mirror of
https://github.com/justLV/onju-v2
synced 2026-04-21 15:47:55 +00:00
7bcb94833c
PTT devices (--device name=ip:ptt): skip VAD, buffer audio until packets stop, skip LED commands, interrupt in-flight responses on new audio. Auto-detected from multicast "PTT" announcement. HTTP control server on :3002 for runtime device management: POST/GET/DELETE /devices Firmware: replace per-chunk DC offset with IIR filter to eliminate zipper noise at chunk boundaries (m5_echo + onjuino). Protocol: TCP timeouts use actual timeout param, failures are silent for non-critical commands (LED blink). Pipeline: labeled error logging (ASR/LLM/TTS), env var resolution warning, Gemini OpenAI-compatible endpoint support. Test scripts: rewritten to use pipeline modules, delete redundant test_opus_tts.py, add pyproject.toml (replaces requirements.txt).
117 lines
3.7 KiB
Python
117 lines
3.7 KiB
Python
#!/usr/bin/env python3
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"""
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Receive and record audio from ESP32 via UDP.
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Auto-detects u-law compressed or raw PCM packets.
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Usage:
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python test_mic_receiver.py
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python test_mic_receiver.py --duration 30 --output recording.wav
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"""
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import argparse
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import socket
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import time
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import wave
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from collections import deque
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import numpy as np
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from pipeline.audio import decode_ulaw
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SAMPLE_RATE = 16000
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CHUNK_SIZE = 512
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CHUNK_BYTES_RAW = CHUNK_SIZE * 2
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CHUNK_BYTES_COMPRESSED = CHUNK_SIZE
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def main():
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parser = argparse.ArgumentParser(description="ESP32 mic UDP receiver")
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parser.add_argument("--port", type=int, default=3000, help="UDP port (default: 3000)")
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parser.add_argument("--output", type=str, default="test_recording.wav", help="Output WAV file")
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parser.add_argument("--duration", type=int, default=10, help="Recording duration in seconds (0 = infinite)")
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args = parser.parse_args()
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sock = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
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sock.bind(("0.0.0.0", args.port))
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sock.settimeout(1.0)
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print(f"Listening on UDP port {args.port}...")
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print(f"Output: {args.output}")
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print("Waiting for packets...\n")
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audio_frames = []
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packet_count = 0
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total_bytes = 0
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start_time = None
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last_stats_time = None
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rms_history = deque(maxlen=20)
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try:
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while True:
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try:
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data, addr = sock.recvfrom(2048)
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if start_time is None:
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start_time = time.time()
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last_stats_time = start_time
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print(f"Receiving from {addr[0]}:{addr[1]}\n")
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packet_count += 1
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total_bytes += len(data)
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if len(data) == CHUNK_BYTES_COMPRESSED:
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samples = decode_ulaw(data)
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mode = "u-law"
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elif len(data) == CHUNK_BYTES_RAW:
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samples = np.frombuffer(data, dtype=np.int16)
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mode = "raw"
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else:
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continue
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audio_frames.append(samples)
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rms = np.sqrt(np.mean(samples.astype(np.float32) ** 2))
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rms_history.append(rms)
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current_time = time.time()
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if current_time - last_stats_time >= 1.0:
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elapsed = current_time - start_time
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bandwidth_kbps = (total_bytes * 8) / (elapsed * 1000)
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avg_rms = np.mean(list(rms_history))
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print(f"\r[{elapsed:6.1f}s] Packets: {packet_count:4d} | "
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f"Bandwidth: {bandwidth_kbps:5.1f} kbps | "
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f"RMS: {avg_rms:6.0f} | "
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f"Mode: {mode}",
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end="", flush=True)
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last_stats_time = current_time
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if args.duration > 0 and (current_time - start_time) >= args.duration:
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print("\n\nRecording duration reached.")
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break
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except socket.timeout:
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if start_time is not None:
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print("\n\nTimeout - no packets for 1 second")
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break
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continue
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except KeyboardInterrupt:
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print("\n\nInterrupted")
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if not audio_frames:
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print("No audio received!")
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return
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audio_data = np.concatenate(audio_frames)
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with wave.open(args.output, "wb") as wf:
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wf.setnchannels(1)
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wf.setsampwidth(2)
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wf.setframerate(SAMPLE_RATE)
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wf.writeframes(audio_data.tobytes())
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duration = len(audio_data) / SAMPLE_RATE
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transmitted_kb = total_bytes / 1024
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print(f"\nSaved {duration:.1f}s to {args.output} ({transmitted_kb:.1f} KB received, {packet_count} packets)")
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if __name__ == "__main__":
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main()
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