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New: implemented many, many changes. 10% Phone-level detection: WORKING
278e294
"""
WebSocket streaming for real-time speech pathology diagnosis.
This module provides WebSocket endpoint for streaming audio analysis
with <50ms latency per frame requirement.
"""
import logging
import time
import uuid
import numpy as np
from typing import Optional, Dict
from collections import deque
from datetime import datetime
from fastapi import WebSocket, WebSocketDisconnect, HTTPException
from api.schemas import StreamingDiagnosisResponse, FluencyInfo, ArticulationInfo, ErrorDetailSchema
from models.phoneme_mapper import PhonemeMapper
from models.error_taxonomy import ErrorMapper, ErrorType
from inference.inference_pipeline import InferencePipeline
from config import AudioConfig, default_audio_config
logger = logging.getLogger(__name__)
class StreamingBuffer:
"""
Buffer for managing sliding window in streaming audio.
Maintains a buffer of audio samples and provides frames
for processing with overlap management.
"""
def __init__(self, window_size_samples: int, hop_size_samples: int):
"""
Initialize streaming buffer.
Args:
window_size_samples: Size of analysis window in samples
hop_size_samples: Hop size between windows in samples
"""
self.window_size_samples = window_size_samples
self.hop_size_samples = hop_size_samples
self.buffer = deque(maxlen=window_size_samples + hop_size_samples)
self.frame_index = 0
logger.debug(f"StreamingBuffer initialized: window={window_size_samples}, hop={hop_size_samples}")
def add_chunk(self, audio_chunk: np.ndarray) -> bool:
"""
Add audio chunk to buffer.
Args:
audio_chunk: Audio samples to add
Returns:
True if buffer has enough data for a frame, False otherwise
"""
self.buffer.extend(audio_chunk)
return len(self.buffer) >= self.window_size_samples
def get_frame(self) -> Optional[np.ndarray]:
"""
Get current frame from buffer.
Returns:
Audio frame array if ready, None otherwise
"""
if len(self.buffer) < self.window_size_samples:
return None
# Extract window (last window_size_samples)
frame = np.array(list(self.buffer)[-self.window_size_samples:])
return frame
def slide(self):
"""Advance buffer by hop size."""
# Remove oldest hop_size_samples
for _ in range(min(self.hop_size_samples, len(self.buffer))):
if self.buffer:
self.buffer.popleft()
self.frame_index += 1
# Global instances (will be injected)
inference_pipeline: Optional[InferencePipeline] = None
phoneme_mapper: Optional[PhonemeMapper] = None
error_mapper: Optional[ErrorMapper] = None
# Active streaming sessions
streaming_sessions: Dict[str, Dict] = {}
def initialize_streaming(
pipeline: InferencePipeline,
mapper: Optional[PhonemeMapper] = None,
error_mapper_instance: Optional[ErrorMapper] = None
):
"""
Initialize streaming with dependencies.
Args:
pipeline: InferencePipeline instance
mapper: Optional PhonemeMapper instance
error_mapper_instance: Optional ErrorMapper instance
"""
global inference_pipeline, phoneme_mapper, error_mapper
inference_pipeline = pipeline
if mapper is None:
try:
phoneme_mapper = PhonemeMapper(
frame_duration_ms=default_audio_config.chunk_duration_ms,
sample_rate=default_audio_config.sample_rate
)
logger.info("βœ… PhonemeMapper initialized for streaming")
except Exception as e:
logger.warning(f"⚠️ PhonemeMapper not available: {e}")
phoneme_mapper = None
if error_mapper_instance is None:
try:
error_mapper = ErrorMapper()
logger.info("βœ… ErrorMapper initialized for streaming")
except Exception as e:
logger.error(f"❌ ErrorMapper failed to initialize: {e}")
error_mapper = None
async def handle_streaming_websocket(websocket: WebSocket, session_id: Optional[str] = None):
"""
Handle WebSocket connection for streaming diagnosis.
Args:
websocket: WebSocket connection
session_id: Optional session ID (auto-generated if not provided)
"""
if inference_pipeline is None:
await websocket.close(code=1003, reason="Inference pipeline not loaded")
return
# Generate session ID
if not session_id:
session_id = str(uuid.uuid4())
# Accept connection
await websocket.accept()
logger.info(f"πŸ”Œ WebSocket connected: session_id={session_id}")
# Initialize buffer
window_size_samples = int(
inference_pipeline.inference_config.window_size_ms *
inference_pipeline.audio_config.sample_rate / 1000
)
hop_size_samples = int(
inference_pipeline.inference_config.hop_size_ms *
inference_pipeline.audio_config.sample_rate / 1000
)
buffer = StreamingBuffer(window_size_samples, hop_size_samples)
# Session metadata
streaming_sessions[session_id] = {
"session_id": session_id,
"connected_at": datetime.now(),
"frame_count": 0,
"total_latency_ms": 0.0
}
frame_index = 0
start_time = time.time()
try:
while True:
# Receive audio chunk
try:
data = await websocket.receive_bytes()
# Convert bytes to numpy array
# Assuming 16-bit PCM, mono, 16kHz
audio_chunk = np.frombuffer(data, dtype=np.int16).astype(np.float32) / 32768.0
# Add to buffer
buffer.add_chunk(audio_chunk)
# Process if buffer is ready
if buffer.get_frame() is not None:
frame_start_time = time.time()
# Get frame
frame = buffer.get_frame()
# Run inference
try:
result = inference_pipeline.predict_phone_level(
frame,
return_timestamps=False
)
if result.frame_predictions:
frame_pred = result.frame_predictions[0] # Single frame result
# Map to error detail
class_id = frame_pred.articulation_class
if frame_pred.fluency_label == 'stutter':
class_id += 4
error_detail = None
phoneme = '' # Streaming doesn't have text input
if error_mapper:
try:
error_detail_obj = error_mapper.map_classifier_output(
class_id=class_id,
confidence=frame_pred.confidence,
phoneme=phoneme,
fluency_label=frame_pred.fluency_label
)
if error_detail_obj.error_type != ErrorType.NORMAL:
error_detail = ErrorDetailSchema(
phoneme=error_detail_obj.phoneme,
error_type=error_detail_obj.error_type.value,
wrong_sound=error_detail_obj.wrong_sound,
severity=error_detail_obj.severity,
confidence=error_detail_obj.confidence,
therapy=error_detail_obj.therapy,
frame_indices=[frame_index]
)
except Exception as e:
logger.warning(f"Error mapping failed: {e}")
# Calculate latency
latency_ms = (time.time() - frame_start_time) * 1000
# Get severity level
severity_level = "none"
if error_detail and error_mapper:
severity_level = error_mapper.get_severity_level(error_detail.severity).value
# Create response
response = StreamingDiagnosisResponse(
session_id=session_id,
frame_id=frame_index,
timestamp=frame_index * (inference_pipeline.inference_config.hop_size_ms / 1000.0),
phoneme=phoneme,
fluency=FluencyInfo(
label=frame_pred.fluency_label,
confidence=frame_pred.fluency_prob if frame_pred.fluency_label == 'stutter' else (1.0 - frame_pred.fluency_prob)
),
articulation=ArticulationInfo(
label=frame_pred.articulation_label,
confidence=frame_pred.confidence,
class_id=frame_pred.articulation_class
),
error=error_detail,
severity_level=severity_level,
confidence=frame_pred.confidence,
latency_ms=latency_ms
)
# Send response
await websocket.send_json(response.model_dump())
# Update session stats
streaming_sessions[session_id]["frame_count"] += 1
streaming_sessions[session_id]["total_latency_ms"] += latency_ms
# Check latency requirement
if latency_ms > 50.0:
logger.warning(f"⚠️ Latency exceeded 50ms: {latency_ms:.1f}ms")
# Slide buffer
buffer.slide()
frame_index += 1
except Exception as e:
logger.error(f"❌ Inference failed: {e}", exc_info=True)
await websocket.send_json({
"error": f"Inference failed: {str(e)}",
"frame_id": frame_index
})
except Exception as e:
logger.error(f"❌ Error processing chunk: {e}", exc_info=True)
await websocket.send_json({
"error": f"Processing failed: {str(e)}",
"frame_id": frame_index
})
except WebSocketDisconnect:
logger.info(f"πŸ”Œ WebSocket disconnected: session_id={session_id}")
except Exception as e:
logger.error(f"❌ WebSocket error: {e}", exc_info=True)
finally:
# Cleanup session
if session_id in streaming_sessions:
session_data = streaming_sessions[session_id]
avg_latency = session_data["total_latency_ms"] / session_data["frame_count"] if session_data["frame_count"] > 0 else 0.0
logger.info(f"πŸ“Š Session {session_id} stats: {session_data['frame_count']} frames, "
f"avg_latency={avg_latency:.1f}ms")
del streaming_sessions[session_id]