Spaces:
Running
Running
oceansweep
commited on
Commit
•
74055ee
1
Parent(s):
0f1bc91
Upload 3 files
Browse files
App_Function_Libraries/Audio/Audio_Files.py
CHANGED
@@ -19,25 +19,25 @@ import logging
|
|
19 |
import os
|
20 |
import subprocess
|
21 |
import tempfile
|
|
|
22 |
import uuid
|
23 |
from datetime import datetime
|
24 |
from pathlib import Path
|
25 |
-
|
|
|
26 |
import requests
|
27 |
import yt_dlp
|
28 |
-
|
29 |
-
from App_Function_Libraries.Audio.Audio_Transcription_Lib import speech_to_text
|
30 |
-
from App_Function_Libraries.Chunk_Lib import improved_chunking_process
|
31 |
#
|
32 |
# Local Imports
|
33 |
-
from App_Function_Libraries.DB.DB_Manager import
|
34 |
check_media_and_whisper_model
|
35 |
-
from App_Function_Libraries.
|
36 |
-
|
37 |
-
from App_Function_Libraries.Utils.Utils import
|
38 |
-
sanitize_filename
|
39 |
from App_Function_Libraries.Video_DL_Ingestion_Lib import extract_metadata
|
40 |
-
|
|
|
41 |
#
|
42 |
#######################################################################################################################
|
43 |
# Function Definitions
|
@@ -106,168 +106,34 @@ def download_audio_file(url, current_whisper_model="", use_cookies=False, cookie
|
|
106 |
logging.error(f"Unexpected error downloading audio file: {str(e)}")
|
107 |
raise
|
108 |
|
109 |
-
|
110 |
-
def process_audio(
|
111 |
-
audio_file_path,
|
112 |
-
num_speakers=2,
|
113 |
-
whisper_model="small.en",
|
114 |
-
custom_prompt_input=None,
|
115 |
-
offset=0,
|
116 |
-
api_name=None,
|
117 |
-
api_key=None,
|
118 |
-
vad_filter=False,
|
119 |
-
rolling_summarization=False,
|
120 |
-
detail_level=0.01,
|
121 |
-
keywords="default,no_keyword_set",
|
122 |
-
chunk_text_by_words=False,
|
123 |
-
max_words=0,
|
124 |
-
chunk_text_by_sentences=False,
|
125 |
-
max_sentences=0,
|
126 |
-
chunk_text_by_paragraphs=False,
|
127 |
-
max_paragraphs=0,
|
128 |
-
chunk_text_by_tokens=False,
|
129 |
-
max_tokens=0
|
130 |
-
):
|
131 |
-
try:
|
132 |
-
|
133 |
-
# Perform transcription
|
134 |
-
audio_file_path, segments = perform_transcription(audio_file_path, offset, whisper_model, vad_filter)
|
135 |
-
|
136 |
-
if audio_file_path is None or segments is None:
|
137 |
-
logging.error("Process_Audio: Transcription failed or segments not available.")
|
138 |
-
return "Process_Audio: Transcription failed.", None, None, None, None, None
|
139 |
-
|
140 |
-
logging.debug(f"Process_Audio: Transcription audio_file: {audio_file_path}")
|
141 |
-
logging.debug(f"Process_Audio: Transcription segments: {segments}")
|
142 |
-
|
143 |
-
transcription_text = {'audio_file': audio_file_path, 'transcription': segments}
|
144 |
-
logging.debug(f"Process_Audio: Transcription text: {transcription_text}")
|
145 |
-
|
146 |
-
# Save segments to JSON
|
147 |
-
segments_json_path = save_segments_to_json(segments)
|
148 |
-
|
149 |
-
# Perform summarization
|
150 |
-
summary_text = None
|
151 |
-
if api_name:
|
152 |
-
if rolling_summarization is not None:
|
153 |
-
pass
|
154 |
-
# FIXME rolling summarization
|
155 |
-
# summary_text = rolling_summarize_function(
|
156 |
-
# transcription_text,
|
157 |
-
# detail=detail_level,
|
158 |
-
# api_name=api_name,
|
159 |
-
# api_key=api_key,
|
160 |
-
# custom_prompt=custom_prompt_input,
|
161 |
-
# chunk_by_words=chunk_text_by_words,
|
162 |
-
# max_words=max_words,
|
163 |
-
# chunk_by_sentences=chunk_text_by_sentences,
|
164 |
-
# max_sentences=max_sentences,
|
165 |
-
# chunk_by_paragraphs=chunk_text_by_paragraphs,
|
166 |
-
# max_paragraphs=max_paragraphs,
|
167 |
-
# chunk_by_tokens=chunk_text_by_tokens,
|
168 |
-
# max_tokens=max_tokens
|
169 |
-
# )
|
170 |
-
else:
|
171 |
-
summary_text = perform_summarization(api_name, segments_json_path, custom_prompt_input, api_key)
|
172 |
-
|
173 |
-
if summary_text is None:
|
174 |
-
logging.error("Summary text is None. Check summarization function.")
|
175 |
-
summary_file_path = None
|
176 |
-
else:
|
177 |
-
summary_text = 'Summary not available'
|
178 |
-
summary_file_path = None
|
179 |
-
|
180 |
-
# Save transcription and summary
|
181 |
-
download_path = create_download_directory("Audio_Processing")
|
182 |
-
json_file_path, summary_file_path = save_transcription_and_summary(transcription_text, summary_text,
|
183 |
-
download_path)
|
184 |
-
|
185 |
-
# Update function call to add_media_to_database so that it properly applies the title, author and file type
|
186 |
-
# Add to database
|
187 |
-
add_media_to_database(None, {'title': 'Audio File', 'author': 'Unknown'}, segments, summary_text, keywords,
|
188 |
-
custom_prompt_input, whisper_model)
|
189 |
-
|
190 |
-
return transcription_text, summary_text, json_file_path, summary_file_path, None, None
|
191 |
-
|
192 |
-
except Exception as e:
|
193 |
-
logging.error(f"Error in process_audio: {str(e)}")
|
194 |
-
return str(e), None, None, None, None, None
|
195 |
-
|
196 |
-
|
197 |
-
def process_single_audio(audio_file_path, whisper_model, api_name, api_key, keep_original,custom_keywords, source,
|
198 |
-
custom_prompt_input, chunk_method, max_chunk_size, chunk_overlap, use_adaptive_chunking,
|
199 |
-
use_multi_level_chunking, chunk_language):
|
200 |
-
progress = []
|
201 |
-
transcription = ""
|
202 |
-
summary = ""
|
203 |
-
|
204 |
-
def update_progress(message):
|
205 |
-
progress.append(message)
|
206 |
-
return "\n".join(progress)
|
207 |
-
|
208 |
-
try:
|
209 |
-
# Check file size before processing
|
210 |
-
file_size = os.path.getsize(audio_file_path)
|
211 |
-
if file_size > MAX_FILE_SIZE:
|
212 |
-
update_progress(f"File size ({file_size / (1024 * 1024):.2f} MB) exceeds the maximum limit of {MAX_FILE_SIZE / (1024 * 1024):.2f} MB. Skipping this file.")
|
213 |
-
return "\n".join(progress), "", ""
|
214 |
-
|
215 |
-
# Perform transcription
|
216 |
-
update_progress("Starting transcription...")
|
217 |
-
segments = speech_to_text(audio_file_path, whisper_model=whisper_model)
|
218 |
-
transcription = " ".join([segment['Text'] for segment in segments])
|
219 |
-
update_progress("Audio transcribed successfully.")
|
220 |
-
|
221 |
-
# Perform summarization if API is provided
|
222 |
-
if api_name and api_key:
|
223 |
-
update_progress("Starting summarization...")
|
224 |
-
summary = perform_summarization(api_name, transcription, "Summarize the following audio transcript",
|
225 |
-
api_key)
|
226 |
-
update_progress("Audio summarized successfully.")
|
227 |
-
else:
|
228 |
-
summary = "No summary available"
|
229 |
-
|
230 |
-
# Prepare keywords
|
231 |
-
keywords = "audio,transcription"
|
232 |
-
if custom_keywords:
|
233 |
-
keywords += f",{custom_keywords}"
|
234 |
-
|
235 |
-
# Add to database
|
236 |
-
add_media_with_keywords(
|
237 |
-
url=source,
|
238 |
-
title=os.path.basename(audio_file_path),
|
239 |
-
media_type='audio',
|
240 |
-
content=transcription,
|
241 |
-
keywords=keywords,
|
242 |
-
prompt="Summarize the following audio transcript",
|
243 |
-
summary=summary,
|
244 |
-
transcription_model=whisper_model,
|
245 |
-
author="Unknown",
|
246 |
-
ingestion_date=None # This will use the current date
|
247 |
-
)
|
248 |
-
update_progress("Audio file added to database successfully.")
|
249 |
-
|
250 |
-
if not keep_original and source != "Uploaded File":
|
251 |
-
os.remove(audio_file_path)
|
252 |
-
update_progress(f"Temporary file {audio_file_path} removed.")
|
253 |
-
elif keep_original and source != "Uploaded File":
|
254 |
-
update_progress(f"Original audio file kept at: {audio_file_path}")
|
255 |
-
|
256 |
-
except Exception as e:
|
257 |
-
update_progress(f"Error processing {source}: {str(e)}")
|
258 |
-
transcription = f"Error: {str(e)}"
|
259 |
-
summary = "No summary due to error"
|
260 |
-
|
261 |
-
return "\n".join(progress), transcription, summary
|
262 |
-
|
263 |
-
|
264 |
def process_audio_files(audio_urls, audio_file, whisper_model, api_name, api_key, use_cookies, cookies, keep_original,
|
265 |
custom_keywords, custom_prompt_input, chunk_method, max_chunk_size, chunk_overlap,
|
266 |
-
use_adaptive_chunking, use_multi_level_chunking, chunk_language, diarize
|
|
|
|
|
|
|
|
|
|
|
267 |
progress = []
|
268 |
-
temp_files = []
|
269 |
all_transcriptions = []
|
270 |
all_summaries = []
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
271 |
|
272 |
def update_progress(message):
|
273 |
progress.append(message)
|
@@ -335,6 +201,12 @@ def process_audio_files(audio_urls, audio_file, whisper_model, api_name, api_key
|
|
335 |
audio_file_path = download_audio_file(url, use_cookies, cookies)
|
336 |
if not os.path.exists(audio_file_path):
|
337 |
update_progress(f"Downloaded file not found: {audio_file_path}")
|
|
|
|
|
|
|
|
|
|
|
|
|
338 |
continue
|
339 |
|
340 |
temp_files.append(audio_file_path)
|
@@ -344,6 +216,12 @@ def process_audio_files(audio_urls, audio_file, whisper_model, api_name, api_key
|
|
344 |
reencoded_mp3_path = reencode_mp3(audio_file_path)
|
345 |
if not os.path.exists(reencoded_mp3_path):
|
346 |
update_progress(f"Re-encoded file not found: {reencoded_mp3_path}")
|
|
|
|
|
|
|
|
|
|
|
|
|
347 |
continue
|
348 |
|
349 |
temp_files.append(reencoded_mp3_path)
|
@@ -352,6 +230,12 @@ def process_audio_files(audio_urls, audio_file, whisper_model, api_name, api_key
|
|
352 |
wav_file_path = convert_mp3_to_wav(reencoded_mp3_path)
|
353 |
if not os.path.exists(wav_file_path):
|
354 |
update_progress(f"Converted WAV file not found: {wav_file_path}")
|
|
|
|
|
|
|
|
|
|
|
|
|
355 |
continue
|
356 |
|
357 |
temp_files.append(wav_file_path)
|
@@ -370,20 +254,36 @@ def process_audio_files(audio_urls, audio_file, whisper_model, api_name, api_key
|
|
370 |
segments = segments['segments']
|
371 |
|
372 |
if isinstance(segments, list):
|
373 |
-
|
|
|
|
|
|
|
374 |
update_progress("Audio transcribed successfully.")
|
375 |
else:
|
376 |
update_progress("Unexpected segments format received from speech_to_text.")
|
377 |
logging.error(f"Unexpected segments format: {segments}")
|
|
|
|
|
|
|
|
|
|
|
|
|
378 |
continue
|
379 |
|
380 |
if not transcription.strip():
|
381 |
update_progress("Transcription is empty.")
|
|
|
|
|
|
|
|
|
|
|
|
|
382 |
else:
|
383 |
# Apply chunking
|
384 |
chunked_text = improved_chunking_process(transcription, chunk_options)
|
385 |
|
386 |
# Summarize
|
|
|
387 |
if api_name:
|
388 |
try:
|
389 |
summary = perform_summarization(api_name, chunked_text, custom_prompt_input, api_key)
|
@@ -391,16 +291,25 @@ def process_audio_files(audio_urls, audio_file, whisper_model, api_name, api_key
|
|
391 |
except Exception as e:
|
392 |
logging.error(f"Error during summarization: {str(e)}")
|
393 |
summary = "Summary generation failed"
|
|
|
|
|
|
|
|
|
|
|
|
|
394 |
else:
|
395 |
summary = "No summary available (API not provided)"
|
396 |
|
397 |
all_transcriptions.append(transcription)
|
398 |
all_summaries.append(summary)
|
399 |
|
|
|
|
|
|
|
400 |
# Add to database
|
401 |
add_media_with_keywords(
|
402 |
url=url,
|
403 |
-
title=
|
404 |
media_type='audio',
|
405 |
content=transcription,
|
406 |
keywords=custom_keywords,
|
@@ -411,79 +320,129 @@ def process_audio_files(audio_urls, audio_file, whisper_model, api_name, api_key
|
|
411 |
ingestion_date=datetime.now().strftime('%Y-%m-%d')
|
412 |
)
|
413 |
update_progress("Audio file processed and added to database.")
|
|
|
|
|
|
|
|
|
|
|
|
|
414 |
|
415 |
# Process uploaded file if provided
|
416 |
if audio_file:
|
|
|
417 |
if os.path.getsize(audio_file.name) > MAX_FILE_SIZE:
|
418 |
update_progress(
|
419 |
f"Uploaded file size exceeds the maximum limit of {MAX_FILE_SIZE / (1024 * 1024):.2f}MB. Skipping this file.")
|
420 |
else:
|
421 |
-
|
422 |
-
|
423 |
-
|
424 |
-
|
425 |
-
|
426 |
-
|
427 |
-
|
428 |
-
|
429 |
-
|
430 |
-
|
431 |
-
|
432 |
-
|
433 |
-
|
434 |
-
|
435 |
-
|
436 |
-
|
437 |
-
|
438 |
-
|
439 |
-
|
440 |
-
|
441 |
-
|
442 |
-
|
443 |
-
|
444 |
-
|
445 |
-
|
446 |
-
|
447 |
-
segments
|
448 |
-
|
449 |
-
|
450 |
-
|
451 |
-
|
452 |
-
|
453 |
-
|
454 |
-
|
455 |
-
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
456 |
|
457 |
-
|
458 |
-
|
459 |
-
|
460 |
-
|
461 |
-
|
462 |
-
|
463 |
-
|
464 |
-
|
465 |
-
summary = "No summary available (API not provided)"
|
466 |
|
467 |
-
|
468 |
-
|
|
|
|
|
|
|
|
|
469 |
|
470 |
-
|
471 |
-
|
472 |
-
|
473 |
-
|
474 |
-
|
475 |
-
keywords=custom_keywords,
|
476 |
-
prompt=custom_prompt_input,
|
477 |
-
summary=summary,
|
478 |
-
transcription_model=whisper_model,
|
479 |
-
author="Unknown",
|
480 |
-
ingestion_date=datetime.now().strftime('%Y-%m-%d')
|
481 |
-
)
|
482 |
-
update_progress("Uploaded file processed and added to database.")
|
483 |
|
484 |
-
# Final cleanup
|
485 |
-
if not keep_original:
|
486 |
-
cleanup_files()
|
487 |
|
488 |
final_progress = update_progress("All processing complete.")
|
489 |
final_transcriptions = "\n\n".join(all_transcriptions)
|
@@ -493,10 +452,39 @@ def process_audio_files(audio_urls, audio_file, whisper_model, api_name, api_key
|
|
493 |
|
494 |
except Exception as e:
|
495 |
logging.error(f"Error processing audio files: {str(e)}")
|
|
|
|
|
|
|
|
|
|
|
496 |
cleanup_files()
|
497 |
return update_progress(f"Processing failed: {str(e)}"), "", ""
|
498 |
|
499 |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
500 |
def download_youtube_audio(url):
|
501 |
try:
|
502 |
# Determine ffmpeg path based on the operating system.
|
@@ -564,12 +552,55 @@ def download_youtube_audio(url):
|
|
564 |
def process_podcast(url, title, author, keywords, custom_prompt, api_name, api_key, whisper_model,
|
565 |
keep_original=False, enable_diarization=False, use_cookies=False, cookies=None,
|
566 |
chunk_method=None, max_chunk_size=300, chunk_overlap=0, use_adaptive_chunking=False,
|
567 |
-
use_multi_level_chunking=False, chunk_language='english'):
|
568 |
-
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
569 |
error_message = ""
|
570 |
temp_files = []
|
571 |
|
|
|
|
|
|
|
|
|
|
|
|
|
572 |
def update_progress(message):
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
573 |
progress.append(message)
|
574 |
return "\n".join(progress)
|
575 |
|
@@ -583,13 +614,21 @@ def process_podcast(url, title, author, keywords, custom_prompt, api_name, api_k
|
|
583 |
except Exception as e:
|
584 |
update_progress(f"Failed to remove temporary file {file}: {str(e)}")
|
585 |
|
|
|
|
|
586 |
try:
|
587 |
-
#
|
588 |
-
|
|
|
|
|
|
|
|
|
|
|
|
|
589 |
temp_files.append(audio_file)
|
590 |
update_progress("Podcast downloaded successfully.")
|
591 |
|
592 |
-
# Extract metadata
|
593 |
metadata = extract_metadata(url)
|
594 |
title = title or metadata.get('title', 'Unknown Podcast')
|
595 |
author = author or metadata.get('uploader', 'Unknown Author')
|
@@ -607,7 +646,7 @@ Duration: {metadata.get('duration', 'N/A')} seconds
|
|
607 |
Description: {metadata.get('description', 'N/A')}
|
608 |
"""
|
609 |
|
610 |
-
# Update keywords
|
611 |
new_keywords = []
|
612 |
if metadata.get('series'):
|
613 |
new_keywords.append(f"series:{metadata['series']}")
|
@@ -617,22 +656,36 @@ Description: {metadata.get('description', 'N/A')}
|
|
617 |
new_keywords.append(f"season:{metadata['season']}")
|
618 |
|
619 |
keywords = f"{keywords},{','.join(new_keywords)}" if keywords else ','.join(new_keywords)
|
620 |
-
|
621 |
update_progress(f"Metadata extracted - Title: {title}, Author: {author}, Keywords: {keywords}")
|
622 |
|
623 |
-
# Transcribe the podcast
|
624 |
try:
|
625 |
if enable_diarization:
|
626 |
segments = speech_to_text(audio_file, whisper_model=whisper_model, diarize=True)
|
627 |
else:
|
628 |
segments = speech_to_text(audio_file, whisper_model=whisper_model)
|
629 |
-
|
630 |
-
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
631 |
except Exception as e:
|
632 |
error_message = f"Transcription failed: {str(e)}"
|
633 |
-
raise
|
634 |
|
635 |
-
# Apply chunking
|
636 |
chunk_options = {
|
637 |
'method': chunk_method,
|
638 |
'max_size': max_chunk_size,
|
@@ -646,17 +699,19 @@ Description: {metadata.get('description', 'N/A')}
|
|
646 |
# Combine metadata and transcription
|
647 |
full_content = metadata_text + "\n\nTranscription:\n" + transcription
|
648 |
|
649 |
-
# Summarize if API is provided
|
650 |
summary = None
|
651 |
-
if api_name
|
652 |
try:
|
653 |
summary = perform_summarization(api_name, chunked_text, custom_prompt, api_key)
|
654 |
update_progress("Podcast summarized successfully.")
|
655 |
except Exception as e:
|
656 |
error_message = f"Summarization failed: {str(e)}"
|
657 |
-
raise
|
|
|
|
|
658 |
|
659 |
-
# Add to database
|
660 |
try:
|
661 |
add_media_with_keywords(
|
662 |
url=url,
|
@@ -673,18 +728,57 @@ Description: {metadata.get('description', 'N/A')}
|
|
673 |
update_progress("Podcast added to database successfully.")
|
674 |
except Exception as e:
|
675 |
error_message = f"Error adding podcast to database: {str(e)}"
|
676 |
-
raise
|
677 |
|
678 |
-
# Cleanup
|
679 |
cleanup_files()
|
680 |
|
681 |
-
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
682 |
title, author, keywords, error_message)
|
683 |
|
684 |
except Exception as e:
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
685 |
logging.error(f"Error processing podcast: {str(e)}")
|
686 |
cleanup_files()
|
687 |
-
|
|
|
688 |
|
689 |
|
690 |
#
|
|
|
19 |
import os
|
20 |
import subprocess
|
21 |
import tempfile
|
22 |
+
import time
|
23 |
import uuid
|
24 |
from datetime import datetime
|
25 |
from pathlib import Path
|
26 |
+
#
|
27 |
+
# External Imports
|
28 |
import requests
|
29 |
import yt_dlp
|
|
|
|
|
|
|
30 |
#
|
31 |
# Local Imports
|
32 |
+
from App_Function_Libraries.DB.DB_Manager import add_media_with_keywords, \
|
33 |
check_media_and_whisper_model
|
34 |
+
from App_Function_Libraries.Metrics.metrics_logger import log_counter, log_histogram
|
35 |
+
from App_Function_Libraries.Summarization.Summarization_General_Lib import perform_summarization
|
36 |
+
from App_Function_Libraries.Utils.Utils import downloaded_files, \
|
37 |
+
sanitize_filename, generate_unique_id, temp_files
|
38 |
from App_Function_Libraries.Video_DL_Ingestion_Lib import extract_metadata
|
39 |
+
from App_Function_Libraries.Audio.Audio_Transcription_Lib import speech_to_text
|
40 |
+
from App_Function_Libraries.Chunk_Lib import improved_chunking_process
|
41 |
#
|
42 |
#######################################################################################################################
|
43 |
# Function Definitions
|
|
|
106 |
logging.error(f"Unexpected error downloading audio file: {str(e)}")
|
107 |
raise
|
108 |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
109 |
def process_audio_files(audio_urls, audio_file, whisper_model, api_name, api_key, use_cookies, cookies, keep_original,
|
110 |
custom_keywords, custom_prompt_input, chunk_method, max_chunk_size, chunk_overlap,
|
111 |
+
use_adaptive_chunking, use_multi_level_chunking, chunk_language, diarize,
|
112 |
+
keep_timestamps, custom_title):
|
113 |
+
|
114 |
+
start_time = time.time() # Start time for processing
|
115 |
+
processed_count = 0
|
116 |
+
failed_count = 0
|
117 |
progress = []
|
|
|
118 |
all_transcriptions = []
|
119 |
all_summaries = []
|
120 |
+
#v2
|
121 |
+
def format_transcription_with_timestamps(segments):
|
122 |
+
if keep_timestamps:
|
123 |
+
formatted_segments = []
|
124 |
+
for segment in segments:
|
125 |
+
start = segment.get('Time_Start', 0)
|
126 |
+
end = segment.get('Time_End', 0)
|
127 |
+
text = segment.get('Text', '').strip() # Ensure text is stripped of leading/trailing spaces
|
128 |
+
|
129 |
+
# Add the formatted timestamp and text to the list, followed by a newline
|
130 |
+
formatted_segments.append(f"[{start:.2f}-{end:.2f}] {text}")
|
131 |
+
|
132 |
+
# Join the segments with a newline to ensure proper formatting
|
133 |
+
return "\n".join(formatted_segments)
|
134 |
+
else:
|
135 |
+
# Join the text without timestamps
|
136 |
+
return "\n".join([segment.get('Text', '').strip() for segment in segments])
|
137 |
|
138 |
def update_progress(message):
|
139 |
progress.append(message)
|
|
|
201 |
audio_file_path = download_audio_file(url, use_cookies, cookies)
|
202 |
if not os.path.exists(audio_file_path):
|
203 |
update_progress(f"Downloaded file not found: {audio_file_path}")
|
204 |
+
failed_count += 1
|
205 |
+
log_counter(
|
206 |
+
metric_name="audio_files_failed_total",
|
207 |
+
labels={"whisper_model": whisper_model, "api_name": api_name},
|
208 |
+
value=1
|
209 |
+
)
|
210 |
continue
|
211 |
|
212 |
temp_files.append(audio_file_path)
|
|
|
216 |
reencoded_mp3_path = reencode_mp3(audio_file_path)
|
217 |
if not os.path.exists(reencoded_mp3_path):
|
218 |
update_progress(f"Re-encoded file not found: {reencoded_mp3_path}")
|
219 |
+
failed_count += 1
|
220 |
+
log_counter(
|
221 |
+
metric_name="audio_files_failed_total",
|
222 |
+
labels={"whisper_model": whisper_model, "api_name": api_name},
|
223 |
+
value=1
|
224 |
+
)
|
225 |
continue
|
226 |
|
227 |
temp_files.append(reencoded_mp3_path)
|
|
|
230 |
wav_file_path = convert_mp3_to_wav(reencoded_mp3_path)
|
231 |
if not os.path.exists(wav_file_path):
|
232 |
update_progress(f"Converted WAV file not found: {wav_file_path}")
|
233 |
+
failed_count += 1
|
234 |
+
log_counter(
|
235 |
+
metric_name="audio_files_failed_total",
|
236 |
+
labels={"whisper_model": whisper_model, "api_name": api_name},
|
237 |
+
value=1
|
238 |
+
)
|
239 |
continue
|
240 |
|
241 |
temp_files.append(wav_file_path)
|
|
|
254 |
segments = segments['segments']
|
255 |
|
256 |
if isinstance(segments, list):
|
257 |
+
# Log first 5 segments for debugging
|
258 |
+
logging.debug(f"Segments before formatting: {segments[:5]}")
|
259 |
+
transcription = format_transcription_with_timestamps(segments)
|
260 |
+
logging.debug(f"Formatted transcription (first 500 chars): {transcription[:500]}")
|
261 |
update_progress("Audio transcribed successfully.")
|
262 |
else:
|
263 |
update_progress("Unexpected segments format received from speech_to_text.")
|
264 |
logging.error(f"Unexpected segments format: {segments}")
|
265 |
+
failed_count += 1
|
266 |
+
log_counter(
|
267 |
+
metric_name="audio_files_failed_total",
|
268 |
+
labels={"whisper_model": whisper_model, "api_name": api_name},
|
269 |
+
value=1
|
270 |
+
)
|
271 |
continue
|
272 |
|
273 |
if not transcription.strip():
|
274 |
update_progress("Transcription is empty.")
|
275 |
+
failed_count += 1
|
276 |
+
log_counter(
|
277 |
+
metric_name="audio_files_failed_total",
|
278 |
+
labels={"whisper_model": whisper_model, "api_name": api_name},
|
279 |
+
value=1
|
280 |
+
)
|
281 |
else:
|
282 |
# Apply chunking
|
283 |
chunked_text = improved_chunking_process(transcription, chunk_options)
|
284 |
|
285 |
# Summarize
|
286 |
+
logging.debug(f"Audio Transcription API Name: {api_name}")
|
287 |
if api_name:
|
288 |
try:
|
289 |
summary = perform_summarization(api_name, chunked_text, custom_prompt_input, api_key)
|
|
|
291 |
except Exception as e:
|
292 |
logging.error(f"Error during summarization: {str(e)}")
|
293 |
summary = "Summary generation failed"
|
294 |
+
failed_count += 1
|
295 |
+
log_counter(
|
296 |
+
metric_name="audio_files_failed_total",
|
297 |
+
labels={"whisper_model": whisper_model, "api_name": api_name},
|
298 |
+
value=1
|
299 |
+
)
|
300 |
else:
|
301 |
summary = "No summary available (API not provided)"
|
302 |
|
303 |
all_transcriptions.append(transcription)
|
304 |
all_summaries.append(summary)
|
305 |
|
306 |
+
# Use custom_title if provided, otherwise use the original filename
|
307 |
+
title = custom_title if custom_title else os.path.basename(wav_file_path)
|
308 |
+
|
309 |
# Add to database
|
310 |
add_media_with_keywords(
|
311 |
url=url,
|
312 |
+
title=title,
|
313 |
media_type='audio',
|
314 |
content=transcription,
|
315 |
keywords=custom_keywords,
|
|
|
320 |
ingestion_date=datetime.now().strftime('%Y-%m-%d')
|
321 |
)
|
322 |
update_progress("Audio file processed and added to database.")
|
323 |
+
processed_count += 1
|
324 |
+
log_counter(
|
325 |
+
metric_name="audio_files_processed_total",
|
326 |
+
labels={"whisper_model": whisper_model, "api_name": api_name},
|
327 |
+
value=1
|
328 |
+
)
|
329 |
|
330 |
# Process uploaded file if provided
|
331 |
if audio_file:
|
332 |
+
url = generate_unique_id()
|
333 |
if os.path.getsize(audio_file.name) > MAX_FILE_SIZE:
|
334 |
update_progress(
|
335 |
f"Uploaded file size exceeds the maximum limit of {MAX_FILE_SIZE / (1024 * 1024):.2f}MB. Skipping this file.")
|
336 |
else:
|
337 |
+
try:
|
338 |
+
# Re-encode MP3 to fix potential issues
|
339 |
+
reencoded_mp3_path = reencode_mp3(audio_file.name)
|
340 |
+
if not os.path.exists(reencoded_mp3_path):
|
341 |
+
update_progress(f"Re-encoded file not found: {reencoded_mp3_path}")
|
342 |
+
return update_progress("Processing failed: Re-encoded file not found"), "", ""
|
343 |
+
|
344 |
+
temp_files.append(reencoded_mp3_path)
|
345 |
+
|
346 |
+
# Convert re-encoded MP3 to WAV
|
347 |
+
wav_file_path = convert_mp3_to_wav(reencoded_mp3_path)
|
348 |
+
if not os.path.exists(wav_file_path):
|
349 |
+
update_progress(f"Converted WAV file not found: {wav_file_path}")
|
350 |
+
return update_progress("Processing failed: Converted WAV file not found"), "", ""
|
351 |
+
|
352 |
+
temp_files.append(wav_file_path)
|
353 |
+
|
354 |
+
# Initialize transcription
|
355 |
+
transcription = ""
|
356 |
+
|
357 |
+
if diarize:
|
358 |
+
segments = speech_to_text(wav_file_path, whisper_model=whisper_model, diarize=True)
|
359 |
+
else:
|
360 |
+
segments = speech_to_text(wav_file_path, whisper_model=whisper_model)
|
361 |
+
|
362 |
+
# Handle segments nested under 'segments' key
|
363 |
+
if isinstance(segments, dict) and 'segments' in segments:
|
364 |
+
segments = segments['segments']
|
365 |
+
|
366 |
+
if isinstance(segments, list):
|
367 |
+
transcription = format_transcription_with_timestamps(segments)
|
368 |
+
else:
|
369 |
+
update_progress("Unexpected segments format received from speech_to_text.")
|
370 |
+
logging.error(f"Unexpected segments format: {segments}")
|
371 |
+
|
372 |
+
chunked_text = improved_chunking_process(transcription, chunk_options)
|
373 |
+
|
374 |
+
logging.debug(f"Audio Transcription API Name: {api_name}")
|
375 |
+
if api_name:
|
376 |
+
try:
|
377 |
+
summary = perform_summarization(api_name, chunked_text, custom_prompt_input, api_key)
|
378 |
+
update_progress("Audio summarized successfully.")
|
379 |
+
except Exception as e:
|
380 |
+
logging.error(f"Error during summarization: {str(e)}")
|
381 |
+
summary = "Summary generation failed"
|
382 |
+
else:
|
383 |
+
summary = "No summary available (API not provided)"
|
384 |
+
|
385 |
+
all_transcriptions.append(transcription)
|
386 |
+
all_summaries.append(summary)
|
387 |
+
|
388 |
+
# Use custom_title if provided, otherwise use the original filename
|
389 |
+
title = custom_title if custom_title else os.path.basename(wav_file_path)
|
390 |
+
|
391 |
+
add_media_with_keywords(
|
392 |
+
url="Uploaded File",
|
393 |
+
title=title,
|
394 |
+
media_type='audio',
|
395 |
+
content=transcription,
|
396 |
+
keywords=custom_keywords,
|
397 |
+
prompt=custom_prompt_input,
|
398 |
+
summary=summary,
|
399 |
+
transcription_model=whisper_model,
|
400 |
+
author="Unknown",
|
401 |
+
ingestion_date=datetime.now().strftime('%Y-%m-%d')
|
402 |
+
)
|
403 |
+
update_progress("Uploaded file processed and added to database.")
|
404 |
+
processed_count += 1
|
405 |
+
log_counter(
|
406 |
+
metric_name="audio_files_processed_total",
|
407 |
+
labels={"whisper_model": whisper_model, "api_name": api_name},
|
408 |
+
value=1
|
409 |
+
)
|
410 |
+
except Exception as e:
|
411 |
+
update_progress(f"Error processing uploaded file: {str(e)}")
|
412 |
+
logging.error(f"Error processing uploaded file: {str(e)}")
|
413 |
+
failed_count += 1
|
414 |
+
log_counter(
|
415 |
+
metric_name="audio_files_failed_total",
|
416 |
+
labels={"whisper_model": whisper_model, "api_name": api_name},
|
417 |
+
value=1
|
418 |
+
)
|
419 |
+
return update_progress("Processing failed: Error processing uploaded file"), "", ""
|
420 |
+
# Final cleanup
|
421 |
+
if not keep_original:
|
422 |
+
cleanup_files()
|
423 |
|
424 |
+
end_time = time.time()
|
425 |
+
processing_time = end_time - start_time
|
426 |
+
# Log processing time
|
427 |
+
log_histogram(
|
428 |
+
metric_name="audio_processing_time_seconds",
|
429 |
+
value=processing_time,
|
430 |
+
labels={"whisper_model": whisper_model, "api_name": api_name}
|
431 |
+
)
|
|
|
432 |
|
433 |
+
# Optionally, log total counts
|
434 |
+
log_counter(
|
435 |
+
metric_name="total_audio_files_processed",
|
436 |
+
labels={"whisper_model": whisper_model, "api_name": api_name},
|
437 |
+
value=processed_count
|
438 |
+
)
|
439 |
|
440 |
+
log_counter(
|
441 |
+
metric_name="total_audio_files_failed",
|
442 |
+
labels={"whisper_model": whisper_model, "api_name": api_name},
|
443 |
+
value=failed_count
|
444 |
+
)
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
445 |
|
|
|
|
|
|
|
446 |
|
447 |
final_progress = update_progress("All processing complete.")
|
448 |
final_transcriptions = "\n\n".join(all_transcriptions)
|
|
|
452 |
|
453 |
except Exception as e:
|
454 |
logging.error(f"Error processing audio files: {str(e)}")
|
455 |
+
log_counter(
|
456 |
+
metric_name="audio_files_failed_total",
|
457 |
+
labels={"whisper_model": whisper_model, "api_name": api_name},
|
458 |
+
value=1
|
459 |
+
)
|
460 |
cleanup_files()
|
461 |
return update_progress(f"Processing failed: {str(e)}"), "", ""
|
462 |
|
463 |
|
464 |
+
def format_transcription_with_timestamps(segments, keep_timestamps):
|
465 |
+
"""
|
466 |
+
Formats the transcription segments with or without timestamps.
|
467 |
+
|
468 |
+
Parameters:
|
469 |
+
segments (list): List of transcription segments.
|
470 |
+
keep_timestamps (bool): Whether to include timestamps.
|
471 |
+
|
472 |
+
Returns:
|
473 |
+
str: Formatted transcription.
|
474 |
+
"""
|
475 |
+
if keep_timestamps:
|
476 |
+
formatted_segments = []
|
477 |
+
for segment in segments:
|
478 |
+
start = segment.get('Time_Start', 0)
|
479 |
+
end = segment.get('Time_End', 0)
|
480 |
+
text = segment.get('Text', '').strip()
|
481 |
+
|
482 |
+
formatted_segments.append(f"[{start:.2f}-{end:.2f}] {text}")
|
483 |
+
return "\n".join(formatted_segments)
|
484 |
+
else:
|
485 |
+
return "\n".join([segment.get('Text', '').strip() for segment in segments])
|
486 |
+
|
487 |
+
|
488 |
def download_youtube_audio(url):
|
489 |
try:
|
490 |
# Determine ffmpeg path based on the operating system.
|
|
|
552 |
def process_podcast(url, title, author, keywords, custom_prompt, api_name, api_key, whisper_model,
|
553 |
keep_original=False, enable_diarization=False, use_cookies=False, cookies=None,
|
554 |
chunk_method=None, max_chunk_size=300, chunk_overlap=0, use_adaptive_chunking=False,
|
555 |
+
use_multi_level_chunking=False, chunk_language='english', keep_timestamps=True):
|
556 |
+
"""
|
557 |
+
Processes a podcast by downloading the audio, transcribing it, summarizing the transcription,
|
558 |
+
and adding the results to the database. Metrics are logged throughout the process.
|
559 |
+
|
560 |
+
Parameters:
|
561 |
+
url (str): URL of the podcast.
|
562 |
+
title (str): Title of the podcast.
|
563 |
+
author (str): Author of the podcast.
|
564 |
+
keywords (str): Comma-separated keywords.
|
565 |
+
custom_prompt (str): Custom prompt for summarization.
|
566 |
+
api_name (str): API name for summarization.
|
567 |
+
api_key (str): API key for summarization.
|
568 |
+
whisper_model (str): Whisper model to use for transcription.
|
569 |
+
keep_original (bool): Whether to keep the original audio file.
|
570 |
+
enable_diarization (bool): Whether to enable speaker diarization.
|
571 |
+
use_cookies (bool): Whether to use cookies for authenticated downloads.
|
572 |
+
cookies (str): JSON-formatted cookies string.
|
573 |
+
chunk_method (str): Method for chunking text.
|
574 |
+
max_chunk_size (int): Maximum size for each text chunk.
|
575 |
+
chunk_overlap (int): Overlap size between chunks.
|
576 |
+
use_adaptive_chunking (bool): Whether to use adaptive chunking.
|
577 |
+
use_multi_level_chunking (bool): Whether to use multi-level chunking.
|
578 |
+
chunk_language (str): Language for chunking.
|
579 |
+
keep_timestamps (bool): Whether to keep timestamps in transcription.
|
580 |
+
|
581 |
+
Returns:
|
582 |
+
tuple: (progress_message, transcription, summary, title, author, keywords, error_message)
|
583 |
+
"""
|
584 |
+
start_time = time.time() # Start time for processing
|
585 |
error_message = ""
|
586 |
temp_files = []
|
587 |
|
588 |
+
# Define labels for metrics
|
589 |
+
labels = {
|
590 |
+
"whisper_model": whisper_model,
|
591 |
+
"api_name": api_name if api_name else "None"
|
592 |
+
}
|
593 |
+
|
594 |
def update_progress(message):
|
595 |
+
"""
|
596 |
+
Updates the progress messages.
|
597 |
+
|
598 |
+
Parameters:
|
599 |
+
message (str): Progress message to append.
|
600 |
+
|
601 |
+
Returns:
|
602 |
+
str: Combined progress messages.
|
603 |
+
"""
|
604 |
progress.append(message)
|
605 |
return "\n".join(progress)
|
606 |
|
|
|
614 |
except Exception as e:
|
615 |
update_progress(f"Failed to remove temporary file {file}: {str(e)}")
|
616 |
|
617 |
+
progress = [] # Initialize progress messages
|
618 |
+
|
619 |
try:
|
620 |
+
# Handle cookies if required
|
621 |
+
if use_cookies:
|
622 |
+
cookies = json.loads(cookies)
|
623 |
+
|
624 |
+
# Download the podcast audio file
|
625 |
+
audio_file = download_audio_file(url, whisper_model, use_cookies, cookies)
|
626 |
+
if not audio_file:
|
627 |
+
raise RuntimeError("Failed to download podcast audio.")
|
628 |
temp_files.append(audio_file)
|
629 |
update_progress("Podcast downloaded successfully.")
|
630 |
|
631 |
+
# Extract metadata from the podcast
|
632 |
metadata = extract_metadata(url)
|
633 |
title = title or metadata.get('title', 'Unknown Podcast')
|
634 |
author = author or metadata.get('uploader', 'Unknown Author')
|
|
|
646 |
Description: {metadata.get('description', 'N/A')}
|
647 |
"""
|
648 |
|
649 |
+
# Update keywords with metadata information
|
650 |
new_keywords = []
|
651 |
if metadata.get('series'):
|
652 |
new_keywords.append(f"series:{metadata['series']}")
|
|
|
656 |
new_keywords.append(f"season:{metadata['season']}")
|
657 |
|
658 |
keywords = f"{keywords},{','.join(new_keywords)}" if keywords else ','.join(new_keywords)
|
|
|
659 |
update_progress(f"Metadata extracted - Title: {title}, Author: {author}, Keywords: {keywords}")
|
660 |
|
661 |
+
# Transcribe the podcast audio
|
662 |
try:
|
663 |
if enable_diarization:
|
664 |
segments = speech_to_text(audio_file, whisper_model=whisper_model, diarize=True)
|
665 |
else:
|
666 |
segments = speech_to_text(audio_file, whisper_model=whisper_model)
|
667 |
+
# SEems like this could be optimized... FIXME
|
668 |
+
def format_segment(segment):
|
669 |
+
start = segment.get('start', 0)
|
670 |
+
end = segment.get('end', 0)
|
671 |
+
text = segment.get('Text', '')
|
672 |
+
|
673 |
+
if isinstance(segments, dict) and 'segments' in segments:
|
674 |
+
segments = segments['segments']
|
675 |
+
|
676 |
+
if isinstance(segments, list):
|
677 |
+
transcription = format_transcription_with_timestamps(segments, keep_timestamps)
|
678 |
+
update_progress("Podcast transcribed successfully.")
|
679 |
+
else:
|
680 |
+
raise ValueError("Unexpected segments format received from speech_to_text.")
|
681 |
+
|
682 |
+
if not transcription.strip():
|
683 |
+
raise ValueError("Transcription is empty.")
|
684 |
except Exception as e:
|
685 |
error_message = f"Transcription failed: {str(e)}"
|
686 |
+
raise RuntimeError(error_message)
|
687 |
|
688 |
+
# Apply chunking to the transcription
|
689 |
chunk_options = {
|
690 |
'method': chunk_method,
|
691 |
'max_size': max_chunk_size,
|
|
|
699 |
# Combine metadata and transcription
|
700 |
full_content = metadata_text + "\n\nTranscription:\n" + transcription
|
701 |
|
702 |
+
# Summarize the transcription if API is provided
|
703 |
summary = None
|
704 |
+
if api_name:
|
705 |
try:
|
706 |
summary = perform_summarization(api_name, chunked_text, custom_prompt, api_key)
|
707 |
update_progress("Podcast summarized successfully.")
|
708 |
except Exception as e:
|
709 |
error_message = f"Summarization failed: {str(e)}"
|
710 |
+
raise RuntimeError(error_message)
|
711 |
+
else:
|
712 |
+
summary = "No summary available (API not provided)"
|
713 |
|
714 |
+
# Add the processed podcast to the database
|
715 |
try:
|
716 |
add_media_with_keywords(
|
717 |
url=url,
|
|
|
728 |
update_progress("Podcast added to database successfully.")
|
729 |
except Exception as e:
|
730 |
error_message = f"Error adding podcast to database: {str(e)}"
|
731 |
+
raise RuntimeError(error_message)
|
732 |
|
733 |
+
# Cleanup temporary files if required
|
734 |
cleanup_files()
|
735 |
|
736 |
+
# Calculate processing time
|
737 |
+
end_time = time.time()
|
738 |
+
processing_time = end_time - start_time
|
739 |
+
|
740 |
+
# Log successful processing
|
741 |
+
log_counter(
|
742 |
+
metric_name="podcasts_processed_total",
|
743 |
+
labels=labels,
|
744 |
+
value=1
|
745 |
+
)
|
746 |
+
|
747 |
+
# Log processing time
|
748 |
+
log_histogram(
|
749 |
+
metric_name="podcast_processing_time_seconds",
|
750 |
+
value=processing_time,
|
751 |
+
labels=labels
|
752 |
+
)
|
753 |
+
|
754 |
+
# Return the final outputs
|
755 |
+
final_progress = update_progress("Processing complete.")
|
756 |
+
return (final_progress, full_content, summary or "No summary generated.",
|
757 |
title, author, keywords, error_message)
|
758 |
|
759 |
except Exception as e:
|
760 |
+
# Calculate processing time up to the point of failure
|
761 |
+
end_time = time.time()
|
762 |
+
processing_time = end_time - start_time
|
763 |
+
|
764 |
+
# Log failed processing
|
765 |
+
log_counter(
|
766 |
+
metric_name="podcasts_failed_total",
|
767 |
+
labels=labels,
|
768 |
+
value=1
|
769 |
+
)
|
770 |
+
|
771 |
+
# Log processing time even on failure
|
772 |
+
log_histogram(
|
773 |
+
metric_name="podcast_processing_time_seconds",
|
774 |
+
value=processing_time,
|
775 |
+
labels=labels
|
776 |
+
)
|
777 |
+
|
778 |
logging.error(f"Error processing podcast: {str(e)}")
|
779 |
cleanup_files()
|
780 |
+
final_progress = update_progress(f"Processing failed: {str(e)}")
|
781 |
+
return (final_progress, "", "", "", "", "", str(e))
|
782 |
|
783 |
|
784 |
#
|
App_Function_Libraries/Audio/Audio_Transcription_Lib.py
CHANGED
@@ -1,277 +1,335 @@
|
|
1 |
-
# Audio_Transcription_Lib.py
|
2 |
-
#########################################
|
3 |
-
# Transcription Library
|
4 |
-
# This library is used to perform transcription of audio files.
|
5 |
-
# Currently, uses faster_whisper for transcription.
|
6 |
-
#
|
7 |
-
####################
|
8 |
-
# Function List
|
9 |
-
#
|
10 |
-
# 1. convert_to_wav(video_file_path, offset=0, overwrite=False)
|
11 |
-
# 2. speech_to_text(audio_file_path, selected_source_lang='en', whisper_model='small.en', vad_filter=False)
|
12 |
-
#
|
13 |
-
####################
|
14 |
-
#
|
15 |
-
# Import necessary libraries to run solo for testing
|
16 |
-
import gc
|
17 |
-
import json
|
18 |
-
import logging
|
19 |
-
import
|
20 |
-
import
|
21 |
-
import
|
22 |
-
import
|
23 |
-
import
|
24 |
-
import
|
25 |
-
import
|
26 |
-
|
27 |
-
#
|
28 |
-
#import
|
29 |
-
|
30 |
-
from
|
31 |
-
|
32 |
-
#
|
33 |
-
|
34 |
-
|
35 |
-
|
36 |
-
#
|
37 |
-
|
38 |
-
|
39 |
-
#
|
40 |
-
|
41 |
-
#
|
42 |
-
#
|
43 |
-
|
44 |
-
|
45 |
-
|
46 |
-
|
47 |
-
|
48 |
-
|
49 |
-
|
50 |
-
|
51 |
-
|
52 |
-
|
53 |
-
|
54 |
-
|
55 |
-
|
56 |
-
|
57 |
-
|
58 |
-
"
|
59 |
-
|
60 |
-
|
61 |
-
|
62 |
-
|
63 |
-
|
64 |
-
|
65 |
-
|
66 |
-
|
67 |
-
|
68 |
-
|
69 |
-
|
70 |
-
|
71 |
-
|
72 |
-
|
73 |
-
|
74 |
-
|
75 |
-
|
76 |
-
|
77 |
-
|
78 |
-
|
79 |
-
|
80 |
-
|
81 |
-
#
|
82 |
-
|
83 |
-
|
84 |
-
|
85 |
-
|
86 |
-
|
87 |
-
|
88 |
-
|
89 |
-
|
90 |
-
|
91 |
-
|
92 |
-
|
93 |
-
|
94 |
-
|
95 |
-
|
96 |
-
|
97 |
-
|
98 |
-
|
99 |
-
|
100 |
-
|
101 |
-
|
102 |
-
|
103 |
-
|
104 |
-
|
105 |
-
|
106 |
-
#
|
107 |
-
|
108 |
-
|
109 |
-
|
110 |
-
|
111 |
-
|
112 |
-
|
113 |
-
|
114 |
-
|
115 |
-
|
116 |
-
|
117 |
-
|
118 |
-
#
|
119 |
-
#
|
120 |
-
|
121 |
-
|
122 |
-
|
123 |
-
|
124 |
-
|
125 |
-
|
126 |
-
|
127 |
-
|
128 |
-
|
129 |
-
|
130 |
-
|
131 |
-
|
132 |
-
|
133 |
-
|
134 |
-
|
135 |
-
|
136 |
-
|
137 |
-
|
138 |
-
|
139 |
-
|
140 |
-
|
141 |
-
|
142 |
-
|
143 |
-
|
144 |
-
|
145 |
-
|
146 |
-
|
147 |
-
|
148 |
-
|
149 |
-
|
150 |
-
|
151 |
-
|
152 |
-
|
153 |
-
|
154 |
-
|
155 |
-
|
156 |
-
|
157 |
-
|
158 |
-
|
159 |
-
|
160 |
-
|
161 |
-
|
162 |
-
|
163 |
-
|
164 |
-
|
165 |
-
|
166 |
-
|
167 |
-
|
168 |
-
|
169 |
-
|
170 |
-
|
171 |
-
|
172 |
-
|
173 |
-
|
174 |
-
|
175 |
-
|
176 |
-
|
177 |
-
|
178 |
-
|
179 |
-
|
180 |
-
|
181 |
-
|
182 |
-
|
183 |
-
|
184 |
-
|
185 |
-
|
186 |
-
|
187 |
-
|
188 |
-
|
189 |
-
|
190 |
-
|
191 |
-
|
192 |
-
|
193 |
-
|
194 |
-
|
195 |
-
|
196 |
-
|
197 |
-
|
198 |
-
|
199 |
-
|
200 |
-
|
201 |
-
|
202 |
-
|
203 |
-
|
204 |
-
|
205 |
-
|
206 |
-
|
207 |
-
|
208 |
-
|
209 |
-
|
210 |
-
|
211 |
-
|
212 |
-
|
213 |
-
|
214 |
-
|
215 |
-
|
216 |
-
|
217 |
-
|
218 |
-
|
219 |
-
|
220 |
-
|
221 |
-
|
222 |
-
|
223 |
-
|
224 |
-
|
225 |
-
|
226 |
-
|
227 |
-
|
228 |
-
|
229 |
-
chunk
|
230 |
-
|
231 |
-
|
232 |
-
|
233 |
-
|
234 |
-
logging.debug(
|
235 |
-
|
236 |
-
|
237 |
-
|
238 |
-
|
239 |
-
|
240 |
-
|
241 |
-
|
242 |
-
|
243 |
-
|
244 |
-
|
245 |
-
|
246 |
-
|
247 |
-
|
248 |
-
|
249 |
-
|
250 |
-
|
251 |
-
|
252 |
-
|
253 |
-
|
254 |
-
logging.info("speech-to-text: Saving
|
255 |
-
|
256 |
-
|
257 |
-
|
258 |
-
|
259 |
-
|
260 |
-
|
261 |
-
|
262 |
-
|
263 |
-
|
264 |
-
|
265 |
-
|
266 |
-
|
267 |
-
|
268 |
-
|
269 |
-
|
270 |
-
|
271 |
-
|
272 |
-
|
273 |
-
|
274 |
-
|
275 |
-
|
276 |
-
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
277 |
#######################################################################################################################
|
|
|
1 |
+
# Audio_Transcription_Lib.py
|
2 |
+
#########################################
|
3 |
+
# Transcription Library
|
4 |
+
# This library is used to perform transcription of audio files.
|
5 |
+
# Currently, uses faster_whisper for transcription.
|
6 |
+
#
|
7 |
+
####################
|
8 |
+
# Function List
|
9 |
+
#
|
10 |
+
# 1. convert_to_wav(video_file_path, offset=0, overwrite=False)
|
11 |
+
# 2. speech_to_text(audio_file_path, selected_source_lang='en', whisper_model='small.en', vad_filter=False)
|
12 |
+
#
|
13 |
+
####################
|
14 |
+
#
|
15 |
+
# Import necessary libraries to run solo for testing
|
16 |
+
import gc
|
17 |
+
import json
|
18 |
+
import logging
|
19 |
+
import multiprocessing
|
20 |
+
import os
|
21 |
+
import queue
|
22 |
+
import sys
|
23 |
+
import subprocess
|
24 |
+
import tempfile
|
25 |
+
import threading
|
26 |
+
import time
|
27 |
+
# DEBUG Imports
|
28 |
+
#from memory_profiler import profile
|
29 |
+
import pyaudio
|
30 |
+
from faster_whisper import WhisperModel as OriginalWhisperModel
|
31 |
+
from typing import Optional, Union, List, Dict, Any
|
32 |
+
#
|
33 |
+
# Import Local
|
34 |
+
from App_Function_Libraries.Utils.Utils import load_comprehensive_config
|
35 |
+
from App_Function_Libraries.Metrics.metrics_logger import log_counter, log_histogram
|
36 |
+
#
|
37 |
+
#######################################################################################################################
|
38 |
+
# Function Definitions
|
39 |
+
#
|
40 |
+
|
41 |
+
# Convert video .m4a into .wav using ffmpeg
|
42 |
+
# ffmpeg -i "example.mp4" -ar 16000 -ac 1 -c:a pcm_s16le "output.wav"
|
43 |
+
# https://www.gyan.dev/ffmpeg/builds/
|
44 |
+
#
|
45 |
+
|
46 |
+
|
47 |
+
whisper_model_instance = None
|
48 |
+
config = load_comprehensive_config()
|
49 |
+
processing_choice = config.get('Processing', 'processing_choice', fallback='cpu')
|
50 |
+
total_thread_count = multiprocessing.cpu_count()
|
51 |
+
|
52 |
+
|
53 |
+
class WhisperModel(OriginalWhisperModel):
|
54 |
+
tldw_dir = os.path.dirname(os.path.dirname(__file__))
|
55 |
+
default_download_root = os.path.join(tldw_dir, 'models', 'Whisper')
|
56 |
+
|
57 |
+
valid_model_sizes = [
|
58 |
+
"tiny.en", "tiny", "base.en", "base", "small.en", "small", "medium.en", "medium",
|
59 |
+
"large-v1", "large-v2", "large-v3", "large", "distil-large-v2", "distil-medium.en",
|
60 |
+
"distil-small.en", "distil-large-v3",
|
61 |
+
]
|
62 |
+
|
63 |
+
def __init__(
|
64 |
+
self,
|
65 |
+
model_size_or_path: str,
|
66 |
+
device: str = processing_choice,
|
67 |
+
device_index: Union[int, List[int]] = 0,
|
68 |
+
compute_type: str = "default",
|
69 |
+
cpu_threads: int = 0,#total_thread_count, FIXME - I think this should be 0
|
70 |
+
num_workers: int = 1,
|
71 |
+
download_root: Optional[str] = None,
|
72 |
+
local_files_only: bool = False,
|
73 |
+
files: Optional[Dict[str, Any]] = None,
|
74 |
+
**model_kwargs: Any
|
75 |
+
):
|
76 |
+
if download_root is None:
|
77 |
+
download_root = self.default_download_root
|
78 |
+
|
79 |
+
os.makedirs(download_root, exist_ok=True)
|
80 |
+
|
81 |
+
# FIXME - validate....
|
82 |
+
# Also write an integration test...
|
83 |
+
# Check if model_size_or_path is a valid model size
|
84 |
+
if model_size_or_path in self.valid_model_sizes:
|
85 |
+
# It's a model size, so we'll use the download_root
|
86 |
+
model_path = os.path.join(download_root, model_size_or_path)
|
87 |
+
if not os.path.isdir(model_path):
|
88 |
+
# If it doesn't exist, we'll let the parent class download it
|
89 |
+
model_size_or_path = model_size_or_path # Keep the original model size
|
90 |
+
else:
|
91 |
+
# If it exists, use the full path
|
92 |
+
model_size_or_path = model_path
|
93 |
+
else:
|
94 |
+
# It's not a valid model size, so assume it's a path
|
95 |
+
model_size_or_path = os.path.abspath(model_size_or_path)
|
96 |
+
|
97 |
+
super().__init__(
|
98 |
+
model_size_or_path,
|
99 |
+
device=device,
|
100 |
+
device_index=device_index,
|
101 |
+
compute_type=compute_type,
|
102 |
+
cpu_threads=cpu_threads,
|
103 |
+
num_workers=num_workers,
|
104 |
+
download_root=download_root,
|
105 |
+
local_files_only=local_files_only,
|
106 |
+
# Maybe? idk, FIXME
|
107 |
+
# files=files,
|
108 |
+
# **model_kwargs
|
109 |
+
)
|
110 |
+
|
111 |
+
def get_whisper_model(model_name, device):
|
112 |
+
global whisper_model_instance
|
113 |
+
if whisper_model_instance is None:
|
114 |
+
logging.info(f"Initializing new WhisperModel with size {model_name} on device {device}")
|
115 |
+
whisper_model_instance = WhisperModel(model_name, device=device)
|
116 |
+
return whisper_model_instance
|
117 |
+
|
118 |
+
# os.system(r'.\Bin\ffmpeg.exe -ss 00:00:00 -i "{video_file_path}" -ar 16000 -ac 1 -c:a pcm_s16le "{out_path}"')
|
119 |
+
#DEBUG
|
120 |
+
#@profile
|
121 |
+
def convert_to_wav(video_file_path, offset=0, overwrite=False):
|
122 |
+
log_counter("convert_to_wav_attempt", labels={"file_path": video_file_path})
|
123 |
+
start_time = time.time()
|
124 |
+
|
125 |
+
out_path = os.path.splitext(video_file_path)[0] + ".wav"
|
126 |
+
|
127 |
+
if os.path.exists(out_path) and not overwrite:
|
128 |
+
print(f"File '{out_path}' already exists. Skipping conversion.")
|
129 |
+
logging.info(f"Skipping conversion as file already exists: {out_path}")
|
130 |
+
log_counter("convert_to_wav_skipped", labels={"file_path": video_file_path})
|
131 |
+
return out_path
|
132 |
+
|
133 |
+
print("Starting conversion process of .m4a to .WAV")
|
134 |
+
out_path = os.path.splitext(video_file_path)[0] + ".wav"
|
135 |
+
|
136 |
+
try:
|
137 |
+
if os.name == "nt":
|
138 |
+
logging.debug("ffmpeg being ran on windows")
|
139 |
+
|
140 |
+
if sys.platform.startswith('win'):
|
141 |
+
ffmpeg_cmd = ".\\Bin\\ffmpeg.exe"
|
142 |
+
logging.debug(f"ffmpeg_cmd: {ffmpeg_cmd}")
|
143 |
+
else:
|
144 |
+
ffmpeg_cmd = 'ffmpeg' # Assume 'ffmpeg' is in PATH for non-Windows systems
|
145 |
+
|
146 |
+
command = [
|
147 |
+
ffmpeg_cmd, # Assuming the working directory is correctly set where .\Bin exists
|
148 |
+
"-ss", "00:00:00", # Start at the beginning of the video
|
149 |
+
"-i", video_file_path,
|
150 |
+
"-ar", "16000", # Audio sample rate
|
151 |
+
"-ac", "1", # Number of audio channels
|
152 |
+
"-c:a", "pcm_s16le", # Audio codec
|
153 |
+
out_path
|
154 |
+
]
|
155 |
+
try:
|
156 |
+
# Redirect stdin from null device to prevent ffmpeg from waiting for input
|
157 |
+
with open(os.devnull, 'rb') as null_file:
|
158 |
+
result = subprocess.run(command, stdin=null_file, text=True, capture_output=True)
|
159 |
+
if result.returncode == 0:
|
160 |
+
logging.info("FFmpeg executed successfully")
|
161 |
+
logging.debug("FFmpeg output: %s", result.stdout)
|
162 |
+
else:
|
163 |
+
logging.error("Error in running FFmpeg")
|
164 |
+
logging.error("FFmpeg stderr: %s", result.stderr)
|
165 |
+
raise RuntimeError(f"FFmpeg error: {result.stderr}")
|
166 |
+
except Exception as e:
|
167 |
+
logging.error("Error occurred - ffmpeg doesn't like windows")
|
168 |
+
raise RuntimeError("ffmpeg failed")
|
169 |
+
elif os.name == "posix":
|
170 |
+
os.system(f'ffmpeg -ss 00:00:00 -i "{video_file_path}" -ar 16000 -ac 1 -c:a pcm_s16le "{out_path}"')
|
171 |
+
else:
|
172 |
+
raise RuntimeError("Unsupported operating system")
|
173 |
+
logging.info("Conversion to WAV completed: %s", out_path)
|
174 |
+
log_counter("convert_to_wav_success", labels={"file_path": video_file_path})
|
175 |
+
except Exception as e:
|
176 |
+
logging.error("speech-to-text: Error transcribing audio: %s", str(e))
|
177 |
+
log_counter("convert_to_wav_error", labels={"file_path": video_file_path, "error": str(e)})
|
178 |
+
return {"error": str(e)}
|
179 |
+
|
180 |
+
conversion_time = time.time() - start_time
|
181 |
+
log_histogram("convert_to_wav_duration", conversion_time, labels={"file_path": video_file_path})
|
182 |
+
|
183 |
+
gc.collect()
|
184 |
+
return out_path
|
185 |
+
|
186 |
+
|
187 |
+
# Transcribe .wav into .segments.json
|
188 |
+
#DEBUG
|
189 |
+
#@profile
|
190 |
+
# FIXME - I feel like the `vad_filter` shoudl be enabled by default....
|
191 |
+
def speech_to_text(audio_file_path, selected_source_lang='en', whisper_model='medium.en', vad_filter=False, diarize=False):
|
192 |
+
log_counter("speech_to_text_attempt", labels={"file_path": audio_file_path, "model": whisper_model})
|
193 |
+
time_start = time.time()
|
194 |
+
|
195 |
+
if audio_file_path is None:
|
196 |
+
log_counter("speech_to_text_error", labels={"error": "No audio file provided"})
|
197 |
+
raise ValueError("speech-to-text: No audio file provided")
|
198 |
+
logging.info("speech-to-text: Audio file path: %s", audio_file_path)
|
199 |
+
|
200 |
+
try:
|
201 |
+
_, file_ending = os.path.splitext(audio_file_path)
|
202 |
+
out_file = audio_file_path.replace(file_ending, "-whisper_model-"+whisper_model+".segments.json")
|
203 |
+
prettified_out_file = audio_file_path.replace(file_ending, "-whisper_model-"+whisper_model+".segments_pretty.json")
|
204 |
+
if os.path.exists(out_file):
|
205 |
+
logging.info("speech-to-text: Segments file already exists: %s", out_file)
|
206 |
+
with open(out_file) as f:
|
207 |
+
global segments
|
208 |
+
segments = json.load(f)
|
209 |
+
return segments
|
210 |
+
|
211 |
+
logging.info('speech-to-text: Starting transcription...')
|
212 |
+
# FIXME - revisit this
|
213 |
+
options = dict(language=selected_source_lang, beam_size=10, best_of=10, vad_filter=vad_filter)
|
214 |
+
transcribe_options = dict(task="transcribe", **options)
|
215 |
+
# use function and config at top of file
|
216 |
+
logging.debug("speech-to-text: Using whisper model: %s", whisper_model)
|
217 |
+
whisper_model_instance = get_whisper_model(whisper_model, processing_choice)
|
218 |
+
# faster_whisper transcription right here - FIXME -test batching - ha
|
219 |
+
segments_raw, info = whisper_model_instance.transcribe(audio_file_path, **transcribe_options)
|
220 |
+
|
221 |
+
segments = []
|
222 |
+
for segment_chunk in segments_raw:
|
223 |
+
chunk = {
|
224 |
+
"Time_Start": segment_chunk.start,
|
225 |
+
"Time_End": segment_chunk.end,
|
226 |
+
"Text": segment_chunk.text
|
227 |
+
}
|
228 |
+
logging.debug("Segment: %s", chunk)
|
229 |
+
segments.append(chunk)
|
230 |
+
# Print to verify its working
|
231 |
+
logging.info(f"{segment_chunk.start:.2f}s - {segment_chunk.end:.2f}s | {segment_chunk.text}")
|
232 |
+
|
233 |
+
# Log it as well.
|
234 |
+
logging.debug(
|
235 |
+
f"Transcribed Segment: {segment_chunk.start:.2f}s - {segment_chunk.end:.2f}s | {segment_chunk.text}")
|
236 |
+
|
237 |
+
if segments:
|
238 |
+
segments[0]["Text"] = f"This text was transcribed using whisper model: {whisper_model}\n\n" + segments[0]["Text"]
|
239 |
+
|
240 |
+
if not segments:
|
241 |
+
log_counter("speech_to_text_error", labels={"error": "No transcription produced"})
|
242 |
+
raise RuntimeError("No transcription produced. The audio file may be invalid or empty.")
|
243 |
+
|
244 |
+
transcription_time = time.time() - time_start
|
245 |
+
logging.info("speech-to-text: Transcription completed in %.2f seconds", transcription_time)
|
246 |
+
log_histogram("speech_to_text_duration", transcription_time, labels={"file_path": audio_file_path, "model": whisper_model})
|
247 |
+
log_counter("speech_to_text_success", labels={"file_path": audio_file_path, "model": whisper_model})
|
248 |
+
# Save the segments to a JSON file - prettified and non-prettified
|
249 |
+
# FIXME refactor so this is an optional flag to save either the prettified json file or the normal one
|
250 |
+
save_json = True
|
251 |
+
if save_json:
|
252 |
+
logging.info("speech-to-text: Saving segments to JSON file")
|
253 |
+
output_data = {'segments': segments}
|
254 |
+
logging.info("speech-to-text: Saving prettified JSON to %s", prettified_out_file)
|
255 |
+
with open(prettified_out_file, 'w') as f:
|
256 |
+
json.dump(output_data, f, indent=2)
|
257 |
+
|
258 |
+
logging.info("speech-to-text: Saving JSON to %s", out_file)
|
259 |
+
with open(out_file, 'w') as f:
|
260 |
+
json.dump(output_data, f)
|
261 |
+
|
262 |
+
logging.debug(f"speech-to-text: returning {segments[:500]}")
|
263 |
+
gc.collect()
|
264 |
+
return segments
|
265 |
+
|
266 |
+
except Exception as e:
|
267 |
+
logging.error("speech-to-text: Error transcribing audio: %s", str(e))
|
268 |
+
log_counter("speech_to_text_error", labels={"file_path": audio_file_path, "model": whisper_model, "error": str(e)})
|
269 |
+
raise RuntimeError("speech-to-text: Error transcribing audio")
|
270 |
+
|
271 |
+
|
272 |
+
def record_audio(duration, sample_rate=16000, chunk_size=1024):
|
273 |
+
log_counter("record_audio_attempt", labels={"duration": duration})
|
274 |
+
p = pyaudio.PyAudio()
|
275 |
+
stream = p.open(format=pyaudio.paInt16,
|
276 |
+
channels=1,
|
277 |
+
rate=sample_rate,
|
278 |
+
input=True,
|
279 |
+
frames_per_buffer=chunk_size)
|
280 |
+
|
281 |
+
print("Recording...")
|
282 |
+
frames = []
|
283 |
+
stop_recording = threading.Event()
|
284 |
+
audio_queue = queue.Queue()
|
285 |
+
|
286 |
+
def audio_callback():
|
287 |
+
for _ in range(0, int(sample_rate / chunk_size * duration)):
|
288 |
+
if stop_recording.is_set():
|
289 |
+
break
|
290 |
+
data = stream.read(chunk_size)
|
291 |
+
audio_queue.put(data)
|
292 |
+
|
293 |
+
audio_thread = threading.Thread(target=audio_callback)
|
294 |
+
audio_thread.start()
|
295 |
+
|
296 |
+
return p, stream, audio_queue, stop_recording, audio_thread
|
297 |
+
|
298 |
+
|
299 |
+
def stop_recording(p, stream, audio_queue, stop_recording_event, audio_thread):
|
300 |
+
log_counter("stop_recording_attempt")
|
301 |
+
start_time = time.time()
|
302 |
+
stop_recording_event.set()
|
303 |
+
audio_thread.join()
|
304 |
+
|
305 |
+
frames = []
|
306 |
+
while not audio_queue.empty():
|
307 |
+
frames.append(audio_queue.get())
|
308 |
+
|
309 |
+
print("Recording finished.")
|
310 |
+
|
311 |
+
stream.stop_stream()
|
312 |
+
stream.close()
|
313 |
+
p.terminate()
|
314 |
+
|
315 |
+
stop_time = time.time() - start_time
|
316 |
+
log_histogram("stop_recording_duration", stop_time)
|
317 |
+
log_counter("stop_recording_success")
|
318 |
+
return b''.join(frames)
|
319 |
+
|
320 |
+
def save_audio_temp(audio_data, sample_rate=16000):
|
321 |
+
log_counter("save_audio_temp_attempt")
|
322 |
+
with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as temp_file:
|
323 |
+
import wave
|
324 |
+
wf = wave.open(temp_file.name, 'wb')
|
325 |
+
wf.setnchannels(1)
|
326 |
+
wf.setsampwidth(2)
|
327 |
+
wf.setframerate(sample_rate)
|
328 |
+
wf.writeframes(audio_data)
|
329 |
+
wf.close()
|
330 |
+
log_counter("save_audio_temp_success")
|
331 |
+
return temp_file.name
|
332 |
+
|
333 |
+
#
|
334 |
+
#
|
335 |
#######################################################################################################################
|