FreeDATA/tnc/modem.py
dj2ls 0db0fcc27e changed clip range for snr to -128/128 instead of 0/255
This possibly fixes the SNR problem, but needs to be tested carefully as we had a reason for using np.clip()
2022-05-20 09:26:01 +02:00

742 lines
32 KiB
Python

#!/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 logging
import os
import pathlib
import queue
import re
import sys
import threading
import time
from collections import deque
import numpy as np
import sounddevice as sd
import structlog
import ujson as json
import audio
import codec2
import data_handler
import helpers
import log_handler
import sock
import static
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():
""" """
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)
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:
# 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 e:
structlog.get_logger("structlog").error(f"[MDM] init:mkfifo: Exception: {e}")
pass
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 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)
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 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))
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:
Returns:
Nothing
"""
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 not self.datac1_buffer.nbuffer + length_x > self.datac1_buffer.size:
if RECEIVE_DATAC1:
self.datac1_buffer.push(x)
else:
static.BUFFER_OVERFLOW_COUNTER[1] += 1
# Avoid buffer overflow by filling only if buffer not full and selected datachannel mode
if not self.datac3_buffer.nbuffer + length_x > self.datac3_buffer.size:
if RECEIVE_DATAC3:
self.datac3_buffer.push(x)
else:
static.BUFFER_OVERFLOW_COUNTER[2] += 1
# Avoid buffer overflow by filling only if buffer not full and selected datachannel mode
if not self.fsk_ldpc_buffer_0.nbuffer + length_x > self.fsk_ldpc_buffer_0.size:
if static.ENABLE_FSK:
self.fsk_ldpc_buffer_0.push(x)
else:
static.BUFFER_OVERFLOW_COUNTER[3] += 1
# Avoid buffer overflow by filling only if buffer not full and selected datachannel mode
if not self.fsk_ldpc_buffer_1.nbuffer + length_x > self.fsk_ldpc_buffer_1.size:
if RECEIVE_FSK_LDPC_1 and static.ENABLE_FSK:
self.fsk_ldpc_buffer_1.push(x)
else:
static.BUFFER_OVERFLOW_COUNTER[4] += 1
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:
Returns:
"""
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))
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 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)
#structlog.get_logger("structlog").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()
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()
# 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:
Returns:
"""
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:
Returns:
"""
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:
Returns:
"""
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):
""" """
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 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, n_frames_per_burst):
"""
Args:
n_frames_per_burst:
Returns:
"""
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:
Returns:
"""
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)