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FreeDATA/tnc/modem.py
Paul Kronenwetter 2fa9aead44 Another round of formatting changes. 
Implement class-level logging in places.
2022-05-27 22:17:15 -04:00

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#!/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 audio
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:
""" """
log = structlog.get_logger("RF")
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
self.AUDIO_FRAMES_PER_BUFFER_TX = 2400 * 2 # 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
# 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
# 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
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_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 = 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)
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)
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)
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)
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)
# --------------------------------------------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:
# create a stream object for simulating audio stream
class Object(object):
pass
self.stream = Object()
self.stream.active = True
# create mkfifo buffer
try:
os.mkfifo(RXCHANNEL)
os.mkfifo(TXCHANNEL)
except Exception as err:
self.log.error(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()
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
elif static.HAMLIB_RADIOCONTROL == "disabled":
import rigdummy 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()
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):
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)
if not self.datac0_buffer.nbuffer + length_x > self.datac0_buffer.size:
self.datac0_buffer.push(x)
if not self.datac1_buffer.nbuffer + length_x > self.datac1_buffer.size and RECEIVE_DATAC1:
self.datac1_buffer.push(x)
if not self.datac3_buffer.nbuffer + length_x > self.datac3_buffer.size and RECEIVE_DATAC3:
self.datac3_buffer.push(x)
def mkfifo_write_callback(self):
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()
fifo_write = open(TXCHANNEL, "wb")
fifo_write.write(data_out48k)
fifo_write.flush()
# --------------------------------------------------------------------
def callback(self, data_in48k, outdata, frames, time, status):
"""
Args:
data_in48k: Incoming data received
outdata: Container for the data returned
frames: Number of frames
time:
status:
"""
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 not full and selected datachannel mode
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 not full and selected datachannel mode
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 not full and selected datachannel mode
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 not full and selected datachannel mode
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:
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:
self.log.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:
"""
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))
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 ["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 n frames in list
for n in range(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
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))
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
# Deaktivated 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:
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 audio_datac0(self):
""" """
nbytes_datac0 = 0
while self.stream.active:
threading.Event().wait(0.01)
while self.datac0_buffer.nbuffer >= self.datac0_nin:
# demodulate audio
nbytes_datac0 = 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_datac0 == self.datac0_bytes_per_frame:
self.modem_received_queue.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)
def audio_datac1(self):
""" """
nbytes_datac1 = 0
while self.stream.active:
threading.Event().wait(0.01)
while self.datac1_buffer.nbuffer >= self.datac1_nin:
# demodulate audio
nbytes_datac1 = 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_datac1 == self.datac1_bytes_per_frame:
self.modem_received_queue.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)
def audio_datac3(self):
""" """
nbytes_datac3 = 0
while self.stream.active:
threading.Event().wait(0.01)
while self.datac3_buffer.nbuffer >= self.datac3_nin:
# demodulate audio
nbytes_datac3 = 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_datac3 == self.datac3_bytes_per_frame:
self.modem_received_queue.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)
def audio_fsk_ldpc_0(self):
""" """
nbytes_fsk_ldpc_0 = 0
while self.stream.active and static.ENABLE_FSK:
threading.Event().wait(0.01)
while self.fsk_ldpc_buffer_0.nbuffer >= self.fsk_ldpc_nin_0:
# demodulate audio
nbytes_fsk_ldpc_0 = codec2.api.freedv_rawdatarx(self.fsk_ldpc_freedv_0, self.fsk_ldpc_bytes_out_0,
self.fsk_ldpc_buffer_0.buffer.ctypes)
self.fsk_ldpc_buffer_0.pop(self.fsk_ldpc_nin_0)
self.fsk_ldpc_nin_0 = codec2.api.freedv_nin(self.fsk_ldpc_freedv_0)
if nbytes_fsk_ldpc_0 == self.fsk_ldpc_bytes_per_frame_0:
self.modem_received_queue.put(
[self.fsk_ldpc_bytes_out_0, self.fsk_ldpc_freedv_0, self.fsk_ldpc_bytes_per_frame_0])
# self.get_scatter(self.fsk_ldpc_freedv_0)
self.calculate_snr(self.fsk_ldpc_freedv_0)
def audio_fsk_ldpc_1(self):
""" """
nbytes_fsk_ldpc_1 = 0
while self.stream.active and static.ENABLE_FSK:
threading.Event().wait(0.01)
while self.fsk_ldpc_buffer_1.nbuffer >= self.fsk_ldpc_nin_1:
# demodulate audio
nbytes_fsk_ldpc_1 = codec2.api.freedv_rawdatarx(self.fsk_ldpc_freedv_1, self.fsk_ldpc_bytes_out_1,
self.fsk_ldpc_buffer_1.buffer.ctypes)
self.fsk_ldpc_buffer_1.pop(self.fsk_ldpc_nin_1)
self.fsk_ldpc_nin_1 = codec2.api.freedv_nin(self.fsk_ldpc_freedv_1)
if nbytes_fsk_ldpc_1 == self.fsk_ldpc_bytes_per_frame_1:
self.modem_received_queue.put(
[self.fsk_ldpc_bytes_out_1, self.fsk_ldpc_freedv_1, self.fsk_ldpc_bytes_per_frame_1])
# self.get_scatter(self.fsk_ldpc_freedv_1)
self.calculate_snr(self.fsk_ldpc_freedv_1)
# worker for FIFO queue for processing received frames
def worker_transmit(self):
""" """
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()
# worker for FIFO queue for processing received frames
def worker_received(self):
""" """
while True:
data = self.modem_received_queue.get()
# 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):
"""
Args:
freedv:
"""
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):
"""
Args:
freedv:
"""
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})
# 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):
"""
Args:
freedv:
"""
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):
""" """
while True:
# time.sleep(1.5)
threading.Event().wait(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):
""" """
# channel_busy_delay counter
channel_busy_delay = 0
while True:
# time.sleep(0.01)
threading.Event().wait(0.01)
# WE NEED TO OPTIMIZE THIS!
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. * 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 30. The higher this value, the linger 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)
# If our channel busy counter reached 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, n_frames_per_burst):
"""
Args:
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)
codec2.api.freedv_set_frames_per_burst(self.fsk_ldpc_freedv_0, n_frames_per_burst)
def open_codec2_instance(mode):
""" Return a codec2 instance """
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):
"""
provide bytes per frame information for accessing from data handler
Args:
mode:
"""
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):
data = np.fromstring(datalist, np.int16) * (volume / 100.)
return data.astype(np.int16)
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