avm/hassio-addons
1
0
Fork 0
mirror of https://github.com/alexbelgium/hassio-addons.git synced 2026-01-11 18:31:02 +01:00
Code Issues Packages Projects Releases Wiki Activity

Update DOCS.md

Browse Source
This commit is contained in:
Alexandre
2024-10-31 23:11:10 +01:00
committed by GitHub
parent 9a6ddc450f
commit 92f258d597
1 changed files with 43 additions and 64 deletions
Download Patch File Download Diff File Expand all files Collapse all files

107
birdnet-pi/DOCS.md
View File

@@ -95,11 +95,11 @@ sleep 60
# Run focusrite and autogain scripts if present
if [ -f "$HOME/focusrite.sh" ]; then
"$HOME/focusrite.sh" >/tmp/log_focusrite 2>/tmp/log_focusrite_error &
python3 -u "$HOME/focusrite.sh" >/tmp/log_focusrite 2>/tmp/log_focusrite_error &
fi
if [ -f "$HOME/autogain.py" ]; then
"$HOME/autogain.py" >/tmp/log_autogain 2>/tmp/log_autogain_error &
python3 -u "$HOME/autogain.py" >/tmp/log_autogain 2>/tmp/log_autogain_error &
fi
```
@@ -408,7 +408,7 @@ Add this content in "$HOME/autogain.py" && chmod +x "$HOME/autogain.py"
#!/usr/bin/env python3
"""
Microphone Gain Adjustment Script with THD and Overload Detection
def set_gain_db
This script captures audio from an RTSP stream, processes it to calculate the RMS
within the 2000-8000 Hz frequency band, detects clipping, calculates Total Harmonic
Distortion (THD) over the full frequency range, and adjusts the microphone gain based
@@ -431,25 +431,20 @@ 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 = 40 # Capped at 40 dB
MAX_GAIN_DB = 40
DECREASE_GAIN_STEP_DB = 1
INCREASE_GAIN_STEP_DB = 5
CLIPPING_REDUCTION_DB = 3
@@ -462,10 +457,7 @@ NOISE_THRESHOLD_LOW = 0.00035
TREND_COUNT_THRESHOLD = 3
# Sampling Rate
SAMPLING_RATE = 44100 # Reduced from 48000 Hz to 44100 Hz
# Audio Capture Duration
AUDIO_CAPTURE_DURATION = 2 # Reduced from 5 seconds to 2 seconds
SAMPLING_RATE = 44100
# RTSP Stream URL
RTSP_URL = "rtsp://192.168.178.124:8554/birdmic"
@@ -485,11 +477,6 @@ 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
# -----------------------------------------------------------------------
@@ -499,7 +486,7 @@ def debug_print(msg, level="info"):
:param msg: The debug message to print.
:param level: Logging level - "info", "warning", "error".
"""
if DEBUG and not SUMMARY_MODE:
if DEBUG:
current_time = time.strftime("%Y-%m-%d %H:%M:%S", time.localtime())
print(f"[{current_time}] [{level.upper()}] {msg}")
@@ -528,25 +515,32 @@ 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:
return float(match.group(1))
gain_db = float(match.group(1))
debug_print(f"Retrieved gain: {gain_db} dB", "info")
return gain_db
else:
debug_print("No gain information found in amixer output.", "warning")
return None
except subprocess.CalledProcessError:
except subprocess.CalledProcessError as e:
debug_print(f"amixer sget failed: {e}", "error")
return None
def set_gain_db(mic_name, gain_db):
"""
Sets the gain of the specified microphone using amixer.
Suppresses output by redirecting to /dev/null.
"""
if int(gain_db) > MAX_GAIN_DB:
return True # Do not exceed max gain
gain_db_int = int(gain_db)
if gain_db_int > MAX_GAIN_DB:
debug_print(f"Requested gain {gain_db_int} dB exceeds MAX_GAIN_DB {MAX_GAIN_DB} dB. Skipping.", "warning")
return False # Do not exceed max gain
cmd = ['amixer', 'sset', mic_name, f'{gain_db}dB']
try:
subprocess.check_call(cmd, stderr=subprocess.STDOUT, stdout=subprocess.DEVNULL)
subprocess.check_call(cmd, stdout=subprocess.DEVNULL, stderr=subprocess.STDOUT)
debug_print(f"Set gain to: {gain_db} dB", "info")
return True
except subprocess.CalledProcessError:
except subprocess.CalledProcessError as e:
debug_print(f"amixer sset failed: {e}", "error")
return False
@@ -577,8 +571,8 @@ def thd_calculation(audio, sampling_rate, num_harmonics=5):
"""
Calculates Total Harmonic Distortion (THD) for the audio signal.
"""
fft_vals = rfft(audio)
fft_freqs = rfftfreq(len(audio), 1 / sampling_rate)
fft_vals = np.fft.rfft(audio)
fft_freqs = np.fft.rfftfreq(len(audio), 1 / sampling_rate)
fft_magnitude = np.abs(fft_vals)
fundamental_freq, fundamental_amplitude = find_fundamental_frequency(fft_freqs, fft_magnitude)
@@ -628,14 +622,13 @@ def detect_microphone_overload(spl, mic_clipping_spl):
return False
def calculate_noise_rms_and_thd(rtsp_url, bandpass_sos, sampling_rate):
def calculate_noise_rms_and_thd(rtsp_url, bandpass_sos, sampling_rate, num_bins=5):
"""
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', str(AUDIO_CAPTURE_DURATION), '-'
'-vn', '-f', 's16le', '-acodec', 'pcm_s16le', '-ar', str(sampling_rate), '-ac', '1', '-t', '5', '-'
]
retries = 3
@@ -672,25 +665,18 @@ def calculate_noise_rms_and_thd(rtsp_url, bandpass_sos, sampling_rate):
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 multiprocessing for audio capture and processing.
and adjusts microphone gain with retry logic for RTSP stream resilience.
"""
TREND_COUNT = 0
PREVIOUS_TREND = 0
# Precompute bandpass filter coefficients with updated SAMPLING_RATE and lower FILTER_ORDER
# Precompute bandpass filter coefficients with updated SAMPLING_RATE
LOWCUT = 2000
HIGHCUT = 8000
FILTER_ORDER = 5
sos = butter(FILTER_ORDER, [LOWCUT, HIGHCUT], btype='band', fs=SAMPLING_RATE, output='sos')
# Set the microphone gain to the maximum gain at the start
@@ -699,26 +685,14 @@ def main():
print(f"Microphone gain set to {MAX_GAIN_DB} dB at start.")
else:
print("Failed to set microphone gain at start. Exiting.")
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")
return
while True:
if not queue.empty():
rms, thd, spl, overload = queue.get()
else:
time.sleep(1)
continue
rms, thd, spl, overload = calculate_noise_rms_and_thd(RTSP_URL, sos, SAMPLING_RATE)
if rms is None:
print("Failed to compute noise RMS. Retrying in 1 minute...")
time.sleep(60)
continue
# Adjust gain if overload detected
@@ -728,10 +702,12 @@ 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")
current_gain_db = NEW_GAIN_DB # Update current gain
# Output summary log
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
summary_log(current_gain_db if current_gain_db else MIN_GAIN_DB, True, rms, thd)
continue # Skip trend adjustment in case of clipping
time.sleep(60)
continue
# Handle THD if SPL is above threshold
if spl >= THD_FUNDAMENTAL_THRESHOLD_DB:
@@ -742,7 +718,7 @@ 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")
current_gain_db = NEW_GAIN_DB # Update current gain
debug_print(f"Gain decreased to {NEW_GAIN_DB} dB due to high THD.", "info")
else:
debug_print("THD within acceptable limits.", "info")
else:
@@ -773,8 +749,11 @@ 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)
@@ -785,14 +764,14 @@ 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")
current_gain_db = NEW_GAIN_DB # Update current gain
debug_print(f"Gain decreased to {NEW_GAIN_DB} dB due to high noise.", "info")
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")
current_gain_db = NEW_GAIN_DB # Update current gain
debug_print(f"Gain increased to {NEW_GAIN_DB} dB due to low noise.", "info")
TREND_COUNT = 0
else:
debug_print("No gain adjustment needed based on noise trend.", "info")