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FreeDATA/tnc/modem.py
dj2ls e8283b5db0 integrated multimode callback test & FIFO queue to tnc 
this is a first test to see how it performs
2021-12-20 15:38:43 +01:00

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#!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""
Created on Wed Dec 23 07:04:24 2020
@author: DJ2LS
"""
import sys
import ctypes
from ctypes import *
import pathlib
import audioop
#import asyncio
import logging, structlog, log_handler
import time
import threading
import atexit
import numpy as np
import helpers
import static
import data_handler
import re
import queue
import codec2
# option for testing miniaudio instead of audioop for sample rate conversion
#import miniaudio
####################################################
# https://stackoverflow.com/questions/7088672/pyaudio-working-but-spits-out-error-messages-each-time
# https://github.com/DJ2LS/FreeDATA/issues/22
# we need to have a look at this if we want to run this on Windows and MacOS !
# Currently it seems, this is a Linux-only problem
from ctypes import *
from contextlib import contextmanager
import pyaudio
ERROR_HANDLER_FUNC = CFUNCTYPE(None, c_char_p, c_int, c_char_p, c_int, c_char_p)
def py_error_handler(filename, line, function, err, fmt):
pass
c_error_handler = ERROR_HANDLER_FUNC(py_error_handler)
@contextmanager
def noalsaerr():
asound = cdll.LoadLibrary('libasound.so')
asound.snd_lib_error_set_handler(c_error_handler)
yield
asound.snd_lib_error_set_handler(None)
# with noalsaerr():
# p = pyaudio.PyAudio()
######################################################
# try importing hamlib
try:
# get python version
python_version = str(sys.version_info[0]) + "." + str(sys.version_info[1])
# installation path for Ubuntu 20.04 LTS python modules
sys.path.append('/usr/local/lib/python'+ python_version +'/site-packages')
# installation path for Ubuntu 20.10 +
sys.path.append('/usr/local/lib/')
import Hamlib
# https://stackoverflow.com/a/4703409
hamlib_version = re.findall(r"[-+]?\d*\.?\d+|\d+", Hamlib.cvar.hamlib_version)
hamlib_version = float(hamlib_version[0])
min_hamlib_version = 4.1
if hamlib_version > min_hamlib_version:
structlog.get_logger("structlog").info("[TNC] Hamlib found", version=hamlib_version)
else:
structlog.get_logger("structlog").warning("[TNC] Hamlib outdated", found=hamlib_version, recommend=min_hamlib_version)
except Exception as e:
structlog.get_logger("structlog").critical("[TNC] Hamlib not found", error=e)
MODEM_STATS_NR_MAX = 320
MODEM_STATS_NC_MAX = 51
class MODEMSTATS(ctypes.Structure):
_fields_ = [
("Nc", ctypes.c_int),
("snr_est", ctypes.c_float),
("rx_symbols", (ctypes.c_float * MODEM_STATS_NR_MAX)*MODEM_STATS_NC_MAX),
("nr", ctypes.c_int),
("sync", ctypes.c_int),
("foff", ctypes.c_float),
("rx_timing", ctypes.c_float),
("clock_offset", ctypes.c_float),
("sync_metric", ctypes.c_float),
("pre", ctypes.c_int),
("post", ctypes.c_int),
("uw_fails", ctypes.c_int),
]
class RF():
def __init__(self):
self.AUDIO_SAMPLE_RATE_RX = 48000
self.AUDIO_SAMPLE_RATE_TX = 48000
self.MODEM_SAMPLE_RATE = codec2.api.FREEDV_FS_8000
self.AUDIO_FRAMES_PER_BUFFER_RX = 2400*2 #8192
self.AUDIO_FRAMES_PER_BUFFER_TX = 8 #8192 Lets to some tests with very small chunks for TX
self.AUDIO_CHUNKS = 48 #8 * (self.AUDIO_SAMPLE_RATE_RX/self.MODEM_SAMPLE_RATE) #48
self.AUDIO_CHANNELS = 1
# make sure our resampler will work
assert (self.AUDIO_SAMPLE_RATE_RX / self.MODEM_SAMPLE_RATE) == codec2.api.FDMDV_OS_48
# small hack for initializing codec2 via codec2.py module
# TODO: we need to change the entire modem module to integrate codec2 module
self.c_lib = codec2.api
self.resampler = codec2.resampler()
# init FIFO queue to store received frames in
self.dataqueue = queue.Queue()
# open codec2 instance
self.datac0_freedv = cast(codec2.api.freedv_open(codec2.api.FREEDV_MODE_DATAC0), c_void_p)
self.datac0_bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(self.datac0_freedv)/8)
self.datac0_bytes_out = create_string_buffer(self.datac0_bytes_per_frame * 2)
codec2.api.freedv_set_frames_per_burst(self.datac0_freedv,1)
self.datac0_buffer = codec2.audio_buffer(2*self.AUDIO_FRAMES_PER_BUFFER_RX)
self.datac1_freedv = cast(codec2.api.freedv_open(codec2.api.FREEDV_MODE_DATAC1), c_void_p)
self.datac1_bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(self.datac1_freedv)/8)
self.datac1_bytes_out = create_string_buffer(self.datac1_bytes_per_frame * 2)
codec2.api.freedv_set_frames_per_burst(self.datac1_freedv,1)
self.datac1_buffer = codec2.audio_buffer(2*self.AUDIO_FRAMES_PER_BUFFER_RX)
self.datac3_freedv = cast(codec2.api.freedv_open(codec2.api.FREEDV_MODE_DATAC3), c_void_p)
self.datac3_bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(self.datac3_freedv)/8)
self.datac3_bytes_out = create_string_buffer(self.datac3_bytes_per_frame * 2)
codec2.api.freedv_set_frames_per_burst(self.datac3_freedv,1)
self.datac3_buffer = codec2.audio_buffer(2*self.AUDIO_FRAMES_PER_BUFFER_RX)
# initial nin values
self.datac0_nin = codec2.api.freedv_nin(self.datac0_freedv)
self.datac1_nin = codec2.api.freedv_nin(self.datac1_freedv)
self.datac3_nin = codec2.api.freedv_nin(self.datac3_freedv)
# --------------------------------------------CREATE PYAUDIO INSTANCE
try:
# we need to "try" this, because sometimes libasound.so isn't in the default place
# try to supress error messages
with noalsaerr(): # https://github.com/DJ2LS/FreeDATA/issues/22
self.p = pyaudio.PyAudio()
# else do it the default way
except:
self.p = pyaudio.PyAudio()
atexit.register(self.p.terminate)
# --------------------------------------------OPEN RX AUDIO CHANNEL
# optional auto selection of loopback device if using in testmode
if static.AUDIO_INPUT_DEVICE == -2:
loopback_list = []
for dev in range(0,self.p.get_device_count()):
if 'Loopback: PCM' in self.p.get_device_info_by_index(dev)["name"]:
loopback_list.append(dev)
if len(loopback_list) >= 2:
AUDIO_INPUT_DEVICE = loopback_list[0] #0 = RX 1 = TX
print(f"loopback_list rx: {loopback_list}", file=sys.stderr)
self.stream_rx = self.p.open(format=pyaudio.paInt16,
channels=self.AUDIO_CHANNELS,
rate=self.AUDIO_SAMPLE_RATE_RX,
frames_per_buffer=self.AUDIO_FRAMES_PER_BUFFER_RX,
input=True,
output=False,
input_device_index=static.AUDIO_INPUT_DEVICE,
stream_callback=self.callback
)
# --------------------------------------------OPEN TX AUDIO CHANNEL
# optional auto selection of loopback device if using in testmode
if static.AUDIO_OUTPUT_DEVICE == -2:
loopback_list = []
for dev in range(0,self.p.get_device_count()):
if 'Loopback: PCM' in self.p.get_device_info_by_index(dev)["name"]:
loopback_list.append(dev)
if len(loopback_list) >= 2:
static.AUDIO_OUTPUT_DEVICE = loopback_list[1] #0 = RX 1 = TX
print(f"loopback_list tx: {loopback_list}", file=sys.stderr)
self.stream_tx = self.p.open(format=pyaudio.paInt16,
channels=self.AUDIO_CHANNELS,
rate=self.AUDIO_SAMPLE_RATE_TX,
frames_per_buffer=self.AUDIO_FRAMES_PER_BUFFER_TX, # n_nom_modem_samples
output=True,
output_device_index=static.AUDIO_OUTPUT_DEVICE, # static.AUDIO_OUTPUT_DEVICE
)
self.streambuffer = bytes(0)
self.audio_writing_to_stream = False
# --------------------------------------------START DECODER THREAD
DECODER_THREAD = threading.Thread(target=self.receive, name="DECODER_THREAD")
DECODER_THREAD.start()
WORKER_THREAD = threading.Thread(target=self.worker, name="WORKER_THREAD")
WORKER_THREAD.start()
self.fft_data = bytes()
FFT_THREAD = threading.Thread(target=self.calculate_fft, name="FFT_THREAD")
FFT_THREAD.start()
# --------------------------------------------CONFIGURE HAMLIB
# my_rig.set_ptt(Hamlib.RIG_PTT_RIG,0)
# my_rig.set_ptt(Hamlib.RIG_PTT_SERIAL_DTR,0)
# my_rig.set_ptt(Hamlib.RIG_PTT_SERIAL_RTS,1)
#self.my_rig.set_conf("dtr_state", "OFF")
#my_rig.set_conf("rts_state", "OFF")
#self.my_rig.set_conf("ptt_type", "RTS")
#my_rig.set_conf("ptt_type", "RIG_PTT_SERIAL_RTS")
# try to init hamlib
try:
Hamlib.rig_set_debug(Hamlib.RIG_DEBUG_NONE)
# get devicenumber by looking for deviceobject in Hamlib module
try:
devicenumber = getattr(Hamlib, static.HAMLIB_DEVICE_ID)
except:
structlog.get_logger("structlog").error("[DMN] Hamlib: rig not supported...")
devicenumber = 0
self.my_rig = Hamlib.Rig(int(devicenumber))
self.my_rig.set_conf("rig_pathname", static.HAMLIB_DEVICE_PORT)
self.my_rig.set_conf("retry", "5")
self.my_rig.set_conf("serial_speed", static.HAMLIB_SERIAL_SPEED)
self.my_rig.set_conf("serial_handshake", "None")
self.my_rig.set_conf("stop_bits", "1")
self.my_rig.set_conf("data_bits", "8")
if static.HAMLIB_PTT_TYPE == 'RIG':
self.hamlib_ptt_type = Hamlib.RIG_PTT_RIG
elif static.HAMLIB_PTT_TYPE == 'DTR-H':
self.hamlib_ptt_type = Hamlib.RIG_PTT_SERIAL_DTR
self.my_rig.set_conf("dtr_state", "HIGH")
self.my_rig.set_conf("ptt_type", "DTR")
elif static.HAMLIB_PTT_TYPE == 'DTR-L':
self.hamlib_ptt_type = Hamlib.RIG_PTT_SERIAL_DTR
self.my_rig.set_conf("dtr_state", "LOW")
self.my_rig.set_conf("ptt_type", "DTR")
elif static.HAMLIB_PTT_TYPE == 'RTS':
self.hamlib_ptt_type = Hamlib.RIG_PTT_SERIAL_RTS
self.my_rig.set_conf("dtr_state", "OFF")
self.my_rig.set_conf("ptt_type", "RTS")
elif static.HAMLIB_PTT_TYPE == 'PARALLEL':
self.hamlib_ptt_type = Hamlib.RIG_PTT_PARALLEL
elif static.HAMLIB_PTT_TYPE == 'MICDATA':
self.hamlib_ptt_type = Hamlib.RIG_PTT_RIG_MICDATA
elif static.HAMLIB_PTT_TYPE == 'CM108':
self.hamlib_ptt_type = Hamlib.RIG_PTT_CM108
else: # static.HAMLIB_PTT_TYPE == 'RIG_PTT_NONE':
self.hamlib_ptt_type = Hamlib.RIG_PTT_NONE
self.my_rig.open()
atexit.register(self.my_rig.close)
# set rig mode to USB
self.my_rig.set_mode(Hamlib.RIG_MODE_USB)
# start thread for getting hamlib data
HAMLIB_THREAD = threading.Thread(target=self.get_radio_stats, name="HAMLIB_THREAD")
HAMLIB_THREAD.start()
except:
structlog.get_logger("structlog").error("[TNC] Hamlib - can't open rig", e=sys.exc_info()[0])
# --------------------------------------------------------------------------------------------------------
def callback(self, data_in48k, frame_count, time_info, status):
x = np.frombuffer(data_in48k, dtype=np.int16)
x = self.resampler.resample48_to_8(x)
self.datac0_buffer.push(x)
self.datac1_buffer.push(x)
self.datac3_buffer.push(x)
# refill fft_data buffer so we can plot a fft
if len(self.fft_data) < 1024:
self.fft_data += bytes(x)
while self.datac0_buffer.nbuffer >= self.datac0_nin:
# demodulate audio
nbytes = codec2.api.freedv_rawdatarx(self.datac0_freedv, self.datac0_bytes_out, self.datac0_buffer.buffer.ctypes)
self.datac0_buffer.pop(self.datac0_nin)
self.datac0_nin = codec2.api.freedv_nin(self.datac0_freedv)
if nbytes == self.datac0_bytes_per_frame:
self.dataqueue.put([self.datac0_bytes_out, self.datac0_freedv ,self.datac0_bytes_per_frame])
self.get_scatter(self.datac0_freedv)
self.calculate_snr(self.datac0_freedv)
while self.datac1_buffer.nbuffer >= self.datac1_nin:
# demodulate audio
nbytes = codec2.api.freedv_rawdatarx(self.datac1_freedv, self.datac1_bytes_out, self.datac1_buffer.buffer.ctypes)
self.datac1_buffer.pop(self.datac1_nin)
self.datac1_nin = codec2.api.freedv_nin(self.datac1_freedv)
if nbytes == self.datac1_bytes_per_frame:
self.dataqueue.put([self.datac1_bytes_out, self.datac1_freedv ,self.datac1_bytes_per_frame])
self.get_scatter(self.datac1_freedv)
self.calculate_snr(self.datac1_freedv)
while self.datac3_buffer.nbuffer >= self.datac3_nin:
# demodulate audio
nbytes = codec2.api.freedv_rawdatarx(self.datac3_freedv, self.datac3_bytes_out, self.datac3_buffer.buffer.ctypes)
self.datac3_buffer.pop(self.datac3_nin)
self.datac3_nin = codec2.api.freedv_nin(self.datac3_freedv)
if nbytes == self.datac3_bytes_per_frame:
self.dataqueue.put([self.datac3_bytes_out, self.datac3_freedv ,self.datac3_bytes_per_frame])
self.get_scatter(self.datac3_freedv)
self.calculate_snr(self.datac3_freedv)
return (None, pyaudio.paContinue)
def ptt_and_wait(self, state):
static.PTT_STATE = state
if state:
self.my_rig.set_ptt(self.hamlib_ptt_type, 1)
# rigctld.ptt_enable()
ptt_toggle_timeout = time.time() + 0.5
while time.time() < ptt_toggle_timeout:
pass
else:
ptt_toggle_timeout = time.time() + 0.5
while time.time() < ptt_toggle_timeout:
pass
self.my_rig.set_ptt(self.hamlib_ptt_type, 0)
# rigctld.ptt_disable()
return False
# --------------------------------------------------------------------------------------------------------
def transmit_signalling(self, data_out, count):
state_before_transmit = static.CHANNEL_STATE
static.CHANNEL_STATE = 'SENDING_SIGNALLING'
freedv_signalling_mode = 14
freedv = cast(self.c_lib.freedv_open(freedv_signalling_mode), c_void_p)
self.c_lib.freedv_set_clip(freedv, 1)
self.c_lib.freedv_set_tx_bpf(freedv, 1)
bytes_per_frame = int(self.c_lib.freedv_get_bits_per_modem_frame(freedv) / 8)
payload_per_frame = bytes_per_frame - 2
n_nom_modem_samples = self.c_lib.freedv_get_n_nom_modem_samples(freedv)
n_tx_modem_samples = self.c_lib.freedv_get_n_tx_modem_samples(freedv)
n_tx_preamble_modem_samples = self.c_lib.freedv_get_n_tx_preamble_modem_samples(freedv)
n_tx_postamble_modem_samples = self.c_lib.freedv_get_n_tx_postamble_modem_samples(freedv)
mod_out = create_string_buffer(n_tx_modem_samples * 2)
mod_out_preamble = create_string_buffer(n_tx_preamble_modem_samples * 2)
mod_out_postamble = create_string_buffer(n_tx_postamble_modem_samples * 2)
buffer = bytearray(payload_per_frame)
# set buffersize to length of data which will be send
buffer[:len(data_out)] = data_out
crc = ctypes.c_ushort(self.c_lib.freedv_gen_crc16(bytes(buffer), payload_per_frame)) # generate CRC16
# convert crc to 2 byte hex string
crc = crc.value.to_bytes(2, byteorder='big')
buffer += crc # append crc16 to buffer
data = (ctypes.c_ubyte * bytes_per_frame).from_buffer_copy(buffer)
# modulate DATA and safe it into mod_out pointer
self.c_lib.freedv_rawdatapreambletx(freedv, mod_out_preamble)
self.c_lib.freedv_rawdatatx(freedv, mod_out, data)
self.c_lib.freedv_rawdatapostambletx(freedv, mod_out_postamble)
self.streambuffer = bytearray()
self.streambuffer += bytes(mod_out_preamble)
self.streambuffer += bytes(mod_out)
self.streambuffer += bytes(mod_out_postamble)
converted_audio = audioop.ratecv(self.streambuffer, 2, 1, self.MODEM_SAMPLE_RATE, self.AUDIO_SAMPLE_RATE_TX, None)
self.streambuffer = bytes(converted_audio[0])
# append frame again with as much as in count defined
for i in range(1, count):
self.streambuffer += bytes(converted_audio[0])
while self.ptt_and_wait(True):
pass
# set channel state
static.CHANNEL_STATE = 'SENDING_SIGNALLING'
# start writing audio data to audio stream
self.stream_tx.write(self.streambuffer)
# set ptt back to false
self.ptt_and_wait(False)
# we have a problem with the receiving state
if state_before_transmit != 'RECEIVING_DATA':
static.CHANNEL_STATE = 'RECEIVING_SIGNALLING'
else:
static.CHANNEL_STATE = state_before_transmit
self.c_lib.freedv_close(freedv)
return True
# --------------------------------------------------------------------------------------------------------
# GET ARQ BURST FRAME VOM BUFFER AND MODULATE IT
def transmit_arq_burst(self, mode, frames):
# we could place this timing part inside the modem...
# lets see if this is a good idea..
# we need to update our timeout timestamp
state_before_transmit = static.CHANNEL_STATE
static.CHANNEL_STATE = 'SENDING_DATA'
freedv = cast(self.c_lib.freedv_open(mode), c_void_p)
self.c_lib.freedv_set_clip(freedv, 1)
self.c_lib.freedv_set_tx_bpf(freedv, 1)
bytes_per_frame = int(self.c_lib.freedv_get_bits_per_modem_frame(freedv) / 8)
payload_per_frame = bytes_per_frame - 2
n_nom_modem_samples = self.c_lib.freedv_get_n_nom_modem_samples(freedv)
n_tx_modem_samples = self.c_lib.freedv_get_n_tx_modem_samples(freedv)
n_tx_preamble_modem_samples = self.c_lib.freedv_get_n_tx_preamble_modem_samples(freedv)
n_tx_postamble_modem_samples = self.c_lib.freedv_get_n_tx_postamble_modem_samples(freedv)
mod_out = create_string_buffer(n_tx_modem_samples * 2)
mod_out_preamble = create_string_buffer(n_tx_preamble_modem_samples * 2)
mod_out_postamble = create_string_buffer(n_tx_postamble_modem_samples * 2)
self.streambuffer = bytearray()
self.c_lib.freedv_rawdatapreambletx(freedv, mod_out_preamble)
self.streambuffer += bytes(mod_out_preamble)
# loop through list of frames per burst
for n in range(0, len(frames)):
# create TX buffer
buffer = bytearray(payload_per_frame)
# set buffersize to length of data which will be send
buffer[:len(frames[n])] = frames[n]
crc = ctypes.c_ushort(self.c_lib.freedv_gen_crc16(bytes(buffer), payload_per_frame)) # generate CRC16
# convert crc to 2 byte hex string
crc = crc.value.to_bytes(2, byteorder='big')
buffer += crc # append crc16 to buffer
data = (ctypes.c_ubyte * bytes_per_frame).from_buffer_copy(buffer)
# modulate DATA and safe it into mod_out pointer
self.c_lib.freedv_rawdatatx(freedv, mod_out, data)
self.streambuffer += bytes(mod_out)
self.c_lib.freedv_rawdatapostambletx(freedv, mod_out_postamble)
self.streambuffer += bytes(mod_out_postamble)
converted_audio = audioop.ratecv(self.streambuffer, 2, 1, self.MODEM_SAMPLE_RATE, self.AUDIO_SAMPLE_RATE_TX, None)
self.streambuffer = bytes(converted_audio[0])
# -------------- transmit audio
while self.ptt_and_wait(True):
pass
# set channel state
static.CHANNEL_STATE = 'SENDING_DATA'
# write audio to stream
self.stream_tx.write(self.streambuffer)
static.CHANNEL_STATE = 'RECEIVING_SIGNALLING'
self.ptt_and_wait(False)
# close codec2 instance
self.c_lib.freedv_close(freedv)
return True
# --------------------------------------------------------------------------------------------------------
def receive(self):
try:
print(f"starting pyaudio callback", file=sys.stderr)
self.stream_rx.start_stream()
except Exception as e:
print(f"pyAudio error: {e}", file=sys.stderr)
while self.stream_rx.is_active():
time.sleep(1)
# worker for FIFO queue for processing received frames
def worker(self):
while True:
time.sleep(0.01)
data = self.dataqueue.get()
self.process_data(data[0], data[1], data[2])
self.dataqueue.task_done()
# forward data only if broadcast or we are the receiver
# bytes_out[1:2] == callsign check for signalling frames,
# bytes_out[6:7] == callsign check for data frames,
# bytes_out[1:2] == b'\x01' --> broadcasts like CQ with n frames per_burst = 1
# we could also create an own function, which returns True.
def process_data(self, bytes_out, freedv, bytes_per_frame):
if bytes(bytes_out[1:2]) == static.MYCALLSIGN_CRC8 or bytes(bytes_out[6:7]) == static.MYCALLSIGN_CRC8 or bytes(bytes_out[1:2]) == b'\x01':
# CHECK IF FRAMETYPE IS BETWEEN 10 and 50 ------------------------
frametype = int.from_bytes(bytes(bytes_out[:1]), "big")
frame = frametype - 10
n_frames_per_burst = int.from_bytes(bytes(bytes_out[1:2]), "big")
self.c_lib.freedv_set_frames_per_burst(freedv, n_frames_per_burst);
#frequency_offset = self.get_frequency_offset(freedv)
#print("Freq-Offset: " + str(frequency_offset))
if 50 >= frametype >= 10:
# force = True, if we don't simulate a loss of the third frame, else force = False
force = True
if frame != 3 or force:
# send payload data to arq checker without CRC16
data_handler.arq_data_received(bytes(bytes_out[:-2]), bytes_per_frame)
#print("static.ARQ_RX_BURST_BUFFER.count(None) " + str(static.ARQ_RX_BURST_BUFFER.count(None)))
if static.RX_BURST_BUFFER.count(None) <= 1:
logging.debug("FULL BURST BUFFER ---> UNSYNC")
self.c_lib.freedv_set_sync(freedv, 0)
else:
logging.critical("-------------SIMULATED MISSING FRAME")
force = True
# BURST ACK
elif frametype == 60:
logging.debug("ACK RECEIVED....")
data_handler.burst_ack_received()
# FRAME ACK
elif frametype == 61:
logging.debug("FRAME ACK RECEIVED....")
data_handler.frame_ack_received()
# FRAME RPT
elif frametype == 62:
logging.debug("REPEAT REQUEST RECEIVED....")
data_handler.burst_rpt_received(bytes_out[:-2])
# CQ FRAME
elif frametype == 200:
logging.debug("CQ RECEIVED....")
data_handler.received_cq(bytes_out[:-2])
# PING FRAME
elif frametype == 210:
logging.debug("PING RECEIVED....")
frequency_offset = self.get_frequency_offset(freedv)
#print("Freq-Offset: " + str(frequency_offset))
data_handler.received_ping(bytes_out[:-2], frequency_offset)
# PING ACK
elif frametype == 211:
logging.debug("PING ACK RECEIVED....")
# early detection of frequency offset
#frequency_offset = int.from_bytes(bytes(bytes_out[9:11]), "big", signed=True)
#print("Freq-Offset: " + str(frequency_offset))
#current_frequency = self.my_rig.get_freq()
#corrected_frequency = current_frequency + frequency_offset
# temporarely disabled this feature, beacuse it may cause some confusion.
# we also have problems if we are operating at band bordes like 7.000Mhz
# If we get a corrected frequency less 7.000 Mhz, Ham Radio devices will not transmit...
#self.my_rig.set_vfo(Hamlib.RIG_VFO_A)
#self.my_rig.set_freq(Hamlib.RIG_VFO_A, corrected_frequency)
data_handler.received_ping_ack(bytes_out[:-2])
# ARQ FILE TRANSFER RECEIVED!
elif frametype == 225:
logging.debug("ARQ arq_received_data_channel_opener RECEIVED")
data_handler.arq_received_data_channel_opener(bytes_out[:-2])
# ARQ CHANNEL IS OPENED
elif frametype == 226:
logging.debug("ARQ arq_received_channel_is_open RECEIVED")
data_handler.arq_received_channel_is_open(bytes_out[:-2])
# ARQ CONNECT ACK / KEEP ALIVE
elif frametype == 230:
logging.debug("BEACON RECEIVED")
data_handler.received_beacon(bytes_out[:-2])
elif frametype == 255:
structlog.get_logger("structlog").debug("TESTFRAME RECEIVED", frame=bytes_out[:])
else:
structlog.get_logger("structlog").warning("[TNC] ARQ - other frame type", frametype=frametype)
# DO UNSYNC AFTER LAST BURST by checking the frame nums against the total frames per burst
if frame == n_frames_per_burst:
logging.info("LAST FRAME ---> UNSYNC")
self.c_lib.freedv_set_sync(freedv, 0) # FORCE UNSYNC
else:
# for debugging purposes to receive all data
structlog.get_logger("structlog").debug("[TNC] Unknown frame received", frame=bytes_out[:-2])
def get_frequency_offset(self, freedv):
modemStats = MODEMSTATS()
self.c_lib.freedv_get_modem_extended_stats.restype = None
self.c_lib.freedv_get_modem_extended_stats(freedv, ctypes.byref(modemStats))
offset = round(modemStats.foff) * (-1)
static.FREQ_OFFSET = offset
return offset
def get_scatter(self, freedv):
modemStats = MODEMSTATS()
self.c_lib.freedv_get_modem_extended_stats.restype = None
self.c_lib.freedv_get_modem_extended_stats(freedv, ctypes.byref(modemStats))
scatterdata = []
scatterdata_small = []
for i in range(MODEM_STATS_NC_MAX):
for j in range(MODEM_STATS_NR_MAX):
# check if odd or not to get every 2nd item for x
if (j % 2) == 0:
xsymbols = round(modemStats.rx_symbols[i][j]/1000)
ysymbols = round(modemStats.rx_symbols[i][j+1]/1000)
# check if value 0.0 or has real data
if xsymbols != 0.0 and ysymbols != 0.0:
scatterdata.append({"x": xsymbols, "y": ysymbols})
# only append scatter data if new data arrived
if 150 > len(scatterdata) > 0:
static.SCATTER = scatterdata
else:
# only take every tenth data point
scatterdata_small = scatterdata[::10]
static.SCATTER = scatterdata_small
def calculate_ber(self, freedv):
Tbits = self.c_lib.freedv_get_total_bits(freedv)
Terrs = self.c_lib.freedv_get_total_bit_errors(freedv)
if Tbits != 0:
ber = (Terrs / Tbits) * 100
static.BER = int(ber)
self.c_lib.freedv_set_total_bit_errors(freedv, 0)
self.c_lib.freedv_set_total_bits(freedv, 0)
def calculate_snr(self, freedv):
modem_stats_snr = c_float()
modem_stats_sync = c_int()
self.c_lib.freedv_get_modem_stats(freedv, byref(
modem_stats_sync), byref(modem_stats_snr))
modem_stats_snr = modem_stats_snr.value
try:
static.SNR = round(modem_stats_snr, 1)
except:
static.SNR = 0
def get_radio_stats(self):
while True:
time.sleep(0.1)
static.HAMLIB_FREQUENCY = int(self.my_rig.get_freq())
(hamlib_mode, static.HAMLIB_BANDWITH) = self.my_rig.get_mode()
static.HAMLIB_MODE = Hamlib.rig_strrmode(hamlib_mode)
def calculate_fft(self):
while True:
time.sleep(0.01)
# WE NEED TO OPTIMIZE THIS!
if len(self.fft_data) >= 1024:
data_in = self.fft_data
self.fft_data = bytes()
# https://gist.github.com/ZWMiller/53232427efc5088007cab6feee7c6e4c
audio_data = np.fromstring(data_in, np.int16)
# Fast Fourier Transform, 10*log10(abs) is to scale it to dB
# and make sure it's not imaginary
try:
fftarray = np.fft.rfft(audio_data)
# set value 0 to 1 to avoid division by zero
fftarray[fftarray == 0] = 1
dfft = 10.*np.log10(abs(fftarray))
# round data to decrease size
dfft = np.around(dfft, 1)
dfftlist = dfft.tolist()
static.FFT = dfftlist[10:180] #200 --> bandwith 3000
except:
structlog.get_logger("structlog").debug("[TNC] Setting fft=0")
# else 0
static.FFT = [0] * 400
else:
pass
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