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FreeDATA/tnc/modem.py
dj2ls 1f6c53ecf0 improved rx stats
2022-01-03 00:27:05 +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 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
import rig
# 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()
######################################################
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 = 2400 #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()
# init FIFO queue to store modulation out in
self.modoutqueue = queue.Queue()
# define fft_data buffer
self.fft_data = bytes()
# 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_payload_per_frame = self.datac0_bytes_per_frame -2
self.datac0_n_nom_modem_samples = self.c_lib.freedv_get_n_nom_modem_samples(self.datac0_freedv)
self.datac0_n_tx_modem_samples = self.c_lib.freedv_get_n_tx_modem_samples(self.datac0_freedv)
self.datac0_n_tx_preamble_modem_samples = self.c_lib.freedv_get_n_tx_preamble_modem_samples(self.datac0_freedv)
self.datac0_n_tx_postamble_modem_samples = self.c_lib.freedv_get_n_tx_postamble_modem_samples(self.datac0_freedv)
self.datac0_bytes_out = create_string_buffer(self.datac0_bytes_per_frame)
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)
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)
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:
static.AUDIO_INPUT_DEVICE = loopback_list[0] #0 = RX
static.AUDIO_OUTPUT_DEVICE = loopback_list[1] #1 = TX
print(f"loopback_list rx: {loopback_list}", file=sys.stderr)
self.audio_stream = 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=True,
input_device_index=static.AUDIO_INPUT_DEVICE,
output_device_index=static.AUDIO_OUTPUT_DEVICE,
stream_callback=self.audio_callback
)
# --------------------------------------------INIT AND OPEN HAMLIB
self.hamlib = rig.radio()
self.hamlib.open_rig(devicename=static.HAMLIB_DEVICE_NAME, deviceport=static.HAMLIB_DEVICE_PORT, hamlib_ptt_type=static.HAMLIB_PTT_TYPE, serialspeed=static.HAMLIB_SERIAL_SPEED, pttport=static.HAMLIB_PTT_PORT, data_bits=static.HAMLIB_DATA_BITS, stop_bits=static.HAMLIB_STOP_BITS, handshake=static.HAMLIB_HANDSHAKE)
# --------------------------------------------START DECODER THREAD
FFT_THREAD = threading.Thread(target=self.calculate_fft, name="FFT_THREAD")
FFT_THREAD.start()
AUDIO_THREAD = threading.Thread(target=self.audio, name="AUDIO_THREAD")
AUDIO_THREAD.start()
HAMLIB_THREAD = threading.Thread(target=self.update_rig_data, name="HAMLIB_THREAD")
HAMLIB_THREAD.start()
WORKER_THREAD = threading.Thread(target=self.worker, name="WORKER_THREAD")
WORKER_THREAD.start()
# --------------------------------------------------------------------------------------------------------
def audio_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)
self.fft_data += bytes(x)
if self.modoutqueue.empty():
data_out48k = bytes(self.AUDIO_FRAMES_PER_BUFFER_TX*2*2)
else:
data_out48k = self.modoutqueue.get()
return (data_out48k, pyaudio.paContinue)
# --------------------------------------------------------------------------------------------------------
def transmit(self, mode, repeats, repeat_delay, frames):
# open codec2 instance
#self.MODE = codec2.freedv_get_mode_value_by_name(mode)
self.MODE = mode
freedv = cast(codec2.api.freedv_open(self.MODE), c_void_p)
# get number of bytes per frame for mode
bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(freedv)/8)
payload_bytes_per_frame = bytes_per_frame -2
# init buffer for data
n_tx_modem_samples = codec2.api.freedv_get_n_tx_modem_samples(freedv)
mod_out = create_string_buffer(n_tx_modem_samples * 2)
# init buffer for preample
n_tx_preamble_modem_samples = codec2.api.freedv_get_n_tx_preamble_modem_samples(freedv)
mod_out_preamble = create_string_buffer(n_tx_preamble_modem_samples * 2)
# init buffer for postamble
n_tx_postamble_modem_samples = codec2.api.freedv_get_n_tx_postamble_modem_samples(freedv)
mod_out_postamble = create_string_buffer(n_tx_postamble_modem_samples * 2)
# add empty data to handle ptt toggle time
data_delay_mseconds = 0 #miliseconds
data_delay = int(self.MODEM_SAMPLE_RATE*(data_delay_mseconds/1000))
mod_out_silence = create_string_buffer(data_delay*2)
txbuffer = bytes(mod_out_silence)
for i in range(1,repeats+1):
# write preamble to txbuffer
codec2.api.freedv_rawdatapreambletx(freedv, mod_out_preamble)
time.sleep(0.001)
txbuffer += bytes(mod_out_preamble)
# create modulaton for n frames in list
for n in range(0,len(frames)):
# create buffer for data
buffer = bytearray(payload_bytes_per_frame) # use this if CRC16 checksum is required ( DATA1-3)
buffer[:len(frames[n])] = frames[n] # set buffersize to length of data which will be send
# create crc for data frame - we are using the crc function shipped with codec2 to avoid
# crc algorithm incompatibilities
crc = ctypes.c_ushort(codec2.api.freedv_gen_crc16(bytes(buffer), payload_bytes_per_frame)) # generate CRC16
crc = crc.value.to_bytes(2, byteorder='big') # convert crc to 2 byte hex string
buffer += crc # append crc16 to buffer
data = (ctypes.c_ubyte * bytes_per_frame).from_buffer_copy(buffer)
codec2.api.freedv_rawdatatx(freedv,mod_out,data) # modulate DATA and save it into mod_out pointer
time.sleep(0.001)
txbuffer += bytes(mod_out)
print(len(txbuffer))
# append postamble to txbuffer
codec2.api.freedv_rawdatapostambletx(freedv, mod_out_postamble)
txbuffer += bytes(mod_out_postamble)
time.sleep(0.001)
# add delay to end of frames
samples_delay = int(self.MODEM_SAMPLE_RATE*(repeat_delay/1000))
mod_out_silence = create_string_buffer(samples_delay*2)
txbuffer += bytes(mod_out_silence)
time.sleep(0.001)
# resample up to 48k (resampler works on np.int16)
x = np.frombuffer(txbuffer, dtype=np.int16)
txbuffer_48k = self.resampler.resample8_to_48(x)
# split modualted audio to chunks
#https://newbedev.com/how-to-split-a-byte-string-into-separate-bytes-in-python
txbuffer_48k = bytes(txbuffer_48k)
chunk = [txbuffer_48k[i:i+self.AUDIO_FRAMES_PER_BUFFER_RX*2] for i in range(0, len(txbuffer_48k), self.AUDIO_FRAMES_PER_BUFFER_RX*2)]
# add modulated chunks to fifo buffer
for c in chunk:
# if data is shorter than the expcected audio frames per buffer we need to append 0
# to prevent the callback from stucking/crashing
if len(c) < self.AUDIO_FRAMES_PER_BUFFER_RX*2:
c += bytes(self.AUDIO_FRAMES_PER_BUFFER_RX*2 - len(c))
self.modoutqueue.put(c)
# maybe we need to toggle PTT before craeting modulation because of queue processing
static.PTT_STATE = self.hamlib.set_ptt(True)
while not self.modoutqueue.empty():
pass
static.PTT_STATE = self.hamlib.set_ptt(False)
self.c_lib.freedv_close(freedv)
return True
def audio(self):
try:
print(f"starting pyaudio callback", file=sys.stderr)
self.audio_stream.start_stream()
except Exception as e:
print(f"pyAudio error: {e}", file=sys.stderr)
while self.audio_stream.is_active():
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)
# 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[3:4]) == 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:
# 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)
# 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")
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")
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_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 update_rig_data(self):
while True:
time.sleep(0.5)
#(static.HAMLIB_FREQUENCY, static.HAMLIB_MODE, static.HAMLIB_BANDWITH, static.PTT_STATE) = self.hamlib.get_rig_data()
static.HAMLIB_FREQUENCY = self.hamlib.get_frequency()
static.HAMLIB_MODE = self.hamlib.get_mode()
static.HAMLIB_BANDWITH = self.hamlib.get_bandwith()
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[0:320] #200 --> bandwith 3000
except:
structlog.get_logger("structlog").debug("[TNC] Setting fft=0")
# else 0
static.FFT = [0] * 320
else:
pass
def get_bytes_per_frame(self, mode):
freedv = cast(codec2.api.freedv_open(mode), c_void_p)
# get number of bytes per frame for mode
return int(codec2.api.freedv_get_bits_per_modem_frame(freedv)/8)
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