Files
whisper-translation/transcribe.py
T

388 lines
18 KiB
Python

import sys
import time
import re
import requests
import threading
import json
import argparse
from unittest.mock import MagicMock
from collections import Counter
# Comprehensive workaround for missing _lzma in some Python builds
try:
import lzma
except ImportError:
mock_lzma = MagicMock()
mock_lzma.FORMAT_XZ = 1
mock_lzma.FORMAT_ALONE = 2
mock_lzma.FORMAT_RAW = 3
mock_lzma.CHECK_NONE = 0
mock_lzma.CHECK_CRC32 = 1
mock_lzma.CHECK_CRC64 = 4
mock_lzma.CHECK_SHA256 = 10
sys.modules["_lzma"] = MagicMock()
sys.modules["lzma"] = mock_lzma
import mlx_whisper
import numpy as np
import sounddevice as sd
import queue
import torch
from silero_vad import load_silero_vad, get_speech_timestamps
from transformers import MarianMTModel, MarianTokenizer
# Parameters
WHISPER_MODEL = "mlx-community/whisper-small-mlx"
INGEST_URL = "https://emiapi.reynafamily.com/live-captions/ingest"
# Translation models (English -> Target)
# Map to the specific keys requested by the backend
TARGET_LANGS = {
"es": "Helsinki-NLP/opus-mt-en-es",
"fr": "Helsinki-NLP/opus-mt-en-fr",
"ar": "Helsinki-NLP/opus-mt-en-ar" # Added Arabic as discussed before
}
CHANNELS = 1
SAMPLERATE = 16000
BLOCK_SIZE = 512
VAD_THRESHOLD = 0.5
audio_queue = queue.Queue()
ingest_queue = queue.Queue()
def is_hallucination(text):
"""Detect common Whisper hallucinations or high repetition."""
if not text: return False
# Common hallucinations
hallucinations = [
r"thanks? for watching",
r"please subscribe",
r"youtube",
r"click the link",
r"like and subscribe",
r"tuned in",
r"next time",
]
for pattern in hallucinations:
if re.search(pattern, text, re.IGNORECASE):
return True
# Check for excessive word repetition (e.g. "Hallelujah" repeated 10 times)
words = text.lower().split()
if len(words) >= 8:
counts = Counter(words)
most_common_word, count = counts.most_common(1)[0]
if count / len(words) > 0.75:
return True
return False
def callback(indata, frames, time, status):
if status:
print(status, file=sys.stderr)
audio_queue.put(indata.copy())
def ingest_worker():
"""Background thread to handle server ingestion with retries."""
while True:
payload = ingest_queue.get()
if payload is None: break
delay = 1
max_delay = 15
success = False
while not success:
try:
response = requests.post(INGEST_URL, json=payload, timeout=5)
if response.status_code == 200:
success = True
else:
print(f"\n[Ingest Error] Server returned {response.status_code}. Retrying in {delay}s...")
except Exception as e:
print(f"\n[Ingest Error] {e}. Retrying in {delay}s...")
if not success:
time.sleep(delay)
delay = min(delay * 2, max_delay)
ingest_queue.task_done()
def main():
global WHISPER_MODEL
parser = argparse.ArgumentParser(description="Live transcription and translation with Whisper.")
parser.add_argument("-es", action="store_true", help="Enable Spanish translation")
parser.add_argument("-en", action="store_true", help="Enable English (detected or bridged)")
parser.add_argument("-ar", action="store_true", help="Enable Arabic translation")
parser.add_argument("-fr", action="store_true", help="Enable French translation")
parser.add_argument("-i", "--ingest", action="store_true", help="Enable data transmission to server")
parser.add_argument("-l", "--list-devices", action="store_true", help="Show available audio devices and exit")
parser.add_argument("-d", "--device", type=int, help="Input device index")
parser.add_argument("--loopback", action="store_true", help="Automatically select a loopback device (e.g., BlackHole, Stereo Mix)")
parser.add_argument("-q", "--quantize", action="store_true", help="Use 4-bit quantized Whisper model for speed")
parser.add_argument("-s", "--stream", action="store_true", help="Enable real-time streaming transcription (Draft mode)")
parser.add_argument("-c", "--context", action="store_true", help="Enable prompt caching/rolling context for better continuity")
parser.add_argument("--lang", type=str, help="Hardcode source language (e.g. 'en', 'es') to bypass detection")
parser.add_argument("--silence", type=int, default=1000, help="Minimum silence duration in ms to end a chunk (default: 1000)")
parser.add_argument("--max-buffer", type=int, default=20, help="Maximum buffer duration in seconds before forcing a flush (default: 20)")
args = parser.parse_args()
if args.quantize:
WHISPER_MODEL = "mlx-community/whisper-small-mlx-4bit"
if args.list_devices:
print("\nAvailable Audio Devices:")
print(sd.query_devices())
return
buffer_limit = SAMPLERATE * args.max_buffer
device = "mps" if torch.backends.mps.is_available() else "cpu"
print(f"Using device: {device}")
# Only start ingest thread if enabled
if args.ingest:
threading.Thread(target=ingest_worker, daemon=True).start()
# 1. Load models
print(f"Loading Multilingual Whisper model '{WHISPER_MODEL}'...")
# Filter translation models to only those enabled by args
active_target_langs = {k: v for k, v in TARGET_LANGS.items() if getattr(args, k, False)}
translation_engines = {}
for lang_key, model_id in active_target_langs.items():
print(f"Loading {lang_key} translation model ({model_id})...")
tokenizer = MarianTokenizer.from_pretrained(model_id)
model = MarianMTModel.from_pretrained(model_id).to(device)
translation_engines[lang_key] = (model, tokenizer)
print("Loading Silero VAD model...")
vad_model = load_silero_vad()
print("Models loaded.")
# 2. Select Audio Device
device_index = None
if args.device is not None:
device_index = args.device
elif args.loopback:
print("Searching for loopback device...")
devices = sd.query_devices()
for i, dev in enumerate(devices):
name = dev['name'].lower()
if any(keyword in name for keyword in ["blackhole", "loopback", "soundflower", "stereo mix"]):
if dev['max_input_channels'] > 0:
device_index = i
print(f"Using loopback device: {dev['name']} (Index {i})")
print("\n[TIP] For macOS with BlackHole:")
print("1. Open 'Audio MIDI Setup' and create a 'Multi-Output Device'.")
print("2. Select your speakers AND 'BlackHole 2ch'.")
print("3. Set your system output to this 'Multi-Output Device'.")
print("This way you can hear the audio while it is being captioned.\n")
break
if device_index is None:
print("No loopback device found. Falling back to default.")
if device_index is None and not args.loopback:
print("\nAvailable Audio Devices:")
print(sd.query_devices())
try:
device_input = input("\nSelect input device index (or press Enter for default): ")
device_index = int(device_input) if device_input.strip() else None
except ValueError:
print("Invalid input, using default device.")
device_index = None
if device_index is not None:
print(f"Using device index: {device_index}")
else:
print("Using system default input device.")
print(f"\nStarting live transcription{' & server ingest' if args.ingest else ''}... (Press Ctrl+C to stop)")
audio_buffer = []
speech_started = False
last_stream_time = time.time()
last_change_time = time.time()
rolling_context = ""
last_draft_text = ""
try:
with sd.InputStream(samplerate=SAMPLERATE, channels=CHANNELS, callback=callback, blocksize=BLOCK_SIZE, device=device_index):
while True:
while not audio_queue.empty():
data = audio_queue.get()
audio_buffer.append(data.flatten())
if len(audio_buffer) > 0:
current_audio = np.concatenate(audio_buffer)
audio_tensor = torch.from_numpy(current_audio)
buffer_duration = len(current_audio) / SAMPLERATE
speech_timestamps = get_speech_timestamps(
audio_tensor,
vad_model,
sampling_rate=SAMPLERATE,
threshold=VAD_THRESHOLD,
min_silence_duration_ms=args.silence
)
# --- STUCK WATCHDOG ---
# If buffer is getting long (>12s) and we haven't had a change in draft for 7s,
# OR if buffer is extremely long (>25s) regardless of draft activity.
time_since_last_change = time.time() - last_change_time
if (buffer_duration > 12.0 and time_since_last_change > 7.0) or (buffer_duration > 25.0):
print(f"\n[SYSTEM]: Transcription watchdog triggered (Buffer: {buffer_duration:.1f}s, No change: {time_since_last_change:.1f}s). Resetting...")
audio_buffer = []
speech_started = False
rolling_context = ""
last_draft_text = ""
last_change_time = time.time()
if args.stream:
sys.stdout.write("\r\033[K")
sys.stdout.flush()
continue
if len(speech_timestamps) > 0:
speech_started = True
last_end = speech_timestamps[-1]['end']
buffer_len_samples = len(current_audio)
if (buffer_len_samples - last_end) > (SAMPLERATE * args.silence / 1000) or buffer_len_samples > buffer_limit:
# Clear draft line if it was used
if args.stream:
sys.stdout.write("\r\033[K")
sys.stdout.flush()
# Prepare transcription kwargs
transcribe_kwargs = {"path_or_hf_repo": WHISPER_MODEL}
if args.lang:
transcribe_kwargs["language"] = args.lang
if args.context and rolling_context:
transcribe_kwargs["initial_prompt"] = rolling_context
# 1. Transcribe & Detect Language
transcription_result = mlx_whisper.transcribe(current_audio, **transcribe_kwargs)
original_text = transcription_result['text'].strip()
detected_lang = transcription_result.get('language', args.lang if args.lang else 'en')
if is_hallucination(original_text):
print(f"\n[SYSTEM]: Hallucination detected, ignoring and resetting context.")
original_text = ""
rolling_context = ""
if original_text:
print(f"\n[{detected_lang.upper()}]: {original_text}")
last_change_time = time.time() # Successfully transcribed full segment
# Prepare payload
payload = {"original": original_text}
# ... (payload construction)
# Include detected language if requested or if it's the bridge
if (detected_lang in active_target_langs) or (detected_lang == "en" and args.en):
payload[detected_lang] = original_text
# 2. Bridge to English if not already English
if detected_lang != "en":
bridge_kwargs = {"path_or_hf_repo": WHISPER_MODEL, "task": "translate"}
if args.lang:
bridge_kwargs["language"] = args.lang
bridge_result = mlx_whisper.transcribe(current_audio, **bridge_kwargs)
english_text = bridge_result['text'].strip()
if args.en:
payload["en"] = english_text
print(f"[EN]: {english_text}")
else:
english_text = original_text
if args.en:
payload["en"] = english_text
# Update rolling context for next segment
if args.context:
# keep the last ~200 characters of the source language text
rolling_context = (rolling_context + " " + original_text)[-200:].strip()
# 3. Translate from English to other languages
if english_text and translation_engines:
# Limit input length
clean_en = english_text[:247] + "..." if len(english_text) > 250 else english_text
for lang_key, (model, tokenizer) in translation_engines.items():
# Skip if we already filled this (e.g. detected lang was 'es')
if lang_key in payload:
if lang_key != detected_lang: # Already printed original
print(f"[{lang_key.upper()}]: {payload[lang_key]}")
continue
inputs = tokenizer(clean_en, return_tensors="pt", padding=True).to(device)
with torch.no_grad():
translated_tokens = model.generate(**inputs, max_new_tokens=150)
translated_text = tokenizer.decode(translated_tokens[0], skip_special_tokens=True)
payload[lang_key] = translated_text
print(f"[{lang_key.upper()}]: {translated_text}")
# Queue for background ingestion if enabled
if args.ingest:
ingest_queue.put(payload)
# print(f"Sent to ingest: {list(payload.keys())}")
audio_buffer = []
speech_started = False
last_stream_time = time.time()
stuck_draft_count = 0
last_draft_text = ""
elif args.stream and (time.time() - last_stream_time) > 1.0:
# Draft transcription
draft_kwargs = {"path_or_hf_repo": WHISPER_MODEL}
if args.lang: draft_kwargs["language"] = args.lang
if args.context and rolling_context: draft_kwargs["initial_prompt"] = rolling_context
draft_result = mlx_whisper.transcribe(current_audio, **draft_kwargs)
draft_text = draft_result['text'].strip()
if draft_text:
# Update timestamp ONLY if the text actually changed
if draft_text != last_draft_text:
last_change_time = time.time()
last_draft_text = draft_text
sys.stdout.write(f"\r\033[K[DRAFT]: {draft_text}")
sys.stdout.flush()
# Send draft to ingest if enabled
if args.ingest:
ingest_queue.put({"draft": draft_text})
else:
# If Whisper returns empty, check if we've been silent for too long
# even though VAD says there is speech.
if buffer_duration > 10.0 and (time.time() - last_change_time) > 7.0:
print(f"\n[SYSTEM]: Draft is empty while audio continues. Forcing reset...")
audio_buffer = []
speech_started = False
rolling_context = ""
last_change_time = time.time()
last_draft_text = ""
sys.stdout.write("\r\033[K")
sys.stdout.flush()
continue
last_stream_time = time.time()
elif not speech_started and len(current_audio) > SAMPLERATE * 2:
audio_buffer = []
except KeyboardInterrupt:
print("\nStopped by user.")
except Exception as e:
print(f"\nError: {e}")
if __name__ == "__main__":
import multiprocessing
multiprocessing.freeze_support()
main()