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WIP ARQ - moved calculate_fft to audio

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DJ2LS 2023-12-16 11:05:53 +01:00
parent 7394e8b31b
commit 681bc5703f
3 changed files with 138 additions and 166 deletions
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124
modem/audio.py
View file

@ -7,6 +7,7 @@ import crcengine
import sounddevice as sd
import structlog
import numpy as np
import queue
atexit.register(sd._terminate)
@ -206,4 +207,125 @@ def set_audio_volume(datalist: np.ndarray, dB: float) -> np.ndarray:
scaled_data = datalist * scale_factor
# Clip values to int16 range and convert data type
return np.clip(scaled_data, -32768, 32767).astype(np.int16)
return np.clip(scaled_data, -32768, 32767).astype(np.int16)
RMS_COUNTER = 0
CHANNEL_BUSY_DELAY = 0
def calculate_fft(data, fft_queue, states) -> None:
"""
Calculate an average signal strength of the channel to assess
whether the channel is "busy."
"""
# Initialize dbfs counter
# rms_counter = 0
# https://gist.github.com/ZWMiller/53232427efc5088007cab6feee7c6e4c
# Fast Fourier Transform, 10*log10(abs) is to scale it to dB
# and make sure it's not imaginary
global RMS_COUNTER, CHANNEL_BUSY_DELAY
try:
fftarray = np.fft.rfft(data)
# Set value 0 to 1 to avoid division by zero
fftarray[fftarray == 0] = 1
dfft = 10.0 * np.log10(abs(fftarray))
# get average of dfft
avg = np.mean(dfft)
# Detect signals which are higher than the
# average + 10 (+10 smoothes the output).
# Data higher than the average must be a signal.
# Therefore we are setting it to 100 so it will be highlighted
# Have to do this when we are not transmitting so our
# own sending data will not affect this too much
if not states.isTransmitting():
dfft[dfft > avg + 15] = 100
# Calculate audio dbfs
# https://stackoverflow.com/a/9763652
# calculate dbfs every 50 cycles for reducing CPU load
RMS_COUNTER += 1
if RMS_COUNTER > 5:
d = np.frombuffer(data, np.int16).astype(np.float32)
# calculate RMS and then dBFS
# https://dsp.stackexchange.com/questions/8785/how-to-compute-dbfs
# try except for avoiding runtime errors by division/0
try:
rms = int(np.sqrt(np.max(d ** 2)))
if rms == 0:
raise ZeroDivisionError
audio_dbfs = 20 * np.log10(rms / 32768)
states.set("audio_dbfs", audio_dbfs)
except Exception as e:
states.set("audio_dbfs", -100)
RMS_COUNTER = 0
# Convert data to int to decrease size
dfft = dfft.astype(int)
# Create list of dfft
dfftlist = dfft.tolist()
# Reduce area where the busy detection is enabled
# We want to have this in correlation with mode bandwidth
# TODO This is not correctly and needs to be checked for correct maths
# dfftlist[0:1] = 10,15Hz
# Bandwidth[Hz] / 10,15
# narrowband = 563Hz = 56
# wideband = 1700Hz = 167
# 1500Hz = 148
# 2700Hz = 266
# 3200Hz = 315
# slot
slot = 0
slot1 = [0, 65]
slot2 = [65,120]
slot3 = [120, 176]
slot4 = [176, 231]
slot5 = [231, len(dfftlist)]
slotbusy = [False,False,False,False,False]
# Set to true if we should increment delay count; else false to decrement
addDelay=False
for range in [slot1, slot2, slot3, slot4, slot5]:
range_start = range[0]
range_end = range[1]
# define the area, we are detecting busy state
slotdfft = dfft[range_start:range_end]
# Check for signals higher than average by checking for "100"
# If we have a signal, increment our channel_busy delay counter
# so we have a smoother state toggle
if np.sum(slotdfft[slotdfft > avg + 15]) >= 200 and not states.isTransmitting():
addDelay=True
slotbusy[slot]=True
#states.channel_busy_slot[slot] = True
# increment slot
slot += 1
states.set_channel_slot_busy(slotbusy)
if addDelay:
# Limit delay counter to a maximum of 200. The higher this value,
# the longer we will wait until releasing state
states.set("channel_busy", True)
CHANNEL_BUSY_DELAY = min(CHANNEL_BUSY_DELAY + 10, 200)
else:
# Decrement channel busy counter if no signal has been detected.
CHANNEL_BUSY_DELAY = max(CHANNEL_BUSY_DELAY - 1, 0)
# When our channel busy counter reaches 0, toggle state to False
if CHANNEL_BUSY_DELAY == 0:
states.set("channel_busy", False)
# erase queue if greater than 10
if fft_queue.qsize() >= 10:
fft_queue = queue.Queue()
fft_queue.put(dfftlist[:315]) # 315 --> bandwidth 3200
except Exception as err:
print(f"[MDM] calculate_fft: Exception: {err}")
print("[MDM] Setting fft=0")
fft_queue.put([0])

60
modem/demodulator.py
View file

@ -12,7 +12,7 @@ TESTMODE = False
class Demodulator():
def __init__(self, config, audio_rx_q, modem_rx_q, data_q_rx, states, event_manager):
def __init__(self, config, audio_rx_q, modem_rx_q, data_q_rx, states, event_manager, fft_queue):
self.tuning_range_fmin = config['MODEM']['tuning_range_fmin']
self.tuning_range_fmax = config['MODEM']['tuning_range_fmax']
self.enable_fsk = config['MODEM']['enable_fsk']
@ -40,6 +40,8 @@ class Demodulator():
self.states = states
self.event_manager = event_manager
self.fft_queue = fft_queue
# init codec2 resampler
self.resampler = codec2.resampler()
@ -226,21 +228,14 @@ class Demodulator():
"dat0-datac3"
)
def sd_input_audio_callback(self, indata: np.ndarray, frames: int, time, status) -> None:
x = np.frombuffer(indata, dtype=np.int16)
x = self.resampler.resample48_to_8(x)
x = audio.set_audio_volume(x, self.rx_audio_level)
audio_48k = np.frombuffer(indata, dtype=np.int16)
audio_8k = self.resampler.resample48_to_8(audio_48k)
audio.calculate_fft(audio_8k, self.fft_queue, self.states)
# audio recording for debugging purposes
# TODO Find a nice place for this
#if AudioParam.audio_record:
# AudioParam.audio_record_file.writeframes(x)
audio_8k_level_adjusted = audio.set_audio_volume(audio_8k, self.rx_audio_level)
# Avoid decoding when transmitting to reduce CPU
# TODO Overriding this for testing purposes
# if not self.states.is_transmitting:
length_x = len(x)
length_audio_8k_level_adjusted = len(audio_8k_level_adjusted)
# Avoid buffer overflow by filling only if buffer for
# selected datachannel mode is not full
for audiobuffer, receive, index in [
@ -249,12 +244,12 @@ class Demodulator():
(self.dat0_datac3_buffer, self.RECEIVE_DATAC3, 3),
(self.dat0_datac4_buffer, self.RECEIVE_DATAC4, 4),
]:
if (audiobuffer.nbuffer + length_x) > audiobuffer.size:
if (audiobuffer.nbuffer + length_audio_8k_level_adjusted) > audiobuffer.size:
self.buffer_overflow_counter[index] += 1
self.event_manager.send_buffer_overflow(self.buffer_overflow_counter)
elif receive:
audiobuffer.push(x)
return x
audiobuffer.push(audio_8k_level_adjusted)
return audio_8k_level_adjusted
def worker_received(self) -> None:
"""Worker for FIFO queue for processing received frames"""
@ -409,7 +404,7 @@ class Demodulator():
x = np.frombuffer(x, dtype=np.int16)
# x = self.resampler.resample48_to_8(x)
self.calculate_fft(x)
audio.calculate_fft(x, self.fft_queue, self.states)
length_x = len(x)
for data_buffer, receive in [
@ -424,37 +419,6 @@ class Demodulator():
):
data_buffer.push(x)
def mkfifo_read_callback(self) -> None:
"""
Support testing by reading the audio data from a pipe and
depositing the data into the codec data buffers.
"""
while True:
threading.Event().wait(0.01)
# -----read
data_in48k = bytes()
with open("", "rb") as fifo:
for line in fifo:
data_in48k += line
while len(data_in48k) >= 48:
x = np.frombuffer(data_in48k[:48], dtype=np.int16)
x = self.resampler.resample48_to_8(x)
data_in48k = data_in48k[48:]
length_x = len(x)
for data_buffer, receive in [
(self.signalling_datac13_buffer, self.RECEIVE_SIGNALLING),
(self.dat0_datac1_buffer, self.RECEIVE_DATAC1),
(self.dat0_datac3_buffer, self.RECEIVE_DATAC3),
(self.dat0_datac4_buffer, self.RECEIVE_DATAC4),
]:
if (
not (data_buffer.nbuffer + length_x) > data_buffer.size
and receive
):
data_buffer.push(x)
def set_frames_per_burst(self, frames_per_burst: int) -> None:
"""
Configure codec2 to send the configured number of frames per burst.

120
modem/modem.py
View file

@ -60,7 +60,6 @@ class RF:
self.tci_ip = config['TCI']['tci_ip']
self.tci_port = config['TCI']['tci_port']
self.channel_busy_delay = 0
self.AUDIO_SAMPLE_RATE = 48000
self.MODEM_SAMPLE_RATE = codec2.api.FREEDV_FS_8000
@ -96,7 +95,9 @@ class RF:
self.modem_received_queue,
self.data_queue_received,
self.states,
self.event_manager)
self.event_manager,
self.fft_queue
)
self.beacon = beacon.Beacon(self.config, self.states, event_queue,
self.log, self.modem_transmit_queue)
@ -513,7 +514,6 @@ class RF:
def transmit_audio(self, audio_48k) -> None:
self.radio.set_ptt(True)
self.event_manager.send_ptt_change(True)
self.calculate_fft(audio_48k)
if self.radiocontrol in ["tci"]:
self.tci_tx_callback(audio_48k)
@ -590,117 +590,3 @@ class RF:
)
threading.Event().wait(1)
def calculate_fft(self, data) -> None:
"""
Calculate an average signal strength of the channel to assess
whether the channel is "busy."
"""
# Initialize dbfs counter
# rms_counter = 0
# https://gist.github.com/ZWMiller/53232427efc5088007cab6feee7c6e4c
# Fast Fourier Transform, 10*log10(abs) is to scale it to dB
# and make sure it's not imaginary
try:
fftarray = np.fft.rfft(data)
# Set value 0 to 1 to avoid division by zero
fftarray[fftarray == 0] = 1
dfft = 10.0 * np.log10(abs(fftarray))
# get average of dfft
avg = np.mean(dfft)
# Detect signals which are higher than the
# average + 10 (+10 smoothes the output).
# Data higher than the average must be a signal.
# Therefore we are setting it to 100 so it will be highlighted
# Have to do this when we are not transmitting so our
# own sending data will not affect this too much
if not self.states.isTransmitting():
dfft[dfft > avg + 15] = 100
# Calculate audio dbfs
# https://stackoverflow.com/a/9763652
# calculate dbfs every 50 cycles for reducing CPU load
self.rms_counter += 1
if self.rms_counter > 5:
d = np.frombuffer(data, np.int16).astype(np.float32)
# calculate RMS and then dBFS
# https://dsp.stackexchange.com/questions/8785/how-to-compute-dbfs
# try except for avoiding runtime errors by division/0
try:
rms = int(np.sqrt(np.max(d ** 2)))
if rms == 0:
raise ZeroDivisionError
audio_dbfs = 20 * np.log10(rms / 32768)
self.states.set("audio_dbfs", audio_dbfs)
except Exception as e:
self.states.set("audio_dbfs", -100)
self.rms_counter = 0
# Convert data to int to decrease size
dfft = dfft.astype(int)
# Create list of dfft
dfftlist = dfft.tolist()
# Reduce area where the busy detection is enabled
# We want to have this in correlation with mode bandwidth
# TODO This is not correctly and needs to be checked for correct maths
# dfftlist[0:1] = 10,15Hz
# Bandwidth[Hz] / 10,15
# narrowband = 563Hz = 56
# wideband = 1700Hz = 167
# 1500Hz = 148
# 2700Hz = 266
# 3200Hz = 315
# slot
slot = 0
slot1 = [0, 65]
slot2 = [65,120]
slot3 = [120, 176]
slot4 = [176, 231]
slot5 = [231, len(dfftlist)]
slotbusy = [False,False,False,False,False]
# Set to true if we should increment delay count; else false to decrement
addDelay=False
for range in [slot1, slot2, slot3, slot4, slot5]:
range_start = range[0]
range_end = range[1]
# define the area, we are detecting busy state
slotdfft = dfft[range_start:range_end]
# Check for signals higher than average by checking for "100"
# If we have a signal, increment our channel_busy delay counter
# so we have a smoother state toggle
if np.sum(slotdfft[slotdfft > avg + 15]) >= 200 and not self.states.isTransmitting():
addDelay=True
slotbusy[slot]=True
#self.states.channel_busy_slot[slot] = True
# increment slot
slot += 1
self.states.set_channel_slot_busy(slotbusy)
if addDelay:
# Limit delay counter to a maximum of 200. The higher this value,
# the longer we will wait until releasing state
self.states.set("channel_busy", True)
self.channel_busy_delay = min(self.channel_busy_delay + 10, 200)
else:
# Decrement channel busy counter if no signal has been detected.
self.channel_busy_delay = max(self.channel_busy_delay - 1, 0)
# When our channel busy counter reaches 0, toggle state to False
if self.channel_busy_delay == 0:
self.states.set("channel_busy", False)
# erase queue if greater than 10
if self.fft_queue.qsize() >= 10:
self.fft_queue = queue.Queue()
self.fft_queue.put(dfftlist[:315]) # 315 --> bandwidth 3200
except Exception as err:
self.log.error(f"[MDM] calculate_fft: Exception: {err}")
self.log.debug("[MDM] Setting fft=0")
# else 0
self.fft_queue.put([0])