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Alexandre
2024-10-28 08:02:44 +01:00
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parent 6ff3c5bd84
commit 44e8ebe04c
1 changed files with 57 additions and 34 deletions
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birdnet-pi/DOCS.md
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@@ -455,20 +455,25 @@ Changelog:
- 2024-10-27: Enhanced debug output with logging levels.
- 2024-10-28: Added summary log mode for simplified output.
- 2024-10-28: Removed gain stabilization delay for immediate gain adjustments.
- 2024-10-28: Max gain capped at 40 dB, suppressed `amixer` logging.
- 2024-10-28: Implemented sampling rate reduction, capture duration reduction, scipy.fft usage, lower filter order, and multiprocessing for efficiency.
"""
import subprocess
import numpy as np
from scipy.signal import butter, sosfilt, find_peaks
from scipy.fft import rfft, rfftfreq
import time
import re
import multiprocessing as mp
import sys
# ---------------------------- Configuration ----------------------------
# Microphone Settings
MICROPHONE_NAME = "Line In 1 Gain"
MIN_GAIN_DB = 20
MAX_GAIN_DB = 45
MAX_GAIN_DB = 40 # Capped at 40 dB
DECREASE_GAIN_STEP_DB = 1
INCREASE_GAIN_STEP_DB = 5
CLIPPING_REDUCTION_DB = 3
@@ -481,7 +486,10 @@ NOISE_THRESHOLD_LOW = 0.00035
TREND_COUNT_THRESHOLD = 3
# Sampling Rate
SAMPLING_RATE = 48000 # Updated from 32000 to 48000 Hz
SAMPLING_RATE = 44100 # Reduced from 48000 Hz to 44100 Hz
# Audio Capture Duration
AUDIO_CAPTURE_DURATION = 2 # Reduced from 5 seconds to 2 seconds
# RTSP Stream URL
RTSP_URL = "rtsp://192.168.178.124:8554/birdmic"
@@ -501,6 +509,11 @@ REFERENCE_PRESSURE = 20e-6
THD_FUNDAMENTAL_THRESHOLD_DB = 60
MAX_THD_PERCENTAGE = 5.0
# Filter Settings
LOWCUT = 2000
HIGHCUT = 8000
FILTER_ORDER = 3 # Reduced from 5 to 3 for efficiency
# -----------------------------------------------------------------------
@@ -539,28 +552,23 @@ def get_gain_db(mic_name):
output = subprocess.check_output(cmd, stderr=subprocess.STDOUT).decode()
match = re.search(r'\[(-?\d+(\.\d+)?)dB\]', output)
if match:
gain_db = float(match.group(1))
debug_print(f"Retrieved gain: {gain_db} dB", "info")
return gain_db
return float(match.group(1))
else:
debug_print("No gain information found in amixer output.", "warning")
return None
except subprocess.CalledProcessError as e:
debug_print(f"amixer sget failed: {e}", "error")
except subprocess.CalledProcessError:
return None
def set_gain_db(mic_name, gain_db):
"""
Sets the gain of the specified microphone using amixer.
Suppresses logging related to this action.
"""
cmd = ['amixer', 'sset', mic_name, f'{gain_db}dB']
try:
subprocess.check_call(cmd, stderr=subprocess.STDOUT)
debug_print(f"Set gain to: {gain_db} dB", "info")
return True
except subprocess.CalledProcessError as e:
debug_print(f"amixer sset failed: {e}", "error")
except subprocess.CalledProcessError:
return False
@@ -591,8 +599,8 @@ def thd_calculation(audio, sampling_rate, num_harmonics=5):
"""
Calculates Total Harmonic Distortion (THD) for the audio signal.
"""
fft_vals = np.fft.rfft(audio)
fft_freqs = np.fft.rfftfreq(len(audio), 1 / sampling_rate)
fft_vals = rfft(audio)
fft_freqs = rfftfreq(len(audio), 1 / sampling_rate)
fft_magnitude = np.abs(fft_vals)
fundamental_freq, fundamental_amplitude = find_fundamental_frequency(fft_freqs, fft_magnitude)
@@ -642,13 +650,14 @@ def detect_microphone_overload(spl, mic_clipping_spl):
return False
def calculate_noise_rms_and_thd(rtsp_url, bandpass_sos, sampling_rate, num_bins=5):
def calculate_noise_rms_and_thd(rtsp_url, bandpass_sos, sampling_rate):
"""
Captures audio from an RTSP stream, calculates RMS, THD, and SPL, and detects microphone overload.
"""
cmd = [
'ffmpeg', '-loglevel', 'error', '-rtsp_transport', 'tcp', '-i', rtsp_url,
'-vn', '-f', 's16le', '-acodec', 'pcm_s16le', '-ar', str(sampling_rate), '-ac', '1', '-t', '5', '-'
'-vn', '-f', 's16le', '-acodec', 'pcm_s16le', '-ar', str(sampling_rate),
'-ac', '1', '-t', str(AUDIO_CAPTURE_DURATION), '-'
]
retries = 3
@@ -685,18 +694,25 @@ def calculate_noise_rms_and_thd(rtsp_url, bandpass_sos, sampling_rate, num_bins=
return None, None, None, False
def audio_capture_process(queue, rtsp_url, sos, sampling_rate):
"""
Separate process for capturing and processing audio.
"""
while True:
rms, thd, spl, overload = calculate_noise_rms_and_thd(rtsp_url, sos, sampling_rate)
queue.put((rms, thd, spl, overload))
time.sleep(60) # Wait for the next capture cycle
def main():
"""
Main loop that continuously monitors background noise, detects clipping, calculates THD,
and adjusts microphone gain with retry logic for RTSP stream resilience.
and adjusts microphone gain with multiprocessing for audio capture and processing.
"""
TREND_COUNT = 0
PREVIOUS_TREND = 0
# Precompute bandpass filter coefficients with updated SAMPLING_RATE
LOWCUT = 2000
HIGHCUT = 8000
FILTER_ORDER = 5
# Precompute bandpass filter coefficients with updated SAMPLING_RATE and lower FILTER_ORDER
sos = butter(FILTER_ORDER, [LOWCUT, HIGHCUT], btype='band', fs=SAMPLING_RATE, output='sos')
# Set the microphone gain to the maximum gain at the start
@@ -705,14 +721,26 @@ def main():
print(f"Microphone gain set to {MAX_GAIN_DB} dB at start.")
else:
print("Failed to set microphone gain at start. Exiting.")
return
sys.exit(1)
# Initialize multiprocessing queue
manager = mp.Manager()
queue = manager.Queue()
# Start audio capture and processing in a separate process
p = mp.Process(target=audio_capture_process, args=(queue, RTSP_URL, sos, SAMPLING_RATE))
p.start()
debug_print("Started audio capture and processing process.", "info")
while True:
rms, thd, spl, overload = calculate_noise_rms_and_thd(RTSP_URL, sos, SAMPLING_RATE)
if not queue.empty():
rms, thd, spl, overload = queue.get()
else:
time.sleep(1)
continue
if rms is None:
print("Failed to compute noise RMS. Retrying in 1 minute...")
time.sleep(60)
continue
# Adjust gain if overload detected
@@ -722,12 +750,10 @@ def main():
NEW_GAIN_DB = max(current_gain_db - CLIPPING_REDUCTION_DB, MIN_GAIN_DB)
if set_gain_db(MICROPHONE_NAME, NEW_GAIN_DB):
print(f"Clipping detected. Reduced gain to {NEW_GAIN_DB} dB")
debug_print(f"Gain reduced to {NEW_GAIN_DB} dB due to clipping.", "warning")
# No stabilization delay; continue to next iteration
# Skip trend adjustment in case of clipping
current_gain_db = NEW_GAIN_DB # Update current gain
# Output summary log
summary_log(current_gain_db if current_gain_db else MIN_GAIN_DB, True, rms, thd)
time.sleep(60)
continue
continue # Skip trend adjustment in case of clipping
# Handle THD if SPL is above threshold
if spl >= THD_FUNDAMENTAL_THRESHOLD_DB:
@@ -738,7 +764,7 @@ def main():
NEW_GAIN_DB = max(current_gain_db - DECREASE_GAIN_STEP_DB, MIN_GAIN_DB)
if set_gain_db(MICROPHONE_NAME, NEW_GAIN_DB):
print(f"High THD detected. Decreased gain to {NEW_GAIN_DB} dB")
debug_print(f"Gain decreased to {NEW_GAIN_DB} dB due to high THD.", "info")
current_gain_db = NEW_GAIN_DB # Update current gain
else:
debug_print("THD within acceptable limits.", "info")
else:
@@ -769,11 +795,8 @@ def main():
if current_gain_db is None:
print("Failed to get current gain level. Retrying in 1 minute...")
time.sleep(60)
continue
debug_print(f"Current gain: {current_gain_db} dB", "info")
# Output summary log for the current state
summary_log(current_gain_db, overload, rms, thd)
@@ -784,14 +807,14 @@ def main():
NEW_GAIN_DB = max(current_gain_db - DECREASE_GAIN_STEP_DB, MIN_GAIN_DB)
if set_gain_db(MICROPHONE_NAME, NEW_GAIN_DB):
print(f"Background noise high. Decreased gain to {NEW_GAIN_DB} dB")
debug_print(f"Gain decreased to {NEW_GAIN_DB} dB due to high noise.", "info")
current_gain_db = NEW_GAIN_DB # Update current gain
TREND_COUNT = 0
elif CURRENT_TREND == -1:
# Increase gain by INCREASE_GAIN_STEP_DB dB
NEW_GAIN_DB = min(current_gain_db + INCREASE_GAIN_STEP_DB, MAX_GAIN_DB)
if set_gain_db(MICROPHONE_NAME, NEW_GAIN_DB):
print(f"Background noise low. Increased gain to {NEW_GAIN_DB} dB")
debug_print(f"Gain increased to {NEW_GAIN_DB} dB due to low noise.", "info")
current_gain_db = NEW_GAIN_DB # Update current gain
TREND_COUNT = 0
else:
debug_print("No gain adjustment needed based on noise trend.", "info")