mirror of
https://github.com/DJ2LS/FreeDATA
synced 2024-05-14 10:04:33 +02:00
739 lines
32 KiB
Python
739 lines
32 KiB
Python
#!/usr/bin/env python3
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# -*- coding: utf-8 -*-
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"""
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Created on Wed Dec 23 07:04:24 2020
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@author: DJ2LS
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"""
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import sys
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import os
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import ctypes
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from ctypes import *
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import pathlib
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import logging, structlog, log_handler
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import time
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import threading
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import atexit
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import numpy as np
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import helpers
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import static
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import data_handler
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import ujson as json
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import sock
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import re
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import queue
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import codec2
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import audio
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import sounddevice as sd
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from collections import deque
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# init FIFO queue to store received frames in
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MODEM_RECEIVED_QUEUE = queue.Queue()
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MODEM_TRANSMIT_QUEUE = queue.Queue()
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static.TRANSMITTING = False
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# receive only specific modes to reduce cpu load
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RECEIVE_DATAC1 = False
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RECEIVE_DATAC3 = False
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RECEIVE_FSK_LDPC_1 = False
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class RF():
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""" """
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def __init__(self):
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self.sampler_avg = 0
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self.buffer_avg = 0
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self.AUDIO_SAMPLE_RATE_RX = 48000
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self.AUDIO_SAMPLE_RATE_TX = 48000
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self.MODEM_SAMPLE_RATE = codec2.api.FREEDV_FS_8000
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self.AUDIO_FRAMES_PER_BUFFER_RX = 2400*2 #8192
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self.AUDIO_FRAMES_PER_BUFFER_TX = 2400*2 #8192 Lets to some tests with very small chunks for TX
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self.AUDIO_CHUNKS = 48 #8 * (self.AUDIO_SAMPLE_RATE_RX/self.MODEM_SAMPLE_RATE) #48
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self.AUDIO_CHANNELS = 1
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# locking state for mod out so buffer will be filled before we can use it
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# https://github.com/DJ2LS/FreeDATA/issues/127
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# https://github.com/DJ2LS/FreeDATA/issues/99
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self.mod_out_locked = True
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# make sure our resampler will work
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assert (self.AUDIO_SAMPLE_RATE_RX / self.MODEM_SAMPLE_RATE) == codec2.api.FDMDV_OS_48
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# small hack for initializing codec2 via codec2.py module
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# TODO: we need to change the entire modem module to integrate codec2 module
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self.c_lib = codec2.api
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self.resampler = codec2.resampler()
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self.modem_transmit_queue = MODEM_TRANSMIT_QUEUE
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self.modem_received_queue = MODEM_RECEIVED_QUEUE
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# init FIFO queue to store modulation out in
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self.modoutqueue = deque()
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# define fft_data buffer
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self.fft_data = bytes()
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# open codec2 instance
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self.datac0_freedv = cast(codec2.api.freedv_open(codec2.api.FREEDV_MODE_DATAC0), c_void_p)
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self.c_lib.freedv_set_tuning_range(self.datac0_freedv, c_float(static.TUNING_RANGE_FMIN), c_float(static.TUNING_RANGE_FMAX))
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self.datac0_bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(self.datac0_freedv)/8)
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self.datac0_payload_per_frame = self.datac0_bytes_per_frame -2
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self.datac0_n_nom_modem_samples = self.c_lib.freedv_get_n_nom_modem_samples(self.datac0_freedv)
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self.datac0_n_tx_modem_samples = self.c_lib.freedv_get_n_tx_modem_samples(self.datac0_freedv)
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self.datac0_n_tx_preamble_modem_samples = self.c_lib.freedv_get_n_tx_preamble_modem_samples(self.datac0_freedv)
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self.datac0_n_tx_postamble_modem_samples = self.c_lib.freedv_get_n_tx_postamble_modem_samples(self.datac0_freedv)
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self.datac0_bytes_out = create_string_buffer(self.datac0_bytes_per_frame)
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codec2.api.freedv_set_frames_per_burst(self.datac0_freedv,1)
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self.datac0_buffer = codec2.audio_buffer(2*self.AUDIO_FRAMES_PER_BUFFER_RX)
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self.datac1_freedv = cast(codec2.api.freedv_open(codec2.api.FREEDV_MODE_DATAC1), c_void_p)
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self.c_lib.freedv_set_tuning_range(self.datac1_freedv, c_float(static.TUNING_RANGE_FMIN), c_float(static.TUNING_RANGE_FMAX))
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self.datac1_bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(self.datac1_freedv)/8)
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self.datac1_bytes_out = create_string_buffer(self.datac1_bytes_per_frame)
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codec2.api.freedv_set_frames_per_burst(self.datac1_freedv,1)
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self.datac1_buffer = codec2.audio_buffer(2*self.AUDIO_FRAMES_PER_BUFFER_RX)
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self.datac3_freedv = cast(codec2.api.freedv_open(codec2.api.FREEDV_MODE_DATAC3), c_void_p)
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self.c_lib.freedv_set_tuning_range(self.datac3_freedv, c_float(static.TUNING_RANGE_FMIN), c_float(static.TUNING_RANGE_FMAX))
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self.datac3_bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(self.datac3_freedv)/8)
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self.datac3_bytes_out = create_string_buffer(self.datac3_bytes_per_frame)
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codec2.api.freedv_set_frames_per_burst(self.datac3_freedv,1)
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self.datac3_buffer = codec2.audio_buffer(2*self.AUDIO_FRAMES_PER_BUFFER_RX)
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self.fsk_ldpc_freedv_0 = cast(codec2.api.freedv_open_advanced(codec2.api.FREEDV_MODE_FSK_LDPC, ctypes.byref(codec2.api.FREEDV_MODE_FSK_LDPC_0_ADV)), c_void_p)
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self.fsk_ldpc_bytes_per_frame_0 = int(codec2.api.freedv_get_bits_per_modem_frame(self.fsk_ldpc_freedv_0)/8)
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self.fsk_ldpc_bytes_out_0 = create_string_buffer(self.fsk_ldpc_bytes_per_frame_0)
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#codec2.api.freedv_set_frames_per_burst(self.fsk_ldpc_freedv_0,1)
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self.fsk_ldpc_buffer_0 = codec2.audio_buffer(self.AUDIO_FRAMES_PER_BUFFER_RX)
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self.fsk_ldpc_freedv_1 = cast(codec2.api.freedv_open_advanced(codec2.api.FREEDV_MODE_FSK_LDPC, ctypes.byref(codec2.api.FREEDV_MODE_FSK_LDPC_1_ADV)), c_void_p)
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self.fsk_ldpc_bytes_per_frame_1 = int(codec2.api.freedv_get_bits_per_modem_frame(self.fsk_ldpc_freedv_1)/8)
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self.fsk_ldpc_bytes_out_1 = create_string_buffer(self.fsk_ldpc_bytes_per_frame_1)
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#codec2.api.freedv_set_frames_per_burst(self.fsk_ldpc_freedv_0,1)
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self.fsk_ldpc_buffer_1 = codec2.audio_buffer(self.AUDIO_FRAMES_PER_BUFFER_RX)
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# initial nin values
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self.datac0_nin = codec2.api.freedv_nin(self.datac0_freedv)
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self.datac1_nin = codec2.api.freedv_nin(self.datac1_freedv)
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self.datac3_nin = codec2.api.freedv_nin(self.datac3_freedv)
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self.fsk_ldpc_nin_0 = codec2.api.freedv_nin(self.fsk_ldpc_freedv_0)
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self.fsk_ldpc_nin_1 = codec2.api.freedv_nin(self.fsk_ldpc_freedv_1)
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# --------------------------------------------CREATE PYAUDIO INSTANCE
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'''
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try:
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# we need to "try" this, because sometimes libasound.so isn't in the default place
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# try to supress error messages
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with audio.noalsaerr(): # https://github.com/DJ2LS/FreeDATA/issues/22
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self.p = audio.pyaudio.PyAudio()
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# else do it the default way
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except:
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self.p = audio.pyaudio.PyAudio()
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atexit.register(self.p.terminate)
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# --------------------------------------------OPEN RX AUDIO CHANNEL
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# optional auto selection of loopback device if using in testmode
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if static.AUDIO_INPUT_DEVICE == -2:
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loopback_list = []
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for dev in range(0,self.p.get_device_count()):
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if 'Loopback: PCM' in self.p.get_device_info_by_index(dev)["name"]:
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loopback_list.append(dev)
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if len(loopback_list) >= 2:
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static.AUDIO_INPUT_DEVICE = loopback_list[0] #0 = RX
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static.AUDIO_OUTPUT_DEVICE = loopback_list[1] #1 = TX
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print(f"loopback_list rx: {loopback_list}", file=sys.stderr)
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'''
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try:
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'''
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self.audio_stream = self.p.open(format=audio.pyaudio.paInt16,
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channels=self.AUDIO_CHANNELS,
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rate=self.AUDIO_SAMPLE_RATE_RX,
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frames_per_buffer=self.AUDIO_FRAMES_PER_BUFFER_RX,
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input=True,
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output=True,
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input_device_index=static.AUDIO_INPUT_DEVICE,
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output_device_index=static.AUDIO_OUTPUT_DEVICE,
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stream_callback=self.audio_callback
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)
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'''
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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)
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atexit.register(self.stream.stop)
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structlog.get_logger("structlog").info("opened audio devices")
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except Exception as e:
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structlog.get_logger("structlog").error("can't open audio device. Exit", e=e)
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os._exit(1)
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try:
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structlog.get_logger("structlog").debug("[TNC] starting pyaudio callback")
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#self.audio_stream.start_stream()
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self.stream.start()
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except Exception as e:
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structlog.get_logger("structlog").error("[TNC] starting pyaudio callback failed", e=e)
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# --------------------------------------------INIT AND OPEN HAMLIB
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# check how we want to control the radio
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if static.HAMLIB_RADIOCONTROL == 'direct':
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import rig
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elif static.HAMLIB_RADIOCONTROL == 'rigctl':
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import rigctl as rig
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elif static.HAMLIB_RADIOCONTROL == 'rigctld':
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import rigctld as rig
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elif static.HAMLIB_RADIOCONTROL == 'disabled':
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import rigdummy as rig
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else:
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import rigdummy as rig
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self.hamlib = rig.radio()
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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_RGICTLD_IP, rigctld_port = static.HAMLIB_RGICTLD_PORT)
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# --------------------------------------------START DECODER THREAD
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if static.ENABLE_FFT:
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fft_thread = threading.Thread(target=self.calculate_fft, name="FFT_THREAD" ,daemon=True)
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fft_thread.start()
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audio_thread_datac0 = threading.Thread(target=self.audio_datac0, name="AUDIO_THREAD DATAC0",daemon=True)
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audio_thread_datac0.start()
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audio_thread_datac1 = threading.Thread(target=self.audio_datac1, name="AUDIO_THREAD DATAC1",daemon=True)
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audio_thread_datac1.start()
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audio_thread_datac3 = threading.Thread(target=self.audio_datac3, name="AUDIO_THREAD DATAC3",daemon=True)
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audio_thread_datac3.start()
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if static.ENABLE_FSK:
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audio_thread_fsk_ldpc0 = threading.Thread(target=self.audio_fsk_ldpc_0, name="AUDIO_THREAD FSK LDPC0",daemon=True)
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audio_thread_fsk_ldpc0.start()
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audio_thread_fsk_ldpc1 = threading.Thread(target=self.audio_fsk_ldpc_1, name="AUDIO_THREAD FSK LDPC1",daemon=True)
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audio_thread_fsk_ldpc1.start()
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hamlib_thread = threading.Thread(target=self.update_rig_data, name="HAMLIB_THREAD",daemon=True)
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hamlib_thread.start()
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worker_received = threading.Thread(target=self.worker_received, name="WORKER_THREAD",daemon=True)
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worker_received.start()
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worker_transmit = threading.Thread(target=self.worker_transmit, name="WORKER_THREAD",daemon=True)
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worker_transmit.start()
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# --------------------------------------------------------------------------------------------------------
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#def audio_callback(self, data_in48k, frame_count, time_info, status):
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def callback(self, data_in48k, outdata, frames, time, status):
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"""
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Args:
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data_in48k:
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frame_count:
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time_info:
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status:
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Returns:
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"""
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x = np.frombuffer(data_in48k, dtype=np.int16)
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x = self.resampler.resample48_to_8(x)
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length_x = len(x)
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# avoid decoding when transmitting to reduce CPU
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if not static.TRANSMITTING:
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# avoid buffer overflow by filling only if buffer not full
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if not self.datac0_buffer.nbuffer+length_x > self.datac0_buffer.size:
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self.datac0_buffer.push(x)
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else:
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static.BUFFER_OVERFLOW_COUNTER[0] += 1
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# avoid buffer overflow by filling only if buffer not full and selected datachannel mode
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if not self.datac1_buffer.nbuffer+length_x > self.datac1_buffer.size:
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if RECEIVE_DATAC1:
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self.datac1_buffer.push(x)
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else:
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static.BUFFER_OVERFLOW_COUNTER[1] += 1
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# avoid buffer overflow by filling only if buffer not full and selected datachannel mode
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if not self.datac3_buffer.nbuffer+length_x > self.datac3_buffer.size:
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if RECEIVE_DATAC3:
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self.datac3_buffer.push(x)
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else:
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static.BUFFER_OVERFLOW_COUNTER[2] += 1
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# avoid buffer overflow by filling only if buffer not full and selected datachannel mode
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if not self.fsk_ldpc_buffer_0.nbuffer+length_x > self.fsk_ldpc_buffer_0.size:
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if static.ENABLE_FSK:
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self.fsk_ldpc_buffer_0.push(x)
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else:
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static.BUFFER_OVERFLOW_COUNTER[3] += 1
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# avoid buffer overflow by filling only if buffer not full and selected datachannel mode
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if not self.fsk_ldpc_buffer_1.nbuffer+length_x > self.fsk_ldpc_buffer_1.size:
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if RECEIVE_FSK_LDPC_1 and static.ENABLE_FSK:
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self.fsk_ldpc_buffer_1.push(x)
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else:
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static.BUFFER_OVERFLOW_COUNTER[4] += 1
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if len(self.modoutqueue) <= 0 or self.mod_out_locked:
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#if not self.modoutqueue or self.mod_out_locked:
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data_out48k = np.zeros(frames, dtype=np.int16)
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self.fft_data = bytes(x)
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else:
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data_out48k = self.modoutqueue.popleft()
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self.fft_data = bytes(data_out48k)
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try:
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outdata[:] = data_out48k[:frames]
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except Exception as e:
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print(e)
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#return (data_out48k, audio.pyaudio.paContinue)
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# --------------------------------------------------------------------------------------------------------
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def transmit(self, mode, repeats, repeat_delay, frames):
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"""
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Args:
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mode:
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repeats:
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repeat_delay:
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frames:
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Returns:
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"""
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static.TRANSMITTING = True
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# toggle ptt early to save some time and send ptt state via socket
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static.PTT_STATE = self.hamlib.set_ptt(True)
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jsondata = {"ptt":"True"}
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data_out = json.dumps(jsondata)
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sock.SOCKET_QUEUE.put(data_out)
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# open codec2 instance
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self.MODE = mode
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print(self.MODE)
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if self.MODE == 'FSK_LDPC_0' or self.MODE == 200:
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freedv = cast(codec2.api.freedv_open_advanced(codec2.api.FREEDV_MODE_FSK_LDPC, ctypes.byref(codec2.api.FREEDV_MODE_FSK_LDPC_0_ADV)), c_void_p)
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elif self.MODE == 'FSK_LDPC_1' or self.MODE == 201:
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freedv = cast(codec2.api.freedv_open_advanced(codec2.api.FREEDV_MODE_FSK_LDPC, ctypes.byref(codec2.api.FREEDV_MODE_FSK_LDPC_1_ADV)), c_void_p)
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else:
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freedv = cast(codec2.api.freedv_open(self.MODE), c_void_p)
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# get number of bytes per frame for mode
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bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(freedv)/8)
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payload_bytes_per_frame = bytes_per_frame -2
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print(bytes_per_frame)
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# init buffer for data
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n_tx_modem_samples = codec2.api.freedv_get_n_tx_modem_samples(freedv)
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mod_out = create_string_buffer(n_tx_modem_samples * 2)
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# init buffer for preample
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n_tx_preamble_modem_samples = codec2.api.freedv_get_n_tx_preamble_modem_samples(freedv)
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mod_out_preamble = create_string_buffer(n_tx_preamble_modem_samples * 2)
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# init buffer for postamble
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n_tx_postamble_modem_samples = codec2.api.freedv_get_n_tx_postamble_modem_samples(freedv)
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mod_out_postamble = create_string_buffer(n_tx_postamble_modem_samples * 2)
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# add empty data to handle ptt toggle time
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data_delay_mseconds = 0 #miliseconds
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data_delay = int(self.MODEM_SAMPLE_RATE*(data_delay_mseconds/1000))
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mod_out_silence = create_string_buffer(data_delay*2)
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txbuffer = bytes(mod_out_silence)
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for i in range(1,repeats+1):
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# write preamble to txbuffer
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codec2.api.freedv_rawdatapreambletx(freedv, mod_out_preamble)
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txbuffer += bytes(mod_out_preamble)
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# create modulaton for n frames in list
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for n in range(0,len(frames)):
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# create buffer for data
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buffer = bytearray(payload_bytes_per_frame) # use this if CRC16 checksum is required ( DATA1-3)
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buffer[:len(frames[n])] = frames[n] # set buffersize to length of data which will be send
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# create crc for data frame - we are using the crc function shipped with codec2 to avoid
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# crc algorithm incompatibilities
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crc = ctypes.c_ushort(codec2.api.freedv_gen_crc16(bytes(buffer), payload_bytes_per_frame)) # generate CRC16
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crc = crc.value.to_bytes(2, byteorder='big') # convert crc to 2 byte hex string
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buffer += crc # append crc16 to buffer
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data = (ctypes.c_ubyte * bytes_per_frame).from_buffer_copy(buffer)
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codec2.api.freedv_rawdatatx(freedv,mod_out,data) # modulate DATA and save it into mod_out pointer
|
|
txbuffer += bytes(mod_out)
|
|
|
|
|
|
|
|
# append postamble to txbuffer
|
|
codec2.api.freedv_rawdatapostambletx(freedv, mod_out_postamble)
|
|
txbuffer += bytes(mod_out_postamble)
|
|
# add delay to end of frames
|
|
samples_delay = int(self.MODEM_SAMPLE_RATE*(repeat_delay/1000))
|
|
mod_out_silence = create_string_buffer(samples_delay*2)
|
|
txbuffer += bytes(mod_out_silence)
|
|
|
|
# resample 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("[TNC] 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 we want to 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.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 = 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 static.ENABLE_SCATTER:
|
|
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 = c_float()
|
|
modem_stats_sync = c_int()
|
|
|
|
self.c_lib.freedv_get_modem_stats(freedv, byref(modem_stats_sync), byref(modem_stats_snr))
|
|
modem_stats_snr = modem_stats_snr.value
|
|
modem_stats_sync = modem_stats_sync.value
|
|
|
|
snr = round(modem_stats_snr, 1)
|
|
print(snr)
|
|
static.SNR = np.clip(snr, 0, 255) #limit to max value of 255
|
|
return static.SNR
|
|
except:
|
|
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:
|
|
|
|
data_in = self.fft_data
|
|
|
|
# delete fft_buffer
|
|
self.fft_data = bytes()
|
|
|
|
# https://gist.github.com/ZWMiller/53232427efc5088007cab6feee7c6e4c
|
|
audio_data = np.fromstring(data_in, np.int16)
|
|
# Fast Fourier Transform, 10*log10(abs) is to scale it to dB
|
|
# and make sure it's not imaginary
|
|
|
|
try:
|
|
fftarray = np.fft.rfft(audio_data)
|
|
|
|
# set value 0 to 1 to avoid division by zero
|
|
fftarray[fftarray == 0] = 1
|
|
dfft = 10.*np.log10(abs(fftarray))
|
|
|
|
# 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 transmittig so our own sending data will not affect this too much
|
|
if not static.TRANSMITTING:
|
|
dfft[dfft>avg+10] = 100
|
|
|
|
# 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
|
|
channel_busy_delay += 5
|
|
# limit delay counter to a maximun of 30. The higher this value, the linger we will wait until releasing state
|
|
if channel_busy_delay > 50:
|
|
channel_busy_delay = 50
|
|
else:
|
|
# decrement channel busy counter if no signal has been detected.
|
|
channel_busy_delay -= 1
|
|
if channel_busy_delay < 0:
|
|
channel_busy_delay = 0
|
|
# if our channel busy counter reached 0, we 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[0:320] #200 --> bandwith 3000
|
|
|
|
except:
|
|
|
|
structlog.get_logger("structlog").debug("[TNC] Setting fft=0")
|
|
# else 0
|
|
static.FFT = [0]
|
|
else:
|
|
pass
|
|
|
|
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 get_bytes_per_frame(mode):
|
|
"""
|
|
provide bytes per frame information for accessing from data handler
|
|
|
|
Args:
|
|
mode:
|
|
|
|
Returns:
|
|
|
|
"""
|
|
if mode == 200:
|
|
freedv = cast(codec2.api.freedv_open_advanced(codec2.api.FREEDV_MODE_FSK_LDPC, ctypes.byref(codec2.api.FREEDV_MODE_FSK_LDPC_0_ADV)), c_void_p)
|
|
elif mode == 201:
|
|
freedv = cast(codec2.api.freedv_open_advanced(codec2.api.FREEDV_MODE_FSK_LDPC, ctypes.byref(codec2.api.FREEDV_MODE_FSK_LDPC_1_ADV)), c_void_p)
|
|
else:
|
|
freedv = cast(codec2.api.freedv_open(mode), c_void_p)
|
|
|
|
# get number of bytes per frame for mode
|
|
return int(codec2.api.freedv_get_bits_per_modem_frame(freedv)/8)
|
|
|
|
|
|
def set_audio_volume(datalist, volume):
|
|
data = np.fromstring(datalist, np.int16) * (volume / 100.)
|
|
return data.astype(np.int16)
|
|
|
|
|