#!/usr/bin/env python3 # -*- coding: utf-8 -*- """ Created on Wed Dec 23 07:04:24 2020 @author: DJ2LS """ # pylint: disable=invalid-name, line-too-long, c-extension-no-member # pylint: disable=import-outside-toplevel import atexit import ctypes import os import queue import sys import threading import time from collections import deque import codec2 import data_handler import numpy as np import sock import sounddevice as sd import static import structlog import ujson as json TESTMODE = False RXCHANNEL = "" TXCHANNEL = "" # Initialize FIFO queue to store received frames MODEM_RECEIVED_QUEUE = queue.Queue() MODEM_TRANSMIT_QUEUE = queue.Queue() static.TRANSMITTING = False # Receive only specific modes to reduce CPU load RECEIVE_DATAC1 = False RECEIVE_DATAC3 = False RECEIVE_FSK_LDPC_1 = False class RF: """Class to encapsulate interactions between the audio device and codec2""" log = structlog.get_logger("RF") def __init__(self) -> None: """ """ self.sampler_avg = 0 self.buffer_avg = 0 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 # 8192 Let's do some tests with very small chunks for TX self.AUDIO_FRAMES_PER_BUFFER_TX = 2400 * 2 # 8 * (self.AUDIO_SAMPLE_RATE_RX/self.MODEM_SAMPLE_RATE) == 48 self.AUDIO_CHANNELS = 1 self.MODE = 0 # Locking state for mod out so buffer will be filled before we can use it # https://github.com/DJ2LS/FreeDATA/issues/127 # https://github.com/DJ2LS/FreeDATA/issues/99 self.mod_out_locked = True # Make sure our resampler will work assert (self.AUDIO_SAMPLE_RATE_RX / self.MODEM_SAMPLE_RATE) == codec2.api.FDMDV_OS_48 # type: ignore # Small hack for initializing codec2 via codec2.py module # TODO: Need to change the entire modem module to integrate codec2 module self.c_lib = codec2.api self.resampler = codec2.resampler() self.modem_transmit_queue = MODEM_TRANSMIT_QUEUE self.modem_received_queue = MODEM_RECEIVED_QUEUE # Init FIFO queue to store modulation out in self.modoutqueue = deque() # Define fft_data buffer self.fft_data = bytes() # Open codec2 instances # Datac0 - control frames self.datac0_freedv = ctypes.cast( codec2.api.freedv_open(codec2.api.FREEDV_MODE_DATAC0), ctypes.c_void_p ) self.c_lib.freedv_set_tuning_range( self.datac0_freedv, ctypes.c_float(static.TUNING_RANGE_FMIN), ctypes.c_float(static.TUNING_RANGE_FMAX), ) self.datac0_bytes_per_frame = int( codec2.api.freedv_get_bits_per_modem_frame(self.datac0_freedv) / 8 ) self.datac0_bytes_out = ctypes.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) # Additional Datac0-specific information - these are not referenced anywhere else. # 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) # ) # Datac1 - higher-bandwidth data frames self.datac1_freedv = ctypes.cast( codec2.api.freedv_open(codec2.api.FREEDV_MODE_DATAC1), ctypes.c_void_p ) self.c_lib.freedv_set_tuning_range( self.datac1_freedv, ctypes.c_float(static.TUNING_RANGE_FMIN), ctypes.c_float(static.TUNING_RANGE_FMAX), ) self.datac1_bytes_per_frame = int( codec2.api.freedv_get_bits_per_modem_frame(self.datac1_freedv) / 8 ) self.datac1_bytes_out = ctypes.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) # Datac3 - lower-bandwidth data frames self.datac3_freedv = ctypes.cast( codec2.api.freedv_open(codec2.api.FREEDV_MODE_DATAC3), ctypes.c_void_p ) self.c_lib.freedv_set_tuning_range( self.datac3_freedv, ctypes.c_float(static.TUNING_RANGE_FMIN), ctypes.c_float(static.TUNING_RANGE_FMAX), ) self.datac3_bytes_per_frame = int( codec2.api.freedv_get_bits_per_modem_frame(self.datac3_freedv) / 8 ) self.datac3_bytes_out = ctypes.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) # FSK Long-distance Parity Code 0 - data frames self.fsk_ldpc_freedv_0 = ctypes.cast( codec2.api.freedv_open_advanced( codec2.api.FREEDV_MODE_FSK_LDPC, ctypes.byref(codec2.api.FREEDV_MODE_FSK_LDPC_0_ADV), ), ctypes.c_void_p, ) self.fsk_ldpc_bytes_per_frame_0 = int( codec2.api.freedv_get_bits_per_modem_frame(self.fsk_ldpc_freedv_0) / 8 ) self.fsk_ldpc_bytes_out_0 = ctypes.create_string_buffer( self.fsk_ldpc_bytes_per_frame_0 ) # codec2.api.freedv_set_frames_per_burst(self.fsk_ldpc_freedv_0, 1) self.fsk_ldpc_buffer_0 = codec2.audio_buffer(self.AUDIO_FRAMES_PER_BUFFER_RX) # FSK Long-distance Parity Code 1 - data frames self.fsk_ldpc_freedv_1 = ctypes.cast( codec2.api.freedv_open_advanced( codec2.api.FREEDV_MODE_FSK_LDPC, ctypes.byref(codec2.api.FREEDV_MODE_FSK_LDPC_1_ADV), ), ctypes.c_void_p, ) self.fsk_ldpc_bytes_per_frame_1 = int( codec2.api.freedv_get_bits_per_modem_frame(self.fsk_ldpc_freedv_1) / 8 ) self.fsk_ldpc_bytes_out_1 = ctypes.create_string_buffer( self.fsk_ldpc_bytes_per_frame_1 ) # codec2.api.freedv_set_frames_per_burst(self.fsk_ldpc_freedv_0, 1) self.fsk_ldpc_buffer_1 = codec2.audio_buffer(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) self.fsk_ldpc_nin_0 = codec2.api.freedv_nin(self.fsk_ldpc_freedv_0) self.fsk_ldpc_nin_1 = codec2.api.freedv_nin(self.fsk_ldpc_freedv_1) # self.log.debug("[MDM] RF: ",datac0_nin=self.datac0_nin) # --------------------------------------------CREATE PYAUDIO INSTANCE if not TESTMODE: try: self.stream = sd.RawStream( channels=1, dtype="int16", callback=self.callback, device=(static.AUDIO_INPUT_DEVICE, static.AUDIO_OUTPUT_DEVICE), samplerate=self.AUDIO_SAMPLE_RATE_RX, blocksize=4800, ) atexit.register(self.stream.stop) self.log.info("[MDM] init: opened audio devices") except Exception as err: self.log.error("[MDM] init: can't open audio device. Exit", e=err) sys.exit(1) try: self.log.debug("[MDM] init: starting pyaudio callback") # self.audio_stream.start_stream() self.stream.start() except Exception as err: self.log.error("[MDM] init: starting pyaudio callback failed", e=err) else: class Object: """An object for simulating audio stream""" active = True self.stream = Object() # Create mkfifo buffers try: os.mkfifo(RXCHANNEL) os.mkfifo(TXCHANNEL) except Exception as err: self.log.info(f"[MDM] init:mkfifo: Exception: {err}") mkfifo_write_callback_thread = threading.Thread( target=self.mkfifo_write_callback, name="MKFIFO WRITE CALLBACK THREAD", daemon=True, ) mkfifo_write_callback_thread.start() self.log.debug("[MDM] Starting mkfifo_read_callback") mkfifo_read_callback_thread = threading.Thread( target=self.mkfifo_read_callback, name="MKFIFO READ CALLBACK THREAD", daemon=True, ) mkfifo_read_callback_thread.start() mkfifo_read_callback_thread = threading.Thread(target=self.mkfifo_read_callback, name="MKFIFO READ CALLBACK THREAD",daemon=True) mkfifo_read_callback_thread.start() # --------------------------------------------INIT AND OPEN HAMLIB # Check how we want to control the radio if static.HAMLIB_RADIOCONTROL == "direct": import rig elif static.HAMLIB_RADIOCONTROL == "rigctl": import rigctl as rig elif static.HAMLIB_RADIOCONTROL == "rigctld": import rigctld as rig else: import rigdummy as rig 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, rigctld_ip=static.HAMLIB_RIGCTLD_IP, rigctld_port=static.HAMLIB_RIGCTLD_PORT, ) # --------------------------------------------START DECODER THREAD if static.ENABLE_FFT: fft_thread = threading.Thread( target=self.calculate_fft, name="FFT_THREAD", daemon=True ) fft_thread.start() audio_thread_datac0 = threading.Thread( target=self.audio_datac0, name="AUDIO_THREAD DATAC0", daemon=True ) audio_thread_datac0.start() audio_thread_datac1 = threading.Thread( target=self.audio_datac1, name="AUDIO_THREAD DATAC1", daemon=True ) audio_thread_datac1.start() audio_thread_datac3 = threading.Thread( target=self.audio_datac3, name="AUDIO_THREAD DATAC3", daemon=True ) audio_thread_datac3.start() if static.ENABLE_FSK: audio_thread_fsk_ldpc0 = threading.Thread( target=self.audio_fsk_ldpc_0, name="AUDIO_THREAD FSK LDPC0", daemon=True ) audio_thread_fsk_ldpc0.start() audio_thread_fsk_ldpc1 = threading.Thread( target=self.audio_fsk_ldpc_1, name="AUDIO_THREAD FSK LDPC1", daemon=True ) audio_thread_fsk_ldpc1.start() hamlib_thread = threading.Thread( target=self.update_rig_data, name="HAMLIB_THREAD", daemon=True ) hamlib_thread.start() # self.log.debug("[MDM] Starting worker_receive") worker_received = threading.Thread( target=self.worker_received, name="WORKER_THREAD", daemon=True ) worker_received.start() worker_transmit = threading.Thread( target=self.worker_transmit, name="WORKER_THREAD", daemon=True ) worker_transmit.start() # -------------------------------------------------------------------------------------------------------- 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: time.sleep(0.01) # -----read data_in48k = bytes() with open(RXCHANNEL, "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.datac0_buffer, True), (self.datac1_buffer, RECEIVE_DATAC1), (self.datac3_buffer, RECEIVE_DATAC3), # Not enabled yet. # (self.fsk_ldpc_buffer_0, static.ENABLE_FSK), # (self.fsk_ldpc_buffer_1, static.ENABLE_FSK), ]: if ( not data_buffer.nbuffer + length_x > data_buffer.size and receive ): data_buffer.push(x) def mkfifo_write_callback(self) -> None: """Support testing by writing the audio data to a pipe.""" while True: time.sleep(0.01) # -----write if len(self.modoutqueue) <= 0 or self.mod_out_locked: # data_out48k = np.zeros(self.AUDIO_FRAMES_PER_BUFFER_RX, dtype=np.int16) pass else: data_out48k = self.modoutqueue.popleft() # print(len(data_out48k)) with open(TXCHANNEL, "wb") as fifo_write: fifo_write.write(data_out48k) fifo_write.flush() fifo_write.flush() # -------------------------------------------------------------------- def callback(self, data_in48k, outdata, frames, time, status) -> None: """ Receive data into appropriate queue. Args: data_in48k: Incoming data received outdata: Container for the data returned frames: Number of frames time: status: """ # self.log.debug("[MDM] callback") x = np.frombuffer(data_in48k, dtype=np.int16) x = self.resampler.resample48_to_8(x) # Avoid decoding when transmitting to reduce CPU if not static.TRANSMITTING: length_x = len(x) # Avoid buffer overflow by filling only if buffer for # selected datachannel mode is not full for audiobuffer, receive, index in [ (self.datac0_buffer, True, 0), (self.datac1_buffer, RECEIVE_DATAC1, 1), (self.datac3_buffer, RECEIVE_DATAC3, 2), (self.fsk_ldpc_buffer_0, static.ENABLE_FSK, 3), (self.fsk_ldpc_buffer_1, static.ENABLE_FSK, 4), ]: if audiobuffer.nbuffer + length_x > audiobuffer.size: static.BUFFER_OVERFLOW_COUNTER[index] += 1 elif receive: audiobuffer.push(x) if len(self.modoutqueue) <= 0 or self.mod_out_locked: # if not self.modoutqueue or self.mod_out_locked: data_out48k = np.zeros(frames, dtype=np.int16) self.fft_data = x else: data_out48k = self.modoutqueue.popleft() self.fft_data = data_out48k try: outdata[:] = data_out48k[:frames] except IndexError as err: self.log.debug(f"[MDM] callback: IndexError: {err}") # return (data_out48k, audio.pyaudio.paContinue) # -------------------------------------------------------------------- def transmit( self, mode, repeats: int, repeat_delay: int, frames: bytearray ) -> None: """ Args: mode: repeats: repeat_delay: frames: """ self.log.debug("[MDM] transmit", mode=mode) static.TRANSMITTING = True # Toggle ptt early to save some time and send ptt state via socket static.PTT_STATE = self.hamlib.set_ptt(True) jsondata = {"ptt": "True"} data_out = json.dumps(jsondata) sock.SOCKET_QUEUE.put(data_out) # Open codec2 instance self.MODE = mode freedv = open_codec2_instance(self.MODE) # 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 = ctypes.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 = ctypes.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 = ctypes.create_string_buffer( n_tx_postamble_modem_samples * 2 ) # Add empty data to handle ptt toggle time data_delay_mseconds = 0 # milliseconds data_delay = int(self.MODEM_SAMPLE_RATE * (data_delay_mseconds / 1000)) # type: ignore mod_out_silence = ctypes.create_string_buffer(data_delay * 2) txbuffer = bytes(mod_out_silence) self.log.debug( "[MDM] TRANSMIT", mode=self.MODE, payload=payload_bytes_per_frame ) for _ in range(repeats): # codec2 fsk preamble may be broken - # at least it sounds like that, so we are disabling it for testing if self.MODE not in [ codec2.FREEDV_MODE.fsk_ldpc_0.value, codec2.FREEDV_MODE.fsk_ldpc_1.value, ]: # Write preamble to txbuffer codec2.api.freedv_rawdatapreambletx(freedv, mod_out_preamble) txbuffer += bytes(mod_out_preamble) # Create modulaton for all frames in the list for frame in frames: # Create buffer for data # Use this if CRC16 checksum is required (DATAc1-3) buffer = bytearray(payload_bytes_per_frame) # Set buffersize to length of data which will be send buffer[: len(frame)] = frame # type: ignore # Create crc for data frame - # Use the crc function shipped with codec2 # to avoid CRC algorithm incompatibilities # Generate CRC16 crc = ctypes.c_ushort( codec2.api.freedv_gen_crc16(bytes(buffer), payload_bytes_per_frame) ) # Convert crc to 2-byte (16-bit) hex string crc = crc.value.to_bytes(2, byteorder="big") # Append CRC to data buffer buffer += crc data = (ctypes.c_ubyte * bytes_per_frame).from_buffer_copy(buffer) # modulate DATA and save it into mod_out pointer codec2.api.freedv_rawdatatx(freedv, mod_out, data) txbuffer += bytes(mod_out) # codec2 fsk postamble may be broken - # at least it sounds like that, so we are disabling it for testing if self.MODE not in [ codec2.FREEDV_MODE.fsk_ldpc_0.value, codec2.FREEDV_MODE.fsk_ldpc_1.value, ]: # Write postamble to txbuffer codec2.api.freedv_rawdatapostambletx(freedv, mod_out_postamble) # Append postamble to txbuffer txbuffer += bytes(mod_out_postamble) # Add delay to end of frames samples_delay = int(self.MODEM_SAMPLE_RATE * (repeat_delay / 1000)) # type: ignore mod_out_silence = ctypes.create_string_buffer(samples_delay * 2) txbuffer += bytes(mod_out_silence) # Re-sample back up to 48k (resampler works on np.int16) x = np.frombuffer(txbuffer, dtype=np.int16) x = set_audio_volume(x, static.TX_AUDIO_LEVEL) txbuffer_48k = self.resampler.resample8_to_48(x) # Explicitly lock our usage of mod_out_queue if needed # Deactivated for testing purposes self.mod_out_locked = False # ------------------------------- chunk_length = self.AUDIO_FRAMES_PER_BUFFER_TX # 4800 chunk = [ txbuffer_48k[i : i + chunk_length] for i in range(0, len(txbuffer_48k), chunk_length) ] for c in chunk: # Pad the chunk, if needed if len(c) < chunk_length: delta = chunk_length - len(c) delta_zeros = np.zeros(delta, dtype=np.int16) c = np.append(c, delta_zeros) # self.log.debug("[MDM] mod out shorter than audio buffer", delta=delta) self.modoutqueue.append(c) # Release our mod_out_lock so we can use the queue self.mod_out_locked = False while self.modoutqueue: time.sleep(0.01) static.PTT_STATE = self.hamlib.set_ptt(False) # Push ptt state to socket stream jsondata = {"ptt": "False"} data_out = json.dumps(jsondata) sock.SOCKET_QUEUE.put(data_out) # After processing, set the locking state back to true to be prepared for next transmission self.mod_out_locked = True self.c_lib.freedv_close(freedv) self.modem_transmit_queue.task_done() static.TRANSMITTING = False threading.Event().set() def demodulate_audio( self, audiobuffer: codec2.audio_buffer, nin: int, freedv: ctypes.c_void_p, bytes_out, bytes_per_frame, ) -> int: """ De-modulate supplied audio stream with supplied codec2 instance. Decoded audio is placed into `bytes_out`. :param buffer: Incoming audio :type buffer: codec2.audio_buffer :param nin: Number of frames codec2 is expecting :type nin: int :param freedv: codec2 instance :type freedv: ctypes.c_void_p :param bytes_out: Demodulated audio :type bytes_out: _type_ :param bytes_per_frame: Number of bytes per frame :type bytes_per_frame: int :return: NIN from freedv instance :rtype: int """ nbytes = 0 while self.stream.active: threading.Event().wait(0.01) while audiobuffer.nbuffer >= nin: # demodulate audio nbytes = codec2.api.freedv_rawdatarx( freedv, bytes_out, audiobuffer.buffer.ctypes ) audiobuffer.pop(nin) nin = codec2.api.freedv_nin(freedv) if nbytes == bytes_per_frame: self.log.debug( "[MDM] [demod_audio] Pushing received data to received_queue" ) self.modem_received_queue.put([bytes_out, freedv, bytes_per_frame]) # self.get_scatter(freedv) self.calculate_snr(freedv) return nin def audio_datac0(self) -> None: """Receive data encoded with datac0""" self.datac0_nin = self.demodulate_audio( self.datac0_buffer, self.datac0_nin, self.datac0_freedv, self.datac0_bytes_out, self.datac0_bytes_per_frame, ) def audio_datac1(self) -> None: """Receive data encoded with datac1""" self.datac1_nin = self.demodulate_audio( self.datac1_buffer, self.datac1_nin, self.datac1_freedv, self.datac1_bytes_out, self.datac1_bytes_per_frame, ) def audio_datac3(self) -> None: """Receive data encoded with datac3""" self.datac3_nin = self.demodulate_audio( self.datac3_buffer, self.datac3_nin, self.datac3_freedv, self.datac3_bytes_out, self.datac3_bytes_per_frame, ) def audio_fsk_ldpc_0(self) -> None: """Receive data encoded with FSK + LDPC0""" self.fsk_ldpc_nin_0 = self.demodulate_audio( self.fsk_ldpc_buffer_0, self.fsk_ldpc_nin_0, self.fsk_ldpc_freedv_0, self.fsk_ldpc_bytes_out_0, self.fsk_ldpc_bytes_per_frame_0, ) def audio_fsk_ldpc_1(self) -> None: """Receive data encoded with FSK + LDPC1""" self.fsk_ldpc_nin_1 = self.demodulate_audio( self.fsk_ldpc_buffer_1, self.fsk_ldpc_nin_1, self.fsk_ldpc_freedv_1, self.fsk_ldpc_bytes_out_1, self.fsk_ldpc_bytes_per_frame_1, ) def worker_transmit(self) -> None: """Worker for FIFO queue for processing frames to be transmitted""" while True: data = self.modem_transmit_queue.get() self.log.debug("[MDM] worker_transmit", mode=data[0]) self.transmit( mode=data[0], repeats=data[1], repeat_delay=data[2], frames=data[3] ) # self.modem_transmit_queue.task_done() def worker_received(self) -> None: """Worker for FIFO queue for processing received frames""" while True: data = self.modem_received_queue.get() self.log.debug("[MDM] worker_received: received data!") # data[0] = bytes_out # data[1] = freedv session # data[2] = bytes_per_frame data_handler.DATA_QUEUE_RECEIVED.put([data[0], data[1], data[2]]) self.modem_received_queue.task_done() def get_frequency_offset(self, freedv: ctypes.c_void_p) -> float: """ Ask codec2 for the calculated (audio) frequency offset of the received signal. Side-effect: sets static.FREQ_OFFSET :param freedv: codec2 instance to query :type freedv: ctypes.c_void_p :return: Offset of audio frequency in Hz :rtype: float """ modemStats = codec2.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: ctypes.c_void_p) -> None: """ Ask codec2 for data about the received signal and calculate the scatter plot. Side-effect: sets static.SCATTER :param freedv: codec2 instance to query :type freedv: ctypes.c_void_p """ if not static.ENABLE_SCATTER: return modemStats = codec2.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(codec2.MODEM_STATS_NC_MAX): for j in range(codec2.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}) # Send all the data if we have too-few samples, otherwise send a sampling 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: ctypes.c_void_p) -> float: """ Ask codec2 for data about the received signal and calculate the signal-to-noise ratio. Side-effect: sets static.SNR :param freedv: codec2 instance to query :type freedv: ctypes.c_void_p :return: Signal-to-noise ratio of the decoded data :rtype: float """ try: modem_stats_snr = ctypes.c_float() modem_stats_sync = ctypes.c_int() self.c_lib.freedv_get_modem_stats( freedv, ctypes.byref(modem_stats_sync), ctypes.byref(modem_stats_snr) ) modem_stats_snr = modem_stats_snr.value modem_stats_sync = modem_stats_sync.value snr = round(modem_stats_snr, 1) self.log.info("[MDM] calculate_snr: ", snr=snr) # static.SNR = np.clip(snr, 0, 255) # limit to max value of 255 static.SNR = np.clip( snr, -128, 128 ) # limit to max value of -128/128 as a possible fix of #188 return static.SNR except Exception as err: self.log.error(f"[MDM] calculate_snr: Exception: {err}") static.SNR = 0 return static.SNR def update_rig_data(self) -> None: """ Request information about the current state of the radio via hamlib Side-effect: sets - static.HAMLIB_FREQUENCY - static.HAMLIB_MODE - static.HAMLIB_BANDWIDTH """ while True: threading.Event().wait(0.5) static.HAMLIB_FREQUENCY = self.hamlib.get_frequency() static.HAMLIB_MODE = self.hamlib.get_mode() static.HAMLIB_BANDWIDTH = self.hamlib.get_bandwidth() def calculate_fft(self) -> None: """ Calculate an average signal strength of the channel to assess whether the channel is "busy." """ # Initialize channel_busy_delay counter channel_busy_delay = 0 while True: # time.sleep(0.01) threading.Event().wait(0.01) # WE NEED TO OPTIMIZE THIS! # Start calculating the FFT once enough samples are captured. if len(self.fft_data) >= 128: # 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(self.fft_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 static.TRANSMITTING: dfft[dfft > avg + 10] = 100 # Calculate audio max value # static.AUDIO_RMS = np.amax(self.fft_data) # 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(dfft[dfft > avg + 10]) >= 300 and not static.TRANSMITTING: static.CHANNEL_BUSY = True # Limit delay counter to a maximun of 50. The higher this value, # the longer we will wait until releasing state channel_busy_delay = min(channel_busy_delay + 5, 50) 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: static.CHANNEL_BUSY = False # Round data to decrease size dfft = np.around(dfft, 0) dfftlist = dfft.tolist() static.FFT = dfftlist[:320] # 320 --> bandwidth 3000 except Exception as err: self.log.error(f"[MDM] calculate_fft: Exception: {err}") self.log.debug("[MDM] Setting fft=0") # else 0 static.FFT = [0] def set_frames_per_burst(self, frames_per_burst: int) -> None: """ Configure codec2 to send the configured number of frames per burst. :param frames_per_burst: Number of frames per burst requested :type frames_per_burst: int """ # Limit frames per burst to acceptable values frames_per_burst = min(frames_per_burst, 1) frames_per_burst = max(frames_per_burst, 5) codec2.api.freedv_set_frames_per_burst(self.datac1_freedv, frames_per_burst) codec2.api.freedv_set_frames_per_burst(self.datac3_freedv, frames_per_burst) codec2.api.freedv_set_frames_per_burst(self.fsk_ldpc_freedv_0, frames_per_burst) def open_codec2_instance(mode: int) -> ctypes.c_void_p: """ Return a codec2 instance of the type `mode` :param mode: Type of codec2 instance to return :type mode: Union[int, str] :return: C-function of the requested codec2 instance :rtype: ctypes.c_void_p """ if mode in [codec2.FREEDV_MODE.fsk_ldpc_0.value]: return ctypes.cast( codec2.api.freedv_open_advanced( codec2.api.FREEDV_MODE_FSK_LDPC, ctypes.byref(codec2.api.FREEDV_MODE_FSK_LDPC_0_ADV), ), ctypes.c_void_p, ) if mode in [codec2.FREEDV_MODE.fsk_ldpc_1.value]: return ctypes.cast( codec2.api.freedv_open_advanced( codec2.api.FREEDV_MODE_FSK_LDPC, ctypes.byref(codec2.api.FREEDV_MODE_FSK_LDPC_1_ADV), ), ctypes.c_void_p, ) return ctypes.cast(codec2.api.freedv_open(mode), ctypes.c_void_p) def get_bytes_per_frame(mode: int) -> int: """ Provide bytes per frame information for accessing from data handler :param mode: Codec2 mode to query :type mode: int or str :return: Bytes per frame of the supplied codec2 data mode :rtype: int """ freedv = open_codec2_instance(mode) # get number of bytes per frame for mode return int(codec2.api.freedv_get_bits_per_modem_frame(freedv) / 8) def set_audio_volume(datalist, volume: float) -> np.int16: """ Scale values for the provided audio samples by volume, `volume` is clipped to the range of 0-200 :param datalist: Audio samples to scale :type datalist: NDArray[np.int16] :param volume: "Percentage" (0-200) to scale samples :type volume: float :return: Scaled audio samples :rtype: np.int16 """ # Clip volume provided to acceptable values volume = np.clip(volume, 0, 200) # limit to max value of 255 # Scale samples by the ratio of volume / 100.0 data = np.fromstring(datalist, np.int16) * (volume / 100.0) # type: ignore return data.astype(np.int16)