FreeDATA/tnc/modem.py

#!/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
from typing import Union

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"""

    def __init__(self):

        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_CHUNKS = 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)
        # structlog.get_logger("structlog").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)
                structlog.get_logger("structlog").info(
                    "[MDM] init: opened audio devices"
                )
            except Exception as e:
                structlog.get_logger("structlog").error(
                    "[MDM] init: can't open audio device. Exit", e=e
                )
                sys.exit(1)

            try:
                structlog.get_logger("structlog").debug(
                    "[MDM] init: starting pyaudio callback"
                )
                # self.audio_stream.start_stream()
                self.stream.start()
            except Exception as e:
                structlog.get_logger("structlog").error(
                    "[MDM] init: starting pyaudio callback failed", e=e
                )
        else:

            class Object:
                """An object for simulating audio stream"""

                active = True

            self.stream = Object()
            self.stream.active = True

            # Create mkfifo buffers
            try:
                os.mkfifo(RXCHANNEL)
                os.mkfifo(TXCHANNEL)
            except Exception as e:
                structlog.get_logger("structlog").error(
                    f"[MDM] init:mkfifo: Exception: {e}"
                )

            mkfifo_write_callback_thread = threading.Thread(
                target=self.mkfifo_write_callback,
                name="MKFIFO WRITE CALLBACK THREAD",
                daemon=True,
            )
            mkfifo_write_callback_thread.start()

            structlog.get_logger("structlog").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()

        # --------------------------------------------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()

        # structlog.get_logger("structlog").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):
        """
        Support testing by reading the audio data from a pipe and
        depositing the data into the codec data buffers.
        """
        while 1:
            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),
                        ]:
                            if (
                                not data_buffer.nbuffer + length_x > data_buffer.size
                                and receive
                            ):
                                data_buffer.push(x)

    def mkfifo_write_callback(self):
        """Support testing by writing the audio data to a pipe."""
        while 1:
            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):
        """
        Receive data into appropriate queue.

        Args:
            data_in48k: Incoming data received
            outdata: Container for the data returned
            frames: Number of frames
            time:
          status:

        """
        structlog.get_logger("structlog").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 not full
            if not self.datac0_buffer.nbuffer + length_x > self.datac0_buffer.size:
                self.datac0_buffer.push(x)
            else:
                static.BUFFER_OVERFLOW_COUNTER[0] += 1

            # Avoid buffer overflow by filling only if buffer for
            # selected datachannel mode is not full
            if self.datac1_buffer.nbuffer + length_x > self.datac1_buffer.size:
                static.BUFFER_OVERFLOW_COUNTER[1] += 1
            elif RECEIVE_DATAC1:
                self.datac1_buffer.push(x)

            # Avoid buffer overflow by filling only if buffer for
            # selected datachannel mode is not full
            if self.datac3_buffer.nbuffer + length_x > self.datac3_buffer.size:
                static.BUFFER_OVERFLOW_COUNTER[2] += 1
            elif RECEIVE_DATAC3:
                self.datac3_buffer.push(x)

            # Avoid buffer overflow by filling only if buffer for
            # selected datachannel mode is not full
            if self.fsk_ldpc_buffer_0.nbuffer + length_x > self.fsk_ldpc_buffer_0.size:
                static.BUFFER_OVERFLOW_COUNTER[3] += 1
            elif static.ENABLE_FSK:
                self.fsk_ldpc_buffer_0.push(x)

            # Avoid buffer overflow by filling only if buffer for
            # selected datachannel mode is not full
            if self.fsk_ldpc_buffer_1.nbuffer + length_x > self.fsk_ldpc_buffer_1.size:
                static.BUFFER_OVERFLOW_COUNTER[4] += 1
            elif RECEIVE_FSK_LDPC_1 and static.ENABLE_FSK:
                self.fsk_ldpc_buffer_1.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 e:
            structlog.get_logger("structlog").debug(f"[MDM] callback: IndexError: {e}")

        # return (data_out48k, audio.pyaudio.paContinue)

    # --------------------------------------------------------------------
    def transmit(self, mode, repeats: int, repeat_delay: int, frames: bytearray):
        """

        Args:
          mode:
          repeats:
          repeat_delay:
          frames:

        """
        structlog.get_logger("structlog").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)

        structlog.get_logger("structlog").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 ["FSK_LDPC_0", "FSK_LDPC_1", 200, 201]:
                # 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 ["FSK_LDPC_0", "FSK_LDPC_1", 200, 201]:
                # 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.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,
    ):
        """
        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:
                    structlog.get_logger("structlog").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):
        """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):
        """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):
        """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):
        """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):
        """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):
        """Worker for FIFO queue for processing frames to be transmitted"""
        while True:
            data = self.modem_transmit_queue.get()

            structlog.get_logger("structlog").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):
        """Worker for FIFO queue for processing received frames"""
        while True:
            data = self.modem_received_queue.get()
            structlog.get_logger("structlog").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):
        """
        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)
            structlog.get_logger("structlog").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 e:
            structlog.get_logger("structlog").error(
                f"[MDM] calculate_snr: Exception: {e}"
            )
            static.SNR = 0
            return static.SNR

    def update_rig_data(self):
        """
        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_bandwith()

    def calculate_fft(self):
        """
        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 e:
                    structlog.get_logger("structlog").error(
                        f"[MDM] calculate_fft: Exception: {e}"
                    )
                    structlog.get_logger("structlog").debug("[MDM] Setting fft=0")
                    # else 0
                    static.FFT = [0]

    def set_frames_per_burst(self, frames_per_burst: int):
        """
        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: Union[int, str]) -> 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 ["FSK_LDPC_0", 200]:
        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 ["FSK_LDPC_1", 201]:
        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: Union[int, str]) -> 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: np.int16, volume: float) -> np.int16:
    """
    Scale values for the provided audio samples by volume,
    `volume` is clipped to the range of 0-100

    :param datalist: Audio samples to scale
    :type datalist: np.int16
    :param volume: Percentage (0-100) to scale samples
    :type volume: float
    :return: Scaled audio samples
    :rtype: np.int16
    """
    # Clip volume provided to acceptable values
    volume = min(volume, 0.0)
    volume = max(volume, 100.0)

    # 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)