serial_audio_catcher/tdoa_sharp_noises.py

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import numpy as np
import time
from datetime import datetime
FIFO_PATH = "/tmp/esp32_audio"
SAMPLE_RATE = 16000
CHANNELS = 2
BYTES_PER_SAMPLE = 2
# Detection and processing params
BLOCK_FRAMES = 2048 # ~128 ms @16k; large enough to catch an impulse
IMPULSE_WINDOW = 256 # samples around the detected peak for GCC-PHAT
BAND_LOW = 1000 # Hz
BAND_HIGH = 4000 # Hz
MARGIN = 3.0 # multiplier above rolling baseline for impulse detection
ALPHA = 0.01 # rolling baseline EMA smoothing
COOLDOWN = 0.5 # seconds; suppress retriggers from echoes
# Geometry
MIC_DISTANCE = 0.20 # meters between microphones
SPEED_OF_SOUND = 343.0 # m/s
def read_block(f, block_bytes):
data = f.read(block_bytes)
if not data or len(data) < block_bytes:
return None
return np.frombuffer(data, dtype=np.int16)
def bandpass_fft(x, fs, low, high):
"""Simple FFT band-pass: zero out bins outside [low, high]."""
n = len(x)
X = np.fft.rfft(x)
freqs = np.fft.rfftfreq(n, d=1.0/fs)
mask = (freqs >= low) & (freqs <= high)
X_filtered = X * mask
x_filtered = np.fft.irfft(X_filtered, n=n)
return x_filtered.astype(x.dtype)
def gcc_phat(sig, refsig, fs, max_tau=None, interp=1):
"""
GCC-PHAT lag estimation between sig and refsig.
Returns time delay (tau) in seconds.
"""
n = sig.shape[0] + refsig.shape[0]
# FFT
SIG = np.fft.rfft(sig, n=n)
REFSIG = np.fft.rfft(refsig, n=n)
R = SIG * np.conj(REFSIG)
denom = np.abs(R)
R = R / (denom + 1e-15)
cc = np.fft.irfft(R, n=(interp * n))
if max_tau is None:
# physical max tau based on mic distance
max_tau = MIC_DISTANCE / SPEED_OF_SOUND
max_shift = int(interp * fs * max_tau)
mid = cc.shape[0] // 2
cc = np.concatenate((cc[mid - max_shift: mid + max_shift + 1],))
shift = np.argmax(cc) - max_shift
tau = shift / float(interp * fs)
return tau
def tau_to_angle(tau, mic_distance, speed_of_sound):
"""
Convert time difference to angle (-90..+90) assuming linear 2-mic array and far-field.
"""
# clamp sin argument to [-1,1]
arg = (tau * speed_of_sound) / mic_distance
arg = max(-1.0, min(1.0, arg))
angle_rad = np.arcsin(arg)
return np.degrees(angle_rad)
def main():
block_bytes = BLOCK_FRAMES * CHANNELS * BYTES_PER_SAMPLE
baseline = None
last_trigger = 0.0
with open(FIFO_PATH, "rb") as f:
print("Listening: GCC-PHAT + impulse window + band-pass + cooldown")
while True:
audio = read_block(f, block_bytes)
if audio is None:
continue
# Split stereo
left = audio[0::2]
right = audio[1::2]
# Compute per-block peak level for rolling baseline
left_peak = np.max(np.abs(left))
right_peak = np.max(np.abs(right))
current_level = (left_peak + right_peak) / 2.0
if baseline is None:
baseline = current_level
continue
baseline = (1 - ALPHA) * baseline + ALPHA * current_level
threshold = baseline * MARGIN
# Cooldown gate
now = time.time()
if now - last_trigger < COOLDOWN:
continue
# Impulse gate: only proceed if strong spike above rolling threshold
if max(left_peak, right_peak) <= threshold:
continue
# Find impulse index using combined magnitude
combined = np.abs(left) + np.abs(right)
peak_idx = int(np.argmax(combined))
# Window around impulse for robust localization
half = IMPULSE_WINDOW // 2
start = max(0, peak_idx - half)
end = min(len(left), peak_idx + half)
l_win = left[start:end]
r_win = right[start:end]
# Band-pass to 14 kHz to reduce low rumble/high hiss
l_bp = bandpass_fft(l_win.astype(np.float32), SAMPLE_RATE, BAND_LOW, BAND_HIGH)
r_bp = bandpass_fft(r_win.astype(np.float32), SAMPLE_RATE, BAND_LOW, BAND_HIGH)
# GCC-PHAT for TDOA
# cap max_tau to physical limit to avoid spurious peaks
max_tau = MIC_DISTANCE / SPEED_OF_SOUND
tau = gcc_phat(l_bp, r_bp, SAMPLE_RATE, max_tau=max_tau, interp=1)
angle = tau_to_angle(tau, MIC_DISTANCE, SPEED_OF_SOUND)
# Timestamp and report
ts = datetime.now().strftime("%Y-%m-%d %H:%M:%S.%f")[:-3]
louder = "LEFT" if left_peak > right_peak else "RIGHT"
print(f"[{ts}] Loud impulse: {louder} louder | TDOA={tau*1000:.2f} ms | angle≈{angle:.1f}° "
f"(baseline={baseline:.1f}, L={left_peak}, R={right_peak})")
# Arm cooldown
last_trigger = now
if __name__ == "__main__":
main()