#!/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 print(static.HAMLIB_USE_RIGCTL) if static.HAMLIB_USE_RIGCTL: structlog.get_logger("structlog").warning("using rigctl....") import rigctl as rig else: structlog.get_logger("structlog").warning("using rig.......") 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(0.05) 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(0.05) txbuffer += bytes(mod_out) # append postamble to txbuffer codec2.api.freedv_rawdatapostambletx(freedv, mod_out_postamble) txbuffer += bytes(mod_out_postamble) time.sleep(0.05) # 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(0.05) # 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: delta = bytes(self.AUDIO_FRAMES_PER_BUFFER_RX*2 - len(c)) c += delta structlog.get_logger("structlog").debug("[TNC] mod out shorter than audio buffer", delta=len(delta)) 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: structlog.get_logger("structlog").debug("[TNC] starting pyaudio callback") self.audio_stream.start_stream() except Exception as e: structlog.get_logger("structlog").error("[TNC] starting pyaudio callback failed", e=e) 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() # data[0] = bytes_out # data[1] = freedv session # data[2] = bytes_per_frame 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: # get snr of received data snr = self.calculate_snr(freedv) structlog.get_logger("structlog").debug("[TNC] RX SNR", snr=snr) # send payload data to arq checker without CRC16 data_handler.arq_data_received(bytes(bytes_out[:-2]), bytes_per_frame, snr, freedv) # if we received the last frame of a burst or the last remaining rpt frame, do a modem unsync if static.RX_BURST_BUFFER.count(None) <= 1 or (frame+1) == n_frames_per_burst: structlog.get_logger("structlog").debug(f"LAST FRAME OF BURST --> UNSYNC {frame+1}/{n_frames_per_burst}") self.c_lib.freedv_set_sync(freedv, 0) # BURST ACK elif frametype == 60: structlog.get_logger("structlog").debug("ACK RECEIVED....") data_handler.burst_ack_received(bytes_out[:-2]) # FRAME ACK elif frametype == 61: structlog.get_logger("structlog").debug("FRAME ACK RECEIVED....") data_handler.frame_ack_received() # FRAME RPT elif frametype == 62: structlog.get_logger("structlog").debug("REPEAT REQUEST RECEIVED....") data_handler.burst_rpt_received(bytes_out[:-2]) # FRAME NACK elif frametype == 63: structlog.get_logger("structlog").debug("FRAME NOT ACK RECEIVED....") data_handler.frame_nack_received(bytes_out[:-2]) # CQ FRAME elif frametype == 200: structlog.get_logger("structlog").debug("CQ RECEIVED....") data_handler.received_cq(bytes_out[:-2]) # PING FRAME elif frametype == 210: structlog.get_logger("structlog").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: structlog.get_logger("structlog").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: structlog.get_logger("structlog").debug("ARQ arq_received_data_channel_opener") data_handler.arq_received_data_channel_opener(bytes_out[:-2]) # ARQ CHANNEL IS OPENED elif frametype == 226: structlog.get_logger("structlog").debug("ARQ arq_received_channel_is_open") data_handler.arq_received_channel_is_open(bytes_out[:-2]) # ARQ CONNECT ACK / KEEP ALIVE # this is outdated and we may remove it elif frametype == 230: structlog.get_logger("structlog").debug("BEACON RECEIVED") data_handler.received_beacon(bytes_out[:-2]) # TESTFRAMES 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 # this should be changed to a timeout based unsync 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) return static.SNR except: static.SNR = 0 return static.SNR 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) def set_frames_per_burst(self, n_frames_per_burst): codec2.api.freedv_set_frames_per_burst(self.datac1_freedv,n_frames_per_burst) codec2.api.freedv_set_frames_per_burst(self.datac3_freedv,n_frames_per_burst)