Update App_Function_Libraries/Audio_Transcription_Lib.py

App_Function_Libraries/Audio_Transcription_Lib.py

@@ -1,247 +1,247 @@
-# Audio_Transcription_Lib.py
-#########################################
-# Transcription Library
-# This library is used to perform transcription of audio files.
-# Currently, uses faster_whisper for transcription.
-#
-####################
-# Function List
-#
-# 1. convert_to_wav(video_file_path, offset=0, overwrite=False)
-# 2. speech_to_text(audio_file_path, selected_source_lang='en', whisper_model='small.en', vad_filter=False)
-#
-####################
-#
-# Import necessary libraries to run solo for testing
-import gc
-import json
-import logging
-import os
-import queue
-import sys
-import subprocess
-import tempfile
-import threading
-import time
-import configparser
-# DEBUG Imports
-#from memory_profiler import profile
-import pyaudio
-# Import Local
-#
-#######################################################################################################################
-# Function Definitions
-#
-
-# Convert video .m4a into .wav using ffmpeg
-#   ffmpeg -i "example.mp4" -ar 16000 -ac 1 -c:a pcm_s16le "output.wav"
-#       https://www.gyan.dev/ffmpeg/builds/
-#
-
-
-whisper_model_instance = None
-# Retrieve processing choice from the configuration file
-config = configparser.ConfigParser()
-config.read('config.txt')
-processing_choice = config.get('Processing', 'processing_choice', fallback='cpu')
-
-
-# FIXME: This is a temporary solution.
-# This doesn't clear older models, which means potentially a lot of memory is being used...
-def get_whisper_model(model_name, device):
-    global whisper_model_instance
-    if whisper_model_instance is None:
-        from faster_whisper import WhisperModel
-        logging.info(f"Initializing new WhisperModel with size {model_name} on device {device}")
-        whisper_model_instance = WhisperModel(model_name, device=device)
-    return whisper_model_instance
-
-
-# os.system(r'.\Bin\ffmpeg.exe -ss 00:00:00 -i "{video_file_path}" -ar 16000 -ac 1 -c:a pcm_s16le "{out_path}"')
-#DEBUG
-#@profile
-def convert_to_wav(video_file_path, offset=0, overwrite=False):
-    out_path = os.path.splitext(video_file_path)[0] + ".wav"
-
-    if os.path.exists(out_path) and not overwrite:
-        print(f"File '{out_path}' already exists. Skipping conversion.")
-        logging.info(f"Skipping conversion as file already exists: {out_path}")
-        return out_path
-    print("Starting conversion process of .m4a to .WAV")
-    out_path = os.path.splitext(video_file_path)[0] + ".wav"
-
-    try:
-        if os.name == "nt":
-            logging.debug("ffmpeg being ran on windows")
-
-            if sys.platform.startswith('win'):
-                ffmpeg_cmd = ".\\Bin\\ffmpeg.exe"
-                logging.debug(f"ffmpeg_cmd: {ffmpeg_cmd}")
-            else:
-                ffmpeg_cmd = 'ffmpeg'  # Assume 'ffmpeg' is in PATH for non-Windows systems
-
-            command = [
-                ffmpeg_cmd,  # Assuming the working directory is correctly set where .\Bin exists
-                "-ss", "00:00:00",  # Start at the beginning of the video
-                "-i", video_file_path,
-                "-ar", "16000",  # Audio sample rate
-                "-ac", "1",  # Number of audio channels
-                "-c:a", "pcm_s16le",  # Audio codec
-                out_path
-            ]
-            try:
-                # Redirect stdin from null device to prevent ffmpeg from waiting for input
-                with open(os.devnull, 'rb') as null_file:
-                    result = subprocess.run(command, stdin=null_file, text=True, capture_output=True)
-                if result.returncode == 0:
-                    logging.info("FFmpeg executed successfully")
-                    logging.debug("FFmpeg output: %s", result.stdout)
-                else:
-                    logging.error("Error in running FFmpeg")
-                    logging.error("FFmpeg stderr: %s", result.stderr)
-                    raise RuntimeError(f"FFmpeg error: {result.stderr}")
-            except Exception as e:
-                logging.error("Error occurred - ffmpeg doesn't like windows")
-                raise RuntimeError("ffmpeg failed")
-        elif os.name == "posix":
-            os.system(f'ffmpeg -ss 00:00:00 -i "{video_file_path}" -ar 16000 -ac 1 -c:a pcm_s16le "{out_path}"')
-        else:
-            raise RuntimeError("Unsupported operating system")
-        logging.info("Conversion to WAV completed: %s", out_path)
-    except subprocess.CalledProcessError as e:
-        logging.error("Error executing FFmpeg command: %s", str(e))
-        raise RuntimeError("Error converting video file to WAV")
-    except Exception as e:
-        logging.error("speech-to-text: Error transcribing audio: %s", str(e))
-        return {"error": str(e)}
-    gc.collect()
-    return out_path
-
-
-# Transcribe .wav into .segments.json
-#DEBUG
-#@profile
-def speech_to_text(audio_file_path, selected_source_lang='en', whisper_model='medium.en', vad_filter=False, diarize=False):
-    global whisper_model_instance, processing_choice
-    logging.info('speech-to-text: Loading faster_whisper model: %s', whisper_model)
-
-    time_start = time.time()
-    if audio_file_path is None:
-        raise ValueError("speech-to-text: No audio file provided")
-    logging.info("speech-to-text: Audio file path: %s", audio_file_path)
-
-    try:
-        _, file_ending = os.path.splitext(audio_file_path)
-        out_file = audio_file_path.replace(file_ending, ".segments.json")
-        prettified_out_file = audio_file_path.replace(file_ending, ".segments_pretty.json")
-        if os.path.exists(out_file):
-            logging.info("speech-to-text: Segments file already exists: %s", out_file)
-            with open(out_file) as f:
-                global segments
-                segments = json.load(f)
-            return segments
-
-        logging.info('speech-to-text: Starting transcription...')
-        options = dict(language=selected_source_lang, beam_size=5, best_of=5, vad_filter=vad_filter)
-        transcribe_options = dict(task="transcribe", **options)
-        # use function and config at top of file
-        whisper_model_instance = get_whisper_model(whisper_model, processing_choice)
-        segments_raw, info = whisper_model_instance.transcribe(audio_file_path, **transcribe_options)
-
-        segments = []
-        for segment_chunk in segments_raw:
-            chunk = {
-                "Time_Start": segment_chunk.start,
-                "Time_End": segment_chunk.end,
-                "Text": segment_chunk.text
-            }
-            logging.debug("Segment: %s", chunk)
-            segments.append(chunk)
-
-        if segments:
-            segments[0]["Text"] = f"This text was transcribed using whisper model: {whisper_model}\n\n" + segments[0]["Text"]
-
-        if not segments:
-            raise RuntimeError("No transcription produced. The audio file may be invalid or empty.")
-        logging.info("speech-to-text: Transcription completed in %.2f seconds", time.time() - time_start)
-
-        # Save the segments to a JSON file - prettified and non-prettified
-        # FIXME so this is an optional flag to save either the prettified json file or the normal one
-        save_json = True
-        if save_json:
-            logging.info("speech-to-text: Saving segments to JSON file")
-            output_data = {'segments': segments}
-
-            logging.info("speech-to-text: Saving prettified JSON to %s", prettified_out_file)
-            with open(prettified_out_file, 'w') as f:
-                json.dump(output_data, f, indent=2)
-
-            logging.info("speech-to-text: Saving JSON to %s", out_file)
-            with open(out_file, 'w') as f:
-                json.dump(output_data, f)
-
-        logging.debug(f"speech-to-text: returning {segments[:500]}")
-        gc.collect()
-        return segments
-
-    except Exception as e:
-        logging.error("speech-to-text: Error transcribing audio: %s", str(e))
-        raise RuntimeError("speech-to-text: Error transcribing audio")
-
-
-def record_audio(duration, sample_rate=16000, chunk_size=1024):
-    p = pyaudio.PyAudio()
-    stream = p.open(format=pyaudio.paInt16,
-                    channels=1,
-                    rate=sample_rate,
-                    input=True,
-                    frames_per_buffer=chunk_size)
-
-    print("Recording...")
-    frames = []
-    stop_recording = threading.Event()
-    audio_queue = queue.Queue()
-
-    def audio_callback():
-        for _ in range(0, int(sample_rate / chunk_size * duration)):
-            if stop_recording.is_set():
-                break
-            data = stream.read(chunk_size)
-            audio_queue.put(data)
-
-    audio_thread = threading.Thread(target=audio_callback)
-    audio_thread.start()
-
-    return p, stream, audio_queue, stop_recording, audio_thread
-
-
-def stop_recording(p, stream, audio_queue, stop_recording_event, audio_thread):
-    stop_recording_event.set()
-    audio_thread.join()
-
-    frames = []
-    while not audio_queue.empty():
-        frames.append(audio_queue.get())
-
-    print("Recording finished.")
-
-    stream.stop_stream()
-    stream.close()
-    p.terminate()
-
-    return b''.join(frames)
-
-def save_audio_temp(audio_data, sample_rate=16000):
-    with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as temp_file:
-        import wave
-        wf = wave.open(temp_file.name, 'wb')
-        wf.setnchannels(1)
-        wf.setsampwidth(2)
-        wf.setframerate(sample_rate)
-        wf.writeframes(audio_data)
-        wf.close()
-        return temp_file.name
-
-#
-#
+# Audio_Transcription_Lib.py
+#########################################
+# Transcription Library
+# This library is used to perform transcription of audio files.
+# Currently, uses faster_whisper for transcription.
+#
+####################
+# Function List
+#
+# 1. convert_to_wav(video_file_path, offset=0, overwrite=False)
+# 2. speech_to_text(audio_file_path, selected_source_lang='en', whisper_model='small.en', vad_filter=False)
+#
+####################
+#
+# Import necessary libraries to run solo for testing
+import gc
+import json
+import logging
+import os
+import queue
+import sys
+import subprocess
+import tempfile
+import threading
+import time
+import configparser
+# DEBUG Imports
+#from memory_profiler import profile
+#import pyaudio
+# Import Local
+#
+#######################################################################################################################
+# Function Definitions
+#
+
+# Convert video .m4a into .wav using ffmpeg
+#   ffmpeg -i "example.mp4" -ar 16000 -ac 1 -c:a pcm_s16le "output.wav"
+#       https://www.gyan.dev/ffmpeg/builds/
+#
+
+
+whisper_model_instance = None
+# Retrieve processing choice from the configuration file
+config = configparser.ConfigParser()
+config.read('config.txt')
+processing_choice = config.get('Processing', 'processing_choice', fallback='cpu')
+
+
+# FIXME: This is a temporary solution.
+# This doesn't clear older models, which means potentially a lot of memory is being used...
+def get_whisper_model(model_name, device):
+    global whisper_model_instance
+    if whisper_model_instance is None:
+        from faster_whisper import WhisperModel
+        logging.info(f"Initializing new WhisperModel with size {model_name} on device {device}")
+        whisper_model_instance = WhisperModel(model_name, device=device)
+    return whisper_model_instance
+
+
+# os.system(r'.\Bin\ffmpeg.exe -ss 00:00:00 -i "{video_file_path}" -ar 16000 -ac 1 -c:a pcm_s16le "{out_path}"')
+#DEBUG
+#@profile
+def convert_to_wav(video_file_path, offset=0, overwrite=False):
+    out_path = os.path.splitext(video_file_path)[0] + ".wav"
+
+    if os.path.exists(out_path) and not overwrite:
+        print(f"File '{out_path}' already exists. Skipping conversion.")
+        logging.info(f"Skipping conversion as file already exists: {out_path}")
+        return out_path
+    print("Starting conversion process of .m4a to .WAV")
+    out_path = os.path.splitext(video_file_path)[0] + ".wav"
+
+    try:
+        if os.name == "nt":
+            logging.debug("ffmpeg being ran on windows")
+
+            if sys.platform.startswith('win'):
+                ffmpeg_cmd = ".\\Bin\\ffmpeg.exe"
+                logging.debug(f"ffmpeg_cmd: {ffmpeg_cmd}")
+            else:
+                ffmpeg_cmd = 'ffmpeg'  # Assume 'ffmpeg' is in PATH for non-Windows systems
+
+            command = [
+                ffmpeg_cmd,  # Assuming the working directory is correctly set where .\Bin exists
+                "-ss", "00:00:00",  # Start at the beginning of the video
+                "-i", video_file_path,
+                "-ar", "16000",  # Audio sample rate
+                "-ac", "1",  # Number of audio channels
+                "-c:a", "pcm_s16le",  # Audio codec
+                out_path
+            ]
+            try:
+                # Redirect stdin from null device to prevent ffmpeg from waiting for input
+                with open(os.devnull, 'rb') as null_file:
+                    result = subprocess.run(command, stdin=null_file, text=True, capture_output=True)
+                if result.returncode == 0:
+                    logging.info("FFmpeg executed successfully")
+                    logging.debug("FFmpeg output: %s", result.stdout)
+                else:
+                    logging.error("Error in running FFmpeg")
+                    logging.error("FFmpeg stderr: %s", result.stderr)
+                    raise RuntimeError(f"FFmpeg error: {result.stderr}")
+            except Exception as e:
+                logging.error("Error occurred - ffmpeg doesn't like windows")
+                raise RuntimeError("ffmpeg failed")
+        elif os.name == "posix":
+            os.system(f'ffmpeg -ss 00:00:00 -i "{video_file_path}" -ar 16000 -ac 1 -c:a pcm_s16le "{out_path}"')
+        else:
+            raise RuntimeError("Unsupported operating system")
+        logging.info("Conversion to WAV completed: %s", out_path)
+    except subprocess.CalledProcessError as e:
+        logging.error("Error executing FFmpeg command: %s", str(e))
+        raise RuntimeError("Error converting video file to WAV")
+    except Exception as e:
+        logging.error("speech-to-text: Error transcribing audio: %s", str(e))
+        return {"error": str(e)}
+    gc.collect()
+    return out_path
+
+
+# Transcribe .wav into .segments.json
+#DEBUG
+#@profile
+def speech_to_text(audio_file_path, selected_source_lang='en', whisper_model='medium.en', vad_filter=False, diarize=False):
+    global whisper_model_instance, processing_choice
+    logging.info('speech-to-text: Loading faster_whisper model: %s', whisper_model)
+
+    time_start = time.time()
+    if audio_file_path is None:
+        raise ValueError("speech-to-text: No audio file provided")
+    logging.info("speech-to-text: Audio file path: %s", audio_file_path)
+
+    try:
+        _, file_ending = os.path.splitext(audio_file_path)
+        out_file = audio_file_path.replace(file_ending, ".segments.json")
+        prettified_out_file = audio_file_path.replace(file_ending, ".segments_pretty.json")
+        if os.path.exists(out_file):
+            logging.info("speech-to-text: Segments file already exists: %s", out_file)
+            with open(out_file) as f:
+                global segments
+                segments = json.load(f)
+            return segments
+
+        logging.info('speech-to-text: Starting transcription...')
+        options = dict(language=selected_source_lang, beam_size=5, best_of=5, vad_filter=vad_filter)
+        transcribe_options = dict(task="transcribe", **options)
+        # use function and config at top of file
+        whisper_model_instance = get_whisper_model(whisper_model, processing_choice)
+        segments_raw, info = whisper_model_instance.transcribe(audio_file_path, **transcribe_options)
+
+        segments = []
+        for segment_chunk in segments_raw:
+            chunk = {
+                "Time_Start": segment_chunk.start,
+                "Time_End": segment_chunk.end,
+                "Text": segment_chunk.text
+            }
+            logging.debug("Segment: %s", chunk)
+            segments.append(chunk)
+
+        if segments:
+            segments[0]["Text"] = f"This text was transcribed using whisper model: {whisper_model}\n\n" + segments[0]["Text"]
+
+        if not segments:
+            raise RuntimeError("No transcription produced. The audio file may be invalid or empty.")
+        logging.info("speech-to-text: Transcription completed in %.2f seconds", time.time() - time_start)
+
+        # Save the segments to a JSON file - prettified and non-prettified
+        # FIXME so this is an optional flag to save either the prettified json file or the normal one
+        save_json = True
+        if save_json:
+            logging.info("speech-to-text: Saving segments to JSON file")
+            output_data = {'segments': segments}
+
+            logging.info("speech-to-text: Saving prettified JSON to %s", prettified_out_file)
+            with open(prettified_out_file, 'w') as f:
+                json.dump(output_data, f, indent=2)
+
+            logging.info("speech-to-text: Saving JSON to %s", out_file)
+            with open(out_file, 'w') as f:
+                json.dump(output_data, f)
+
+        logging.debug(f"speech-to-text: returning {segments[:500]}")
+        gc.collect()
+        return segments
+
+    except Exception as e:
+        logging.error("speech-to-text: Error transcribing audio: %s", str(e))
+        raise RuntimeError("speech-to-text: Error transcribing audio")
+
+
+#def record_audio(duration, sample_rate=16000, chunk_size=1024):
+#    p = pyaudio.PyAudio()
+#    stream = p.open(format=pyaudio.paInt16,
+#                    channels=1,
+#                    rate=sample_rate,
+#                    input=True,
+#                    frames_per_buffer=chunk_size)
+
+#    print("Recording...")
+#    frames = []
+#    stop_recording = threading.Event()
+#    audio_queue = queue.Queue()
+
+    def audio_callback():
+        for _ in range(0, int(sample_rate / chunk_size * duration)):
+            if stop_recording.is_set():
+                break
+            data = stream.read(chunk_size)
+            audio_queue.put(data)
+
+    audio_thread = threading.Thread(target=audio_callback)
+    audio_thread.start()
+
+    return p, stream, audio_queue, stop_recording, audio_thread
+
+
+def stop_recording(p, stream, audio_queue, stop_recording_event, audio_thread):
+    stop_recording_event.set()
+    audio_thread.join()
+
+    frames = []
+    while not audio_queue.empty():
+        frames.append(audio_queue.get())
+
+    print("Recording finished.")
+
+    stream.stop_stream()
+    stream.close()
+    p.terminate()
+
+    return b''.join(frames)
+
+def save_audio_temp(audio_data, sample_rate=16000):
+    with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as temp_file:
+        import wave
+        wf = wave.open(temp_file.name, 'wb')
+        wf.setnchannels(1)
+        wf.setsampwidth(2)
+        wf.setframerate(sample_rate)
+        wf.writeframes(audio_data)
+        wf.close()
+        return temp_file.name
+
+#
+#
 #######################################################################################################################