feat: add hallucination detection and watchdog for stuck transcriptions
This commit is contained in:
@@ -34,6 +34,10 @@ Run the script using `python3 transcribe.py` with optional flags.
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```bash
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```bash
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python3 transcribe.py -l
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python3 transcribe.py -l
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```
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```
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- **Caption system audio (speakers) using a loopback device:**
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```bash
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python3 transcribe.py --loopback -es
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```
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- **Transcribe and translate to Spanish (screen only):**
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- **Transcribe and translate to Spanish (screen only):**
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```bash
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```bash
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python3 transcribe.py -es
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python3 transcribe.py -es
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@@ -55,6 +59,7 @@ Run the script using `python3 transcribe.py` with optional flags.
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- `-i`, `--ingest`: Enable data transmission to the remote server.
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- `-i`, `--ingest`: Enable data transmission to the remote server.
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- `-l`, `--list-devices`: Show available audio devices and exit.
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- `-l`, `--list-devices`: Show available audio devices and exit.
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- `-d`, `--device [ID]`: Input device index (bypasses selection prompt).
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- `-d`, `--device [ID]`: Input device index (bypasses selection prompt).
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- `--loopback`: Automatically select a loopback device (e.g., BlackHole, Stereo Mix) to caption system audio.
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- `-q`, `--quantize`: Use 4-bit quantized Whisper model for faster transcription (Strategy 3).
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- `-q`, `--quantize`: Use 4-bit quantized Whisper model for faster transcription (Strategy 3).
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- `-s`, `--stream`: Enable real-time streaming transcription/draft mode (Strategy 1).
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- `-s`, `--stream`: Enable real-time streaming transcription/draft mode (Strategy 1).
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- `-c`, `--context`: Enable prompt caching/rolling context to help the model maintain sentence continuity across chunks.
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- `-c`, `--context`: Enable prompt caching/rolling context to help the model maintain sentence continuity across chunks.
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+106
-3
@@ -1,10 +1,12 @@
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import sys
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import sys
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import time
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import time
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import re
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import requests
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import requests
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import threading
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import threading
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import json
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import json
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import argparse
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import argparse
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from unittest.mock import MagicMock
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from unittest.mock import MagicMock
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from collections import Counter
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# Comprehensive workaround for missing _lzma in some Python builds
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# Comprehensive workaround for missing _lzma in some Python builds
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try:
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try:
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@@ -45,11 +47,38 @@ CHANNELS = 1
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SAMPLERATE = 16000
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SAMPLERATE = 16000
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BLOCK_SIZE = 512
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BLOCK_SIZE = 512
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VAD_THRESHOLD = 0.5
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VAD_THRESHOLD = 0.5
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BUFFER_LIMIT = SAMPLERATE * 30
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audio_queue = queue.Queue()
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audio_queue = queue.Queue()
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ingest_queue = queue.Queue()
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ingest_queue = queue.Queue()
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def is_hallucination(text):
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"""Detect common Whisper hallucinations or high repetition."""
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if not text: return False
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# Common hallucinations
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hallucinations = [
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r"thanks? for watching",
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r"please subscribe",
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r"youtube",
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r"click the link",
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r"like and subscribe",
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r"tuned in",
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r"next time",
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]
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for pattern in hallucinations:
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if re.search(pattern, text, re.IGNORECASE):
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return True
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# Check for excessive word repetition (e.g. "Hallelujah" repeated 10 times)
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words = text.lower().split()
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if len(words) >= 8:
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counts = Counter(words)
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most_common_word, count = counts.most_common(1)[0]
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if count / len(words) > 0.75:
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return True
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return False
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def callback(indata, frames, time, status):
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def callback(indata, frames, time, status):
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if status:
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if status:
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print(status, file=sys.stderr)
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print(status, file=sys.stderr)
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@@ -91,11 +120,13 @@ def main():
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parser.add_argument("-i", "--ingest", action="store_true", help="Enable data transmission to server")
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parser.add_argument("-i", "--ingest", action="store_true", help="Enable data transmission to server")
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parser.add_argument("-l", "--list-devices", action="store_true", help="Show available audio devices and exit")
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parser.add_argument("-l", "--list-devices", action="store_true", help="Show available audio devices and exit")
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parser.add_argument("-d", "--device", type=int, help="Input device index")
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parser.add_argument("-d", "--device", type=int, help="Input device index")
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parser.add_argument("--loopback", action="store_true", help="Automatically select a loopback device (e.g., BlackHole, Stereo Mix)")
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parser.add_argument("-q", "--quantize", action="store_true", help="Use 4-bit quantized Whisper model for speed")
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parser.add_argument("-q", "--quantize", action="store_true", help="Use 4-bit quantized Whisper model for speed")
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parser.add_argument("-s", "--stream", action="store_true", help="Enable real-time streaming transcription (Draft mode)")
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parser.add_argument("-s", "--stream", action="store_true", help="Enable real-time streaming transcription (Draft mode)")
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parser.add_argument("-c", "--context", action="store_true", help="Enable prompt caching/rolling context for better continuity")
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parser.add_argument("-c", "--context", action="store_true", help="Enable prompt caching/rolling context for better continuity")
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parser.add_argument("--lang", type=str, help="Hardcode source language (e.g. 'en', 'es') to bypass detection")
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parser.add_argument("--lang", type=str, help="Hardcode source language (e.g. 'en', 'es') to bypass detection")
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parser.add_argument("--silence", type=int, default=1000, help="Minimum silence duration in ms to end a chunk (default: 1000)")
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parser.add_argument("--silence", type=int, default=1000, help="Minimum silence duration in ms to end a chunk (default: 1000)")
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parser.add_argument("--max-buffer", type=int, default=20, help="Maximum buffer duration in seconds before forcing a flush (default: 20)")
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args = parser.parse_args()
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args = parser.parse_args()
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@@ -107,6 +138,7 @@ def main():
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print(sd.query_devices())
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print(sd.query_devices())
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return
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return
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buffer_limit = SAMPLERATE * args.max_buffer
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device = "mps" if torch.backends.mps.is_available() else "cpu"
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device = "mps" if torch.backends.mps.is_available() else "cpu"
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print(f"Using device: {device}")
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print(f"Using device: {device}")
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@@ -132,9 +164,28 @@ def main():
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print("Models loaded.")
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print("Models loaded.")
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# 2. Select Audio Device
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# 2. Select Audio Device
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device_index = None
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if args.device is not None:
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if args.device is not None:
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device_index = args.device
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device_index = args.device
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else:
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elif args.loopback:
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print("Searching for loopback device...")
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devices = sd.query_devices()
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for i, dev in enumerate(devices):
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name = dev['name'].lower()
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if any(keyword in name for keyword in ["blackhole", "loopback", "soundflower", "stereo mix"]):
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if dev['max_input_channels'] > 0:
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device_index = i
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print(f"Using loopback device: {dev['name']} (Index {i})")
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print("\n[TIP] For macOS with BlackHole:")
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print("1. Open 'Audio MIDI Setup' and create a 'Multi-Output Device'.")
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print("2. Select your speakers AND 'BlackHole 2ch'.")
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print("3. Set your system output to this 'Multi-Output Device'.")
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print("This way you can hear the audio while it is being captioned.\n")
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break
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if device_index is None:
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print("No loopback device found. Falling back to default.")
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if device_index is None and not args.loopback:
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print("\nAvailable Audio Devices:")
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print("\nAvailable Audio Devices:")
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print(sd.query_devices())
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print(sd.query_devices())
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try:
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try:
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@@ -144,12 +195,19 @@ def main():
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print("Invalid input, using default device.")
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print("Invalid input, using default device.")
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device_index = None
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device_index = None
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if device_index is not None:
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print(f"Using device index: {device_index}")
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else:
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print("Using system default input device.")
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print(f"\nStarting live transcription{' & server ingest' if args.ingest else ''}... (Press Ctrl+C to stop)")
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print(f"\nStarting live transcription{' & server ingest' if args.ingest else ''}... (Press Ctrl+C to stop)")
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audio_buffer = []
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audio_buffer = []
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speech_started = False
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speech_started = False
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last_stream_time = time.time()
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last_stream_time = time.time()
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last_change_time = time.time()
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rolling_context = ""
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rolling_context = ""
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last_draft_text = ""
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try:
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try:
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with sd.InputStream(samplerate=SAMPLERATE, channels=CHANNELS, callback=callback, blocksize=BLOCK_SIZE, device=device_index):
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with sd.InputStream(samplerate=SAMPLERATE, channels=CHANNELS, callback=callback, blocksize=BLOCK_SIZE, device=device_index):
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@@ -161,6 +219,7 @@ def main():
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if len(audio_buffer) > 0:
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if len(audio_buffer) > 0:
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current_audio = np.concatenate(audio_buffer)
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current_audio = np.concatenate(audio_buffer)
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audio_tensor = torch.from_numpy(current_audio)
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audio_tensor = torch.from_numpy(current_audio)
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buffer_duration = len(current_audio) / SAMPLERATE
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speech_timestamps = get_speech_timestamps(
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speech_timestamps = get_speech_timestamps(
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audio_tensor,
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audio_tensor,
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@@ -170,12 +229,28 @@ def main():
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min_silence_duration_ms=args.silence
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min_silence_duration_ms=args.silence
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)
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)
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# --- STUCK WATCHDOG ---
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# If buffer is getting long (>12s) and we haven't had a change in draft for 7s,
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# OR if buffer is extremely long (>25s) regardless of draft activity.
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time_since_last_change = time.time() - last_change_time
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if (buffer_duration > 12.0 and time_since_last_change > 7.0) or (buffer_duration > 25.0):
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print(f"\n[SYSTEM]: Transcription watchdog triggered (Buffer: {buffer_duration:.1f}s, No change: {time_since_last_change:.1f}s). Resetting...")
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audio_buffer = []
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speech_started = False
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rolling_context = ""
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last_draft_text = ""
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last_change_time = time.time()
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if args.stream:
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sys.stdout.write("\r\033[K")
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sys.stdout.flush()
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continue
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if len(speech_timestamps) > 0:
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if len(speech_timestamps) > 0:
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speech_started = True
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speech_started = True
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last_end = speech_timestamps[-1]['end']
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last_end = speech_timestamps[-1]['end']
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buffer_len_samples = len(current_audio)
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buffer_len_samples = len(current_audio)
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if (buffer_len_samples - last_end) > (SAMPLERATE * args.silence / 1000) or buffer_len_samples > BUFFER_LIMIT:
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if (buffer_len_samples - last_end) > (SAMPLERATE * args.silence / 1000) or buffer_len_samples > buffer_limit:
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# Clear draft line if it was used
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# Clear draft line if it was used
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if args.stream:
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if args.stream:
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@@ -194,11 +269,18 @@ def main():
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original_text = transcription_result['text'].strip()
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original_text = transcription_result['text'].strip()
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detected_lang = transcription_result.get('language', args.lang if args.lang else 'en')
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detected_lang = transcription_result.get('language', args.lang if args.lang else 'en')
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if is_hallucination(original_text):
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print(f"\n[SYSTEM]: Hallucination detected, ignoring and resetting context.")
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original_text = ""
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rolling_context = ""
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if original_text:
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if original_text:
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print(f"\n[{detected_lang.upper()}]: {original_text}")
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print(f"\n[{detected_lang.upper()}]: {original_text}")
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last_change_time = time.time() # Successfully transcribed full segment
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# Prepare payload
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# Prepare payload
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payload = {"original": original_text}
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payload = {"original": original_text}
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# ... (payload construction)
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# Include detected language if requested or if it's the bridge
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# Include detected language if requested or if it's the bridge
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if (detected_lang in active_target_langs) or (detected_lang == "en" and args.en):
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if (detected_lang in active_target_langs) or (detected_lang == "en" and args.en):
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@@ -251,6 +333,8 @@ def main():
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audio_buffer = []
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audio_buffer = []
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speech_started = False
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speech_started = False
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last_stream_time = time.time()
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last_stream_time = time.time()
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stuck_draft_count = 0
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last_draft_text = ""
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elif args.stream and (time.time() - last_stream_time) > 1.0:
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elif args.stream and (time.time() - last_stream_time) > 1.0:
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# Draft transcription
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# Draft transcription
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@@ -260,13 +344,32 @@ def main():
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draft_result = mlx_whisper.transcribe(current_audio, **draft_kwargs)
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draft_result = mlx_whisper.transcribe(current_audio, **draft_kwargs)
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draft_text = draft_result['text'].strip()
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draft_text = draft_result['text'].strip()
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if draft_text:
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if draft_text:
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# Update timestamp ONLY if the text actually changed
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if draft_text != last_draft_text:
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last_change_time = time.time()
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last_draft_text = draft_text
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sys.stdout.write(f"\r\033[K[DRAFT]: {draft_text}")
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sys.stdout.write(f"\r\033[K[DRAFT]: {draft_text}")
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sys.stdout.flush()
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sys.stdout.flush()
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# Send draft to ingest if enabled
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# Send draft to ingest if enabled
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if args.ingest:
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if args.ingest:
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ingest_queue.put({"draft": draft_text})
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ingest_queue.put({"draft": draft_text})
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else:
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# If Whisper returns empty, check if we've been silent for too long
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# even though VAD says there is speech.
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if buffer_duration > 10.0 and (time.time() - last_change_time) > 7.0:
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print(f"\n[SYSTEM]: Draft is empty while audio continues. Forcing reset...")
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audio_buffer = []
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speech_started = False
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rolling_context = ""
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last_change_time = time.time()
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last_draft_text = ""
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sys.stdout.write("\r\033[K")
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sys.stdout.flush()
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continue
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last_stream_time = time.time()
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last_stream_time = time.time()
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