments', []): | |
for word in segment.get('words', []): | |
word_text = word.get('word', '').strip() | |
if word_text.startswith(' '): | |
word_text = word_text[1:] | |
words.append({ | |
'word': word_text, | |
'start': word.get('start', 0), | |
'end': word.get('end', 0), | |
'confidence': word.get('confidence', 0) | |
}) | |
return { | |
'audio_path': audio_path, | |
'text': result['text'].strip(), | |
'segments': result.get('segments', []), | |
'words': words, | |
'duration': result.get('duration', 0), | |
'success': True | |
} | |
except Exception as e: | |
return { | |
'audio_path': audio_path, | |
'error': str(e), | |
'success': False | |
} | |
def generate(audio_path, use_v2_fast): | |
if DEBUG_MODE: print(f"Entering generate function...") | |
if DEBUG_MODE: print(f"use_v2_fast: {use_v2_fast}") | |
faster_model = None | |
if use_v2_fast: | |
if torch.cuda.is_available(): | |
try: | |
if DEBUG_MODE: print("[INFO] GPU detected. Loading model on GPU with float16...") | |
faster_model = WhisperModel( | |
MODEL_PATH_V2_FAST, | |
device="cuda", | |
compute_type="float16" | |
) | |
except RuntimeError as e: | |
print(f"[WARNING] Failed to load model on GPU: {e}") | |
if DEBUG_MODE: print("[INFO] Falling back to CPU with int8...") | |
faster_model = WhisperModel( | |
MODEL_PATH_V2_FAST, | |
device="cpu", | |
compute_type="int8" | |
) | |
else: | |
if DEBUG_MODE: print("[INFO] No GPU detected. Loading model on CPU with int8...") | |
faster_model = WhisperModel( | |
MODEL_PATH_V2_FAST, | |
device="cpu", | |
compute_type="int8" | |
) | |
split_stereo_channels(audio_path) | |
left_channel_path = "temp_mono_speaker2.wav" | |
right_channel_path = "temp_mono_speaker1.wav" | |
left_waveform, _ = format_audio(left_channel_path) | |
right_waveform, _ = format_audio(right_channel_path) | |
left_waveform = left_waveform.numpy().astype("float32") | |
right_waveform = right_waveform.numpy().astype("float32") | |
left_result, _ = faster_model.transcribe(left_waveform, beam_size=5, task="transcribe") | |
right_result, _ = faster_model.transcribe(right_waveform, beam_size=5, task="transcribe") | |
left_result = list(left_result) | |
right_result = list(right_result) | |
def get_faster_segments(segments, speaker_label): | |
return [ | |
(seg.start, seg.end, speaker_label, post_process_transcription(seg.text.strip())) | |
for seg in segments if seg.text | |
] | |
left_segs = get_faster_segments(left_result, "Speaker 1") | |
right_segs = get_faster_segments(right_result, "Speaker 2") | |
merged_transcript = sorted( | |
left_segs + right_segs, | |
key=lambda x: float(x[0]) if x[0] is not None else float("inf") | |
) | |
clean_output = "" | |
for start, end, speaker, text in merged_transcript: | |
clean_output += f"[{speaker}]: {text}\n" | |
if DEBUG_MODE: print(f"clean_output: {clean_output}") | |
else: | |
model = load_whisper_model(MODEL_PATH_V2) | |
split_stereo_channels(audio_path) | |
left_channel_path = "temp_mono_speaker2.wav" | |
right_channel_path = "temp_mono_speaker1.wav" | |
left_waveform, _ = format_audio(left_channel_path) | |
right_waveform, _ = format_audio(right_channel_path) | |
left_result = transcribe_audio(model, left_waveform) | |
right_result = transcribe_audio(model, right_waveform) | |
def get_segments(result, speaker_label): | |
segments = result.get("segments", []) | |
if not segments: | |
return [] | |
return [ | |
(seg.get("start", 0.0), seg.get("end", 0.0), speaker_label, | |
post_process_transcription(seg.get("text", "").strip())) | |
for seg in segments if seg.get("text") | |
] | |
left_segs = get_segments(left_result, "Speaker 1") | |
right_segs = get_segments(right_result, "Speaker 2") | |
merged_transcript = sorted( | |
left_segs + right_segs, | |
key=lambda x: float(x[0]) if x[0] is not None else float("inf") | |
) | |
clean_output = "" | |
for start, end, speaker, text in merged_transcript: | |
clean_output += f"[{speaker}]: {text}\n" | |
cleanup_temp_files("temp_mono_speaker1.wav", "temp_mono_speaker2.wav") | |
if DEBUG_MODE: print(f"Exiting generate function...") | |
return clean_output.strip() | |
''' | |
def generate(audio_path, use_v2_fast): | |
if DEBUG_MODE: print(f"Entering generate function...") | |
if DEBUG_MODE: print(f"use_v2_fast: {use_v2_fast}") | |
if use_v2_fast: | |
split_stereo_channels(audio_path) | |
left_channel_path = "temp_mono_speaker2.wav" | |
right_channel_path = "temp_mono_speaker1.wav" | |
left_waveform, left_sr = format_audio(left_channel_path) | |
right_waveform, right_sr = format_audio(right_channel_path) | |
left_waveform = left_waveform.numpy().astype("float32") | |
right_waveform = right_waveform.numpy().astype("float32") | |
left_result, info = faster_model.transcribe(left_waveform, beam_size=5, task="transcribe") | |
right_result, info = faster_model.transcribe(right_waveform, beam_size=5, task="transcribe") | |
left_result = list(left_result) | |
right_result = list(right_result) | |
def get_faster_segments(segments, speaker_label): | |
return [ | |
(seg.start, seg.end, speaker_label, post_process_transcription(seg.text.strip())) | |
for seg in segments if seg.text | |
] | |
left_segs = get_faster_segments(left_result, "Speaker 1") | |
right_segs = get_faster_segments(right_result, "Speaker 2") | |
merged_transcript = sorted( | |
left_segs + right_segs, | |
key=lambda x: float(x[0]) if x[0] is not None else float("inf") | |
) | |
clean_output = "" | |
for start, end, speaker, text in merged_transcript: | |
clean_output += f"[{speaker}]: {text}\n" | |
# FIX Seems that post_merge_consecutive_segments_from_text returns an empty string | |
#clean_output = post_merge_consecutive_segments_from_text(clean_output) | |
#print('clean_output',clean_output) | |
if DEBUG_MODE: print(f"clean_output: {clean_output}") | |
else: | |
model = load_whisper_model(MODEL_PATH_V2) | |
split_stereo_channels(audio_path) | |
left_channel_path = "temp_mono_speaker2.wav" | |
right_channel_path = "temp_mono_speaker1.wav" | |
left_waveform, left_sr = format_audio(left_channel_path) | |
right_waveform, right_sr = format_audio(right_channel_path) | |
left_result = transcribe_audio(model, left_waveform) | |
right_result = transcribe_audio(model, right_waveform) | |
def get_segments(result, speaker_label): | |
segments = result.get("segments", []) | |
if not segments: | |
return [] | |
return [ | |
(seg.get("start", 0.0), seg.get("end", 0.0), speaker_label, post_process_transcription(seg.get("text", "").strip())) | |
for seg in segments if seg.get("text") | |
] | |
left_segs = get_segments(left_result, "Speaker 1") | |
right_segs = get_segments(right_result, "Speaker 2") | |
merged_transcript = sorted( | |
left_segs + right_segs, | |
key=lambda x: float(x[0]) if x[0] is not None else float("inf") | |
) | |
output = "" | |
for start, end, speaker, text in merged_transcript: | |
output += f"[{speaker}]: {text}\n" | |
clean_output = output.strip() | |
if DEBUG_MODE: print(f"Clean output generated.") | |
cleanup_temp_files( | |
"temp_mono_speaker1.wav", | |
"temp_mono_speaker2.wav" | |
) | |
if DEBUG_MODE: print(f"Exiting generate function...") | |
return clean_output | |
''' |