Demo Update 20

This commit is contained in:
2025-03-30 02:53:30 -04:00
parent 10902f1d71
commit 4fb2c9bc52
4 changed files with 196 additions and 55 deletions

View File

@@ -56,12 +56,8 @@ class AppModels:
generator = None
tokenizer = None
llm = None
whisperx_model = None
whisperx_align_model = None
whisperx_align_metadata = None
diarize_model = None
models = AppModels()
asr_model = None
asr_processor = None
def load_models():
"""Load all required models"""
@@ -76,16 +72,22 @@ def load_models():
logger.error(f"Error loading CSM 1B model: {str(e)}")
socketio.emit('model_status', {'model': 'csm', 'status': 'error', 'message': str(e)})
logger.info("Loading WhisperX model...")
logger.info("Loading Whisper ASR model...")
try:
# Use WhisperX instead of the regular Whisper
compute_type = "float16" if DEVICE == "cuda" else "float32"
models.whisperx_model = whisperx.load_model("large-v2", DEVICE, compute_type=compute_type)
logger.info("WhisperX model loaded successfully")
socketio.emit('model_status', {'model': 'whisperx', 'status': 'loaded'})
# Use regular Whisper instead of WhisperX to avoid compatibility issues
from transformers import WhisperProcessor, WhisperForConditionalGeneration
# Use a smaller model for faster processing
model_id = "openai/whisper-small"
models.asr_processor = WhisperProcessor.from_pretrained(model_id)
models.asr_model = WhisperForConditionalGeneration.from_pretrained(model_id).to(DEVICE)
logger.info("Whisper ASR model loaded successfully")
socketio.emit('model_status', {'model': 'asr', 'status': 'loaded'})
except Exception as e:
logger.error(f"Error loading WhisperX model: {str(e)}")
socketio.emit('model_status', {'model': 'whisperx', 'status': 'error', 'message': str(e)})
logger.error(f"Error loading ASR model: {str(e)}")
socketio.emit('model_status', {'model': 'asr', 'status': 'error', 'message': str(e)})
logger.info("Loading Llama 3.2 model...")
try:
@@ -141,7 +143,8 @@ def health_check():
"models_loaded": models.generator is not None and models.llm is not None
})
# Add a system status endpoint
# Fix the system_status function:
@app.route('/api/status')
def system_status():
return jsonify({
@@ -150,7 +153,7 @@ def system_status():
"device": DEVICE,
"models": {
"generator": models.generator is not None,
"whisperx": models.whisperx_model is not None,
"asr": models.asr_model is not None, # Use the correct model name
"llm": models.llm is not None
}
})
@@ -260,8 +263,8 @@ def process_audio_queue(session_id, q):
del user_queues[session_id]
def process_audio_and_respond(session_id, data):
"""Process audio data and generate a response using WhisperX"""
if models.generator is None or models.whisperx_model is None or models.llm is None:
"""Process audio data and generate a response using standard Whisper"""
if models.generator is None or models.asr_model is None or models.llm is None:
logger.warning("Models not yet loaded!")
with app.app_context():
socketio.emit('error', {'message': 'Models still loading, please wait'}, room=session_id)
@@ -293,47 +296,33 @@ def process_audio_and_respond(session_id, data):
temp_path = temp_file.name
try:
# Load audio using WhisperX
# Notify client that transcription is starting
with app.app_context():
socketio.emit('processing_status', {'status': 'transcribing'}, room=session_id)
# Load audio with WhisperX instead of torchaudio
audio = whisperx.load_audio(temp_path)
# Load audio for ASR processing
import librosa
speech_array, sampling_rate = librosa.load(temp_path, sr=16000)
# Transcribe using WhisperX
batch_size = 16 # Adjust based on available memory
result = models.whisperx_model.transcribe(audio, batch_size=batch_size)
# Convert to required format
input_features = models.asr_processor(
speech_array,
sampling_rate=sampling_rate,
return_tensors="pt"
).input_features.to(DEVICE)
# Get the detected language
language_code = result["language"]
logger.info(f"Detected language: {language_code}")
# Load alignment model if not already loaded
if models.whisperx_align_model is None or language_code != getattr(models, 'last_language', None):
# Clear previous models to save memory
if models.whisperx_align_model is not None:
del models.whisperx_align_model
del models.whisperx_align_metadata
gc.collect()
torch.cuda.empty_cache() if DEVICE == "cuda" else None
models.whisperx_align_model, models.whisperx_align_metadata = whisperx.load_align_model(
language_code=language_code, device=DEVICE
)
models.last_language = language_code
# Align the transcript
result = whisperx.align(
result["segments"],
models.whisperx_align_model,
models.whisperx_align_metadata,
audio,
DEVICE,
return_char_alignments=False
# Generate token ids
predicted_ids = models.asr_model.generate(
input_features,
language="en",
task="transcribe"
)
# Combine all segments into a single transcript
user_text = ' '.join([segment['text'] for segment in result['segments']])
# Decode the predicted ids to text
user_text = models.asr_processor.batch_decode(
predicted_ids,
skip_special_tokens=True
)[0]
# If no text was recognized, don't process further
if not user_text or len(user_text.strip()) == 0:
@@ -369,8 +358,7 @@ def process_audio_and_respond(session_id, data):
with app.app_context():
socketio.emit('transcription', {
'text': user_text,
'speaker': speaker_id,
'segments': result['segments'] # Send detailed segments info
'speaker': speaker_id
}, room=session_id)
# Generate AI response using Llama