mirror of
https://github.com/DJ2LS/FreeDATA
synced 2024-05-14 08:04:33 +00:00
1562 lines
59 KiB
Python
1562 lines
59 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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# pylint: disable=invalid-name, line-too-long, c-extension-no-member
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# pylint: disable=import-outside-toplevel
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import atexit
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import ctypes
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import os
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import queue
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import threading
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import time
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from collections import deque
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import codec2
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import itertools
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import numpy as np
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import sounddevice as sd
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import structlog
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import tci
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import cw
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from queues import DATA_QUEUE_RECEIVED, MODEM_RECEIVED_QUEUE, MODEM_TRANSMIT_QUEUE, RIGCTLD_COMMAND_QUEUE, \
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AUDIO_RECEIVED_QUEUE, AUDIO_TRANSMIT_QUEUE, MESH_RECEIVED_QUEUE
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import audio
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import event_manager
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from modem_frametypes import FRAME_TYPE
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TESTMODE = False
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RXCHANNEL = ""
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TXCHANNEL = ""
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# Receive only specific modes to reduce CPU load
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RECEIVE_SIG0 = True
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RECEIVE_SIG1 = False
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RECEIVE_DATAC1 = False
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RECEIVE_DATAC3 = False
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RECEIVE_DATAC4 = False
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# state buffer
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SIG0_DATAC13_STATE = []
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SIG1_DATAC13_STATE = []
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DAT0_DATAC1_STATE = []
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DAT0_DATAC3_STATE = []
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DAT0_DATAC4_STATE = []
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FSK_LDPC0_STATE = []
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FSK_LDPC1_STATE = []
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class RF:
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"""Class to encapsulate interactions between the audio device and codec2"""
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log = structlog.get_logger("RF")
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def __init__(self, config, event_queue, fft_queue, service_queue, states) -> None:
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self.config = config
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print(config)
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self.service_queue = service_queue
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self.states = states
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self.sampler_avg = 0
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self.buffer_avg = 0
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# these are crc ids now
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self.audio_input_device = config['AUDIO']['input_device']
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self.audio_output_device = config['AUDIO']['output_device']
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self.rx_audio_level = config['AUDIO']['rx_audio_level']
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self.tx_audio_level = config['AUDIO']['tx_audio_level']
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self.enable_audio_auto_tune = config['AUDIO']['enable_auto_tune']
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self.enable_fsk = config['MODEM']['enable_fsk']
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self.enable_fft = config['MODEM']['enable_fft']
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self.enable_scatter = config['MODEM']['enable_scatter']
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self.tx_delay = config['MODEM']['tx_delay']
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self.tuning_range_fmin = config['MODEM']['tuning_range_fmin']
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self.tuning_range_fmax = config['MODEM']['tuning_range_fmax']
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self.radiocontrol = config['RADIO']['radiocontrol']
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self.rigctld_ip = config['RADIO']['rigctld_ip']
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self.rigctld_port = config['RADIO']['rigctld_port']
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self.states.set("is_transmitting", False)
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self.ptt_state = False
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self.radio_alc = 0.0
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self.tci_ip = config['TCI']['tci_ip']
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self.tci_port = config['TCI']['tci_port']
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self.buffer_overflow_counter = [0, 0, 0, 0, 0, 0, 0, 0]
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self.channel_busy_delay = 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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# 8192 Let's do some tests with very small chunks for TX
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self.AUDIO_FRAMES_PER_BUFFER_TX = 1200 if self.radiocontrol in ["tci"] else 2400 * 2
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# 8 * (self.AUDIO_SAMPLE_RATE_RX/self.MODEM_SAMPLE_RATE) == 48
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self.AUDIO_CHANNELS = 1
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self.MODE = 0
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self.is_codec2_traffic_cooldown = 20
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self.is_codec2_traffic_counter = 0
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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 # type: ignore
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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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self.audio_received_queue = AUDIO_RECEIVED_QUEUE
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self.audio_transmit_queue = AUDIO_TRANSMIT_QUEUE
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# Init FIFO queue to store modulation out in
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self.modoutqueue = deque()
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self.event_manager = event_manager.EventManager([event_queue])
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self.fft_queue = fft_queue
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self.start_modem()
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# --------------------------------------------------------------------------------------------------------
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def tci_tx_callback(self) -> None:
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"""
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Callback for TCI TX
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"""
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while True:
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threading.Event().wait(0.01)
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if len(self.modoutqueue) > 0 and not self.mod_out_locked:
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self.radio.set_ptt(True)
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self.event_manager.send_ptt_change(True)
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data_out = self.modoutqueue.popleft()
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self.tci_module.push_audio(data_out)
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def start_modem(self):
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if not TESTMODE and self.radiocontrol not in ["tci"]:
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result = self.init_audio()
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elif not TESTMODE:
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result = self.init_tci()
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else:
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result = self.init_mkfifo()
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if result not in [False]:
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# init codec2 instances
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self.init_codec2()
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# init rig control
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self.init_rig_control()
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# init decoders
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self.init_decoders()
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# init decoding threads
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self.init_data_threads()
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atexit.register(self.stream.stop)
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else:
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return False
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def stop_modem(self):
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try:
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# let's stop the modem service
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self.service_queue.put("stop")
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# simulate audio class active state for reducing cli output
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# self.stream = lambda: None
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# self.stream.active = False
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# self.stream.stop
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except Exception:
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self.log.error("[MDM] Error stopping modem")
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def init_audio(self):
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self.log.info(f"[MDM] init: get audio devices", input_device=self.audio_input_device,
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output_device=self.audio_output_device)
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try:
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result = audio.get_device_index_from_crc(self.audio_input_device, True)
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if result is None:
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raise ValueError("Invalid input device")
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else:
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in_dev_index, in_dev_name = result
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result = audio.get_device_index_from_crc(self.audio_output_device, False)
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if result is None:
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raise ValueError("Invalid output device")
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else:
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out_dev_index, out_dev_name = result
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self.log.info(f"[MDM] init: receiving audio from '{in_dev_name}'")
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self.log.info(f"[MDM] init: transmiting audio on '{out_dev_name}'")
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self.log.debug("[MDM] init: starting pyaudio callback and decoding threads")
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# init codec2 resampler
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self.resampler = codec2.resampler()
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# init audio stream
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self.stream = sd.RawStream(
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channels=1,
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dtype="int16",
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callback=self.callback,
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device=(in_dev_index, out_dev_index),
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samplerate=self.AUDIO_SAMPLE_RATE_RX,
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blocksize=4800,
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)
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self.stream.start()
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return True
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except Exception as audioerr:
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self.log.error("[MDM] init: starting pyaudio callback failed", e=audioerr)
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self.stop_modem()
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return False
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def init_tci(self):
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# placeholder area for processing audio via TCI
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# https://github.com/maksimus1210/TCI
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self.log.warning("[MDM] [TCI] Not yet fully implemented", ip=self.tci_ip, port=self.tci_port)
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# we are trying this by simulating an audio stream Object like with mkfifo
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class Object:
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"""An object for simulating audio stream"""
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active = True
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self.stream = Object()
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# lets init TCI module
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self.tci_module = tci.TCICtrl()
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tci_rx_callback_thread = threading.Thread(
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target=self.tci_rx_callback,
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name="TCI RX CALLBACK THREAD",
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daemon=True,
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)
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tci_rx_callback_thread.start()
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# let's start the audio tx callback
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self.log.debug("[MDM] Starting tci tx callback thread")
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tci_tx_callback_thread = threading.Thread(
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target=self.tci_tx_callback,
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name="TCI TX CALLBACK THREAD",
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daemon=True,
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)
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tci_tx_callback_thread.start()
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def init_mkfifo(self):
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class Object:
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"""An object for simulating audio stream"""
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active = True
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self.stream = Object()
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# Create mkfifo buffers
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try:
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os.mkfifo(RXCHANNEL)
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os.mkfifo(TXCHANNEL)
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except Exception as err:
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self.log.info(f"[MDM] init:mkfifo: Exception: {err}")
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mkfifo_write_callback_thread = threading.Thread(
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target=self.mkfifo_write_callback,
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name="MKFIFO WRITE CALLBACK THREAD",
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daemon=True,
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)
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mkfifo_write_callback_thread.start()
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self.log.debug("[MDM] Starting mkfifo_read_callback")
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mkfifo_read_callback_thread = threading.Thread(
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target=self.mkfifo_read_callback,
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name="MKFIFO READ CALLBACK THREAD",
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daemon=True,
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)
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mkfifo_read_callback_thread.start()
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def tci_rx_callback(self) -> None:
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"""
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Callback for TCI RX
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data_in48k must be filled with 48000Hz audio raw data
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"""
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while True:
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x = self.audio_received_queue.get()
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x = np.frombuffer(x, dtype=np.int16)
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# x = self.resampler.resample48_to_8(x)
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if self.enable_fft:
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self.calculate_fft(x)
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length_x = len(x)
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for data_buffer, receive in [
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(self.sig0_datac13_buffer, RECEIVE_SIG0),
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(self.sig1_datac13_buffer, RECEIVE_SIG1),
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(self.dat0_datac1_buffer, RECEIVE_DATAC1),
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(self.dat0_datac3_buffer, RECEIVE_DATAC3),
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(self.dat0_datac4_buffer, RECEIVE_DATAC4),
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(self.fsk_ldpc_buffer_0, self.enable_fsk),
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(self.fsk_ldpc_buffer_1, self.enable_fsk),
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]:
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if (
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not (data_buffer.nbuffer + length_x) > data_buffer.size
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and receive
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):
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data_buffer.push(x)
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def mkfifo_read_callback(self) -> None:
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"""
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Support testing by reading the audio data from a pipe and
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depositing the data into the codec data buffers.
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"""
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while True:
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threading.Event().wait(0.01)
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# -----read
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data_in48k = bytes()
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with open(RXCHANNEL, "rb") as fifo:
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for line in fifo:
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data_in48k += line
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while len(data_in48k) >= 48:
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x = np.frombuffer(data_in48k[:48], dtype=np.int16)
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x = self.resampler.resample48_to_8(x)
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data_in48k = data_in48k[48:]
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length_x = len(x)
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for data_buffer, receive in [
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(self.sig0_datac13_buffer, RECEIVE_SIG0),
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(self.sig1_datac13_buffer, RECEIVE_SIG1),
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(self.dat0_datac1_buffer, RECEIVE_DATAC1),
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(self.dat0_datac3_buffer, RECEIVE_DATAC3),
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(self.dat0_datac4_buffer, RECEIVE_DATAC4),
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(self.fsk_ldpc_buffer_0, self.enable_fsk),
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(self.fsk_ldpc_buffer_1, self.enable_fsk),
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]:
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if (
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not (data_buffer.nbuffer + length_x) > data_buffer.size
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and receive
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):
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data_buffer.push(x)
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def mkfifo_write_callback(self) -> None:
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"""Support testing by writing the audio data to a pipe."""
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while True:
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threading.Event().wait(0.01)
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# -----write
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if len(self.modoutqueue) > 0 and not self.mod_out_locked:
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data_out48k = self.modoutqueue.popleft()
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# print(len(data_out48k))
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with open(TXCHANNEL, "wb") as fifo_write:
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fifo_write.write(data_out48k)
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fifo_write.flush()
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fifo_write.flush()
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# Callback for the audio streaming devices
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def callback(self, data_in48k, outdata, frames, time, status) -> None:
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"""
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Receive data into appropriate queue.
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Args:
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data_in48k: Incoming data received
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outdata: Container for the data returned
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frames: Number of frames
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time:
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status:
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"""
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# self.log.debug("[MDM] callback")
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try:
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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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x = set_audio_volume(x, self.rx_audio_level)
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# audio recording for debugging purposes
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# TODO Find a nice place for this
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#if AudioParam.audio_record:
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# AudioParam.audio_record_file.writeframes(x)
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# Avoid decoding when transmitting to reduce CPU
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# TODO Overriding this for testing purposes
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# if not self.states.is_transmitting:
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length_x = len(x)
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# Avoid buffer overflow by filling only if buffer for
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# selected datachannel mode is not full
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for audiobuffer, receive, index in [
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(self.sig0_datac13_buffer, RECEIVE_SIG0, 0),
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(self.sig1_datac13_buffer, RECEIVE_SIG1, 1),
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(self.dat0_datac1_buffer, RECEIVE_DATAC1, 2),
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(self.dat0_datac3_buffer, RECEIVE_DATAC3, 3),
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(self.dat0_datac4_buffer, RECEIVE_DATAC4, 4),
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(self.fsk_ldpc_buffer_0, self.enable_fsk, 5),
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(self.fsk_ldpc_buffer_1, self.enable_fsk, 6),
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]:
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if (audiobuffer.nbuffer + length_x) > audiobuffer.size:
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self.buffer_overflow_counter[index] += 1
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self.event_manager.send_buffer_overflow(self.buffer_overflow_counter)
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elif receive:
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audiobuffer.push(x)
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# end of "not self.states.is_transmitting" if block
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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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if self.enable_fft:
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self.calculate_fft(x)
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else:
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# TODO Moved to this place for testing
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# Maybe we can avoid moments of silence before transmitting
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self.radio.set_ptt(True)
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self.event_manager.send_ptt_change(True)
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data_out48k = self.modoutqueue.popleft()
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if self.enable_fft:
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self.calculate_fft(data_out48k)
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except Exception as e:
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self.log.warning(f"[MDM] audio callback not ready yet: {e}")
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try:
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outdata[:] = data_out48k[:frames]
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except IndexError as err:
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self.log.debug(f"[MDM] callback writing error: IndexError: {err}")
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# return (data_out48k, audio.pyaudio.paContinue)
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# --------------------------------------------------------------------
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def transmit(
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self, mode, repeats: int, repeat_delay: int, frames: bytearray
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) -> bool:
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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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"""
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self.reset_data_sync()
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if mode == codec2.FREEDV_MODE.datac0.value:
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freedv = self.freedv_datac0_tx
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elif mode == codec2.FREEDV_MODE.datac1.value:
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freedv = self.freedv_datac1_tx
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elif mode == codec2.FREEDV_MODE.datac3.value:
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freedv = self.freedv_datac3_tx
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elif mode == codec2.FREEDV_MODE.datac4.value:
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freedv = self.freedv_datac4_tx
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elif mode == codec2.FREEDV_MODE.datac13.value:
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freedv = self.freedv_datac13_tx
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elif mode == codec2.FREEDV_MODE.fsk_ldpc_0.value:
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freedv = self.freedv_ldpc0_tx
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elif mode == codec2.FREEDV_MODE.fsk_ldpc_1.value:
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freedv = self.freedv_ldpc1_tx
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else:
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return False
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self.states.set("is_transmitting", True)
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# if we're transmitting FreeDATA signals, reset channel busy state
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self.states.set("channel_busy", False)
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start_of_transmission = time.time()
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# TODO Moved ptt toggle some steps before audio is ready for testing
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# Toggle ptt early to save some time and send ptt state via socket
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# self.radio.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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# 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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# 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
|
|
if self.tx_delay > 0:
|
|
data_delay = int(self.MODEM_SAMPLE_RATE * (self.tx_delay / 1000)) # type: ignore
|
|
mod_out_silence = ctypes.create_string_buffer(data_delay * 2)
|
|
txbuffer = bytes(mod_out_silence)
|
|
else:
|
|
txbuffer = bytes()
|
|
|
|
self.log.debug(
|
|
"[MDM] TRANSMIT", mode=self.MODE, payload=payload_bytes_per_frame, delay=self.tx_delay
|
|
)
|
|
|
|
for _ in range(repeats):
|
|
|
|
# Create modulation for all frames in the list
|
|
for frame in frames:
|
|
# Write preamble to txbuffer
|
|
# codec2 fsk preamble may be broken -
|
|
# at least it sounds like that, so we are disabling it for testing
|
|
if self.MODE not in [
|
|
codec2.FREEDV_MODE.fsk_ldpc_0.value,
|
|
codec2.FREEDV_MODE.fsk_ldpc_1.value,
|
|
]:
|
|
# Write preamble to txbuffer
|
|
codec2.api.freedv_rawdatapreambletx(freedv, mod_out_preamble)
|
|
txbuffer += bytes(mod_out_preamble)
|
|
|
|
# Create buffer for data
|
|
# Use this if CRC16 checksum is required (DATAc1-3)
|
|
buffer = bytearray(payload_bytes_per_frame)
|
|
# Set buffersize to length of data which will be send
|
|
buffer[: len(frame)] = frame # type: ignore
|
|
|
|
# Create crc for data frame -
|
|
# Use the crc function shipped with codec2
|
|
# to avoid CRC algorithm incompatibilities
|
|
# Generate CRC16
|
|
crc = ctypes.c_ushort(
|
|
codec2.api.freedv_gen_crc16(bytes(buffer), payload_bytes_per_frame)
|
|
)
|
|
# Convert crc to 2-byte (16-bit) hex string
|
|
crc = crc.value.to_bytes(2, byteorder="big")
|
|
# Append CRC to data buffer
|
|
buffer += crc
|
|
|
|
data = (ctypes.c_ubyte * bytes_per_frame).from_buffer_copy(buffer)
|
|
# modulate DATA and save it into mod_out pointer
|
|
codec2.api.freedv_rawdatatx(freedv, mod_out, data)
|
|
txbuffer += bytes(mod_out)
|
|
|
|
# codec2 fsk postamble may be broken -
|
|
# at least it sounds like that, so we are disabling it for testing
|
|
if self.MODE not in [
|
|
codec2.FREEDV_MODE.fsk_ldpc_0.value,
|
|
codec2.FREEDV_MODE.fsk_ldpc_1.value,
|
|
]:
|
|
# Write postamble to txbuffer
|
|
codec2.api.freedv_rawdatapostambletx(freedv, mod_out_postamble)
|
|
# Append postamble to txbuffer
|
|
txbuffer += bytes(mod_out_postamble)
|
|
|
|
# Add delay to end of frames
|
|
samples_delay = int(self.MODEM_SAMPLE_RATE * (repeat_delay / 1000)) # type: ignore
|
|
mod_out_silence = ctypes.create_string_buffer(samples_delay * 2)
|
|
txbuffer += bytes(mod_out_silence)
|
|
|
|
# Re-sample back up to 48k (resampler works on np.int16)
|
|
x = np.frombuffer(txbuffer, dtype=np.int16)
|
|
|
|
self.audio_auto_tune()
|
|
x = set_audio_volume(x, self.tx_audio_level)
|
|
|
|
if not self.radiocontrol in ["tci"]:
|
|
txbuffer_out = self.resampler.resample8_to_48(x)
|
|
else:
|
|
txbuffer_out = x
|
|
|
|
# Explicitly lock our usage of mod_out_queue if needed
|
|
# This could avoid audio problems on slower CPU
|
|
# we will fill our modout list with all data, then start
|
|
# processing it in audio callback
|
|
self.mod_out_locked = True
|
|
|
|
# -------------------------------
|
|
# add modulation to modout_queue
|
|
self.enqueue_modulation(txbuffer_out)
|
|
|
|
# Release our mod_out_lock, so we can use the queue
|
|
self.mod_out_locked = False
|
|
|
|
# we need to wait manually for tci processing
|
|
if self.radiocontrol in ["tci"]:
|
|
duration = len(txbuffer_out) / 8000
|
|
timestamp_to_sleep = time.time() + duration
|
|
self.log.debug("[MDM] TCI calculated duration", duration=duration)
|
|
tci_timeout_reached = False
|
|
#while time.time() < timestamp_to_sleep:
|
|
# threading.Event().wait(0.01)
|
|
else:
|
|
timestamp_to_sleep = time.time()
|
|
# set tci timeout reached to True for overriding if not used
|
|
tci_timeout_reached = True
|
|
|
|
while self.modoutqueue or not tci_timeout_reached:
|
|
if self.radiocontrol in ["tci"]:
|
|
if time.time() < timestamp_to_sleep:
|
|
tci_timeout_reached = False
|
|
else:
|
|
tci_timeout_reached = True
|
|
threading.Event().wait(0.01)
|
|
# if we're transmitting FreeDATA signals, reset channel busy state
|
|
self.states.set("channel_busy", False)
|
|
|
|
self.radio.set_ptt(False)
|
|
|
|
# Push ptt state to socket stream
|
|
self.event_manager.send_ptt_change(False)
|
|
|
|
# After processing, set the locking state back to true to be prepared for next transmission
|
|
self.mod_out_locked = True
|
|
|
|
self.modem_transmit_queue.task_done()
|
|
self.states.set("is_transmitting", False)
|
|
|
|
end_of_transmission = time.time()
|
|
transmission_time = end_of_transmission - start_of_transmission
|
|
self.log.debug("[MDM] ON AIR TIME", time=transmission_time)
|
|
|
|
def audio_auto_tune(self):
|
|
# enable / disable AUDIO TUNE Feature / ALC correction
|
|
if self.enable_audio_auto_tune:
|
|
if self.radio_alc == 0.0:
|
|
self.tx_audio_level = self.tx_audio_level + 20
|
|
elif 0.0 < self.radio_alc <= 0.1:
|
|
print("0.0 < self.radio_alc <= 0.1")
|
|
self.tx_audio_level = self.tx_audio_level + 2
|
|
self.log.debug("[MDM] AUDIO TUNE", audio_level=str(self.tx_audio_level),
|
|
alc_level=str(self.radio_alc))
|
|
elif 0.1 < self.radio_alc < 0.2:
|
|
print("0.1 < self.radio_alc < 0.2")
|
|
self.tx_audio_level = self.tx_audio_level
|
|
self.log.debug("[MDM] AUDIO TUNE", audio_level=str(self.tx_audio_level),
|
|
alc_level=str(self.radio_alc))
|
|
elif 0.2 < self.radio_alc < 0.99:
|
|
print("0.2 < self.radio_alc < 0.99")
|
|
self.tx_audio_level = self.tx_audio_level - 20
|
|
self.log.debug("[MDM] AUDIO TUNE", audio_level=str(self.tx_audio_level),
|
|
alc_level=str(self.radio_alc))
|
|
elif 1.0 >= self.radio_alc:
|
|
print("1.0 >= self.radio_alc")
|
|
self.tx_audio_level = self.tx_audio_level - 40
|
|
self.log.debug("[MDM] AUDIO TUNE", audio_level=str(self.tx_audio_level),
|
|
alc_level=str(self.radio_alc))
|
|
else:
|
|
self.log.debug("[MDM] AUDIO TUNE", audio_level=str(self.tx_audio_level),
|
|
alc_level=str(self.radio_alc))
|
|
|
|
def transmit_morse(self, repeats, repeat_delay, frames):
|
|
self.states.set("is_transmitting", True)
|
|
# if we're transmitting FreeDATA signals, reset channel busy state
|
|
self.states.set("channel_busy", False)
|
|
self.log.debug(
|
|
"[MDM] TRANSMIT", mode="MORSE"
|
|
)
|
|
start_of_transmission = time.time()
|
|
|
|
txbuffer_out = cw.MorseCodePlayer().text_to_signal("DJ2LS-1")
|
|
|
|
self.mod_out_locked = True
|
|
self.enqueue_modulation(txbuffer_out)
|
|
self.mod_out_locked = False
|
|
|
|
# we need to wait manually for tci processing
|
|
if self.radiocontrol in ["tci"]:
|
|
duration = len(txbuffer_out) / 8000
|
|
timestamp_to_sleep = time.time() + duration
|
|
self.log.debug("[MDM] TCI calculated duration", duration=duration)
|
|
tci_timeout_reached = False
|
|
#while time.time() < timestamp_to_sleep:
|
|
# threading.Event().wait(0.01)
|
|
else:
|
|
timestamp_to_sleep = time.time()
|
|
# set tci timeout reached to True for overriding if not used
|
|
tci_timeout_reached = True
|
|
|
|
while self.modoutqueue or not tci_timeout_reached:
|
|
if self.radiocontrol in ["tci"]:
|
|
if time.time() < timestamp_to_sleep:
|
|
tci_timeout_reached = False
|
|
else:
|
|
tci_timeout_reached = True
|
|
|
|
threading.Event().wait(0.01)
|
|
# if we're transmitting FreeDATA signals, reset channel busy state
|
|
self.states.set("channel_busy", False)
|
|
|
|
self.radio.set_ptt(False)
|
|
|
|
# Push ptt state to socket stream
|
|
self.event_manager.send_ptt_change(False)
|
|
|
|
# After processing, set the locking state back to true to be prepared for next transmission
|
|
self.mod_out_locked = True
|
|
|
|
self.modem_transmit_queue.task_done()
|
|
self.states.set("is_transmitting", False)
|
|
threading.Event().set()
|
|
|
|
end_of_transmission = time.time()
|
|
transmission_time = end_of_transmission - start_of_transmission
|
|
self.log.debug("[MDM] ON AIR TIME", time=transmission_time)
|
|
|
|
def enqueue_modulation(self, txbuffer_out):
|
|
|
|
|
|
chunk_length = self.AUDIO_FRAMES_PER_BUFFER_TX # 4800
|
|
chunk = [
|
|
txbuffer_out[i: i + chunk_length]
|
|
for i in range(0, len(txbuffer_out), chunk_length)
|
|
]
|
|
for c in chunk:
|
|
# Pad the chunk, if needed
|
|
if len(c) < chunk_length:
|
|
delta = chunk_length - len(c)
|
|
delta_zeros = np.zeros(delta, dtype=np.int16)
|
|
c = np.append(c, delta_zeros)
|
|
# self.log.debug("[MDM] mod out shorter than audio buffer", delta=delta)
|
|
self.modoutqueue.append(c)
|
|
|
|
def init_decoders(self):
|
|
|
|
if self.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()
|
|
|
|
else:
|
|
audio_thread_sig0_datac13 = threading.Thread(
|
|
target=self.audio_sig0_datac13, name="AUDIO_THREAD DATAC13 - 0", daemon=True
|
|
)
|
|
audio_thread_sig0_datac13.start()
|
|
|
|
audio_thread_sig1_datac13 = threading.Thread(
|
|
target=self.audio_sig1_datac13, name="AUDIO_THREAD DATAC13 - 1", daemon=True
|
|
)
|
|
audio_thread_sig1_datac13.start()
|
|
|
|
audio_thread_dat0_datac1 = threading.Thread(
|
|
target=self.audio_dat0_datac1, name="AUDIO_THREAD DATAC1", daemon=True
|
|
)
|
|
audio_thread_dat0_datac1.start()
|
|
|
|
audio_thread_dat0_datac3 = threading.Thread(
|
|
target=self.audio_dat0_datac3, name="AUDIO_THREAD DATAC3", daemon=True
|
|
)
|
|
audio_thread_dat0_datac3.start()
|
|
|
|
audio_thread_dat0_datac4 = threading.Thread(
|
|
target=self.audio_dat0_datac4, name="AUDIO_THREAD DATAC4", daemon=True
|
|
)
|
|
audio_thread_dat0_datac4.start()
|
|
|
|
def demodulate_audio(
|
|
self,
|
|
audiobuffer: codec2.audio_buffer,
|
|
nin: int,
|
|
freedv: ctypes.c_void_p,
|
|
bytes_out,
|
|
bytes_per_frame,
|
|
state_buffer,
|
|
mode_name,
|
|
) -> int:
|
|
"""
|
|
De-modulate supplied audio stream with supplied codec2 instance.
|
|
Decoded audio is placed into `bytes_out`.
|
|
|
|
:param audiobuffer: Incoming audio
|
|
:type audiobuffer: codec2.audio_buffer
|
|
:param nin: Number of frames codec2 is expecting
|
|
:type nin: int
|
|
:param freedv: codec2 instance
|
|
:type freedv: ctypes.c_void_p
|
|
:param bytes_out: Demodulated audio
|
|
:type bytes_out: _type_
|
|
:param bytes_per_frame: Number of bytes per frame
|
|
:type bytes_per_frame: int
|
|
:param state_buffer: modem states
|
|
:type state_buffer: int
|
|
:param mode_name: mode name
|
|
:type mode_name: str
|
|
:return: NIN from freedv instance
|
|
:rtype: int
|
|
"""
|
|
|
|
nbytes = 0
|
|
try:
|
|
while self.stream.active:
|
|
threading.Event().wait(0.01)
|
|
while audiobuffer.nbuffer >= nin:
|
|
# demodulate audio
|
|
nbytes = codec2.api.freedv_rawdatarx(
|
|
freedv, bytes_out, audiobuffer.buffer.ctypes
|
|
)
|
|
# get current modem states and write to list
|
|
# 1 trial
|
|
# 2 sync
|
|
# 3 trial sync
|
|
# 6 decoded
|
|
# 10 error decoding == NACK
|
|
rx_status = codec2.api.freedv_get_rx_status(freedv)
|
|
|
|
if rx_status not in [0]:
|
|
# we need to disable this if in testmode as its causing problems with FIFO it seems
|
|
if not TESTMODE:
|
|
self.states.set("is_codec2_traffic", True)
|
|
self.is_codec2_traffic_counter = self.is_codec2_traffic_cooldown
|
|
if not self.states.channel_busy:
|
|
self.log.debug("[MDM] Setting channel_busy since codec2 data detected")
|
|
self.states.set("channel_busy", True)
|
|
self.channel_busy_delay += 10
|
|
self.log.debug(
|
|
"[MDM] [demod_audio] modem state", mode=mode_name, rx_status=rx_status,
|
|
sync_flag=codec2.api.rx_sync_flags_to_text[rx_status]
|
|
)
|
|
else:
|
|
self.states.set("is_codec2_traffic", False)
|
|
|
|
# decrement codec traffic counter for making state smoother
|
|
if self.is_codec2_traffic_counter > 0:
|
|
self.is_codec2_traffic_counter -= 1
|
|
self.states.set("is_codec2_traffic", True)
|
|
else:
|
|
self.states.set("is_codec2_traffic", False)
|
|
|
|
if rx_status == 10:
|
|
state_buffer.append(rx_status)
|
|
|
|
audiobuffer.pop(nin)
|
|
nin = codec2.api.freedv_nin(freedv)
|
|
if nbytes == bytes_per_frame:
|
|
print(bytes(bytes_out))
|
|
|
|
# ignore data channel opener frames for avoiding toggle states
|
|
# use case: opener already received, but ack got lost and we are receiving
|
|
# an opener again
|
|
if mode_name in ["sig1-datac13"] and int.from_bytes(bytes(bytes_out[:1]), "big") in [
|
|
FRAME_TYPE.ARQ_SESSION_OPEN.value,
|
|
FRAME_TYPE.ARQ_DC_OPEN_W.value,
|
|
FRAME_TYPE.ARQ_DC_OPEN_ACK_W.value,
|
|
FRAME_TYPE.ARQ_DC_OPEN_N.value,
|
|
FRAME_TYPE.ARQ_DC_OPEN_ACK_N.value
|
|
]:
|
|
print("dropp")
|
|
elif int.from_bytes(bytes(bytes_out[:1]), "big") in [
|
|
FRAME_TYPE.MESH_BROADCAST.value,
|
|
FRAME_TYPE.MESH_SIGNALLING_PING.value,
|
|
FRAME_TYPE.MESH_SIGNALLING_PING_ACK.value,
|
|
]:
|
|
self.log.debug(
|
|
"[MDM] [demod_audio] moving data to mesh dispatcher", nbytes=nbytes
|
|
)
|
|
MESH_RECEIVED_QUEUE.put([bytes(bytes_out), snr])
|
|
|
|
else:
|
|
self.log.debug(
|
|
"[MDM] [demod_audio] Pushing received data to received_queue", nbytes=nbytes
|
|
)
|
|
snr = self.calculate_snr(freedv)
|
|
self.modem_received_queue.put([bytes_out, freedv, bytes_per_frame, snr])
|
|
self.get_scatter(freedv)
|
|
state_buffer = []
|
|
|
|
except Exception as e:
|
|
self.log.warning("[MDM] [demod_audio] Stream not active anymore", e=e)
|
|
|
|
return nin
|
|
|
|
def init_codec2(self):
|
|
# Open codec2 instances
|
|
|
|
# DATAC13
|
|
# SIGNALLING MODE 0 - Used for Connecting - Payload 14 Bytes
|
|
self.sig0_datac13_freedv, \
|
|
self.sig0_datac13_bytes_per_frame, \
|
|
self.sig0_datac13_bytes_out, \
|
|
self.sig0_datac13_buffer, \
|
|
self.sig0_datac13_nin = \
|
|
self.init_codec2_mode(codec2.FREEDV_MODE.datac13.value, None)
|
|
|
|
# DATAC13
|
|
# SIGNALLING MODE 1 - Used for ACK/NACK - Payload 5 Bytes
|
|
self.sig1_datac13_freedv, \
|
|
self.sig1_datac13_bytes_per_frame, \
|
|
self.sig1_datac13_bytes_out, \
|
|
self.sig1_datac13_buffer, \
|
|
self.sig1_datac13_nin = \
|
|
self.init_codec2_mode(codec2.FREEDV_MODE.datac13.value, None)
|
|
|
|
# DATAC1
|
|
self.dat0_datac1_freedv, \
|
|
self.dat0_datac1_bytes_per_frame, \
|
|
self.dat0_datac1_bytes_out, \
|
|
self.dat0_datac1_buffer, \
|
|
self.dat0_datac1_nin = \
|
|
self.init_codec2_mode(codec2.FREEDV_MODE.datac1.value, None)
|
|
|
|
# DATAC3
|
|
self.dat0_datac3_freedv, \
|
|
self.dat0_datac3_bytes_per_frame, \
|
|
self.dat0_datac3_bytes_out, \
|
|
self.dat0_datac3_buffer, \
|
|
self.dat0_datac3_nin = \
|
|
self.init_codec2_mode(codec2.FREEDV_MODE.datac3.value, None)
|
|
|
|
# DATAC4
|
|
self.dat0_datac4_freedv, \
|
|
self.dat0_datac4_bytes_per_frame, \
|
|
self.dat0_datac4_bytes_out, \
|
|
self.dat0_datac4_buffer, \
|
|
self.dat0_datac4_nin = \
|
|
self.init_codec2_mode(codec2.FREEDV_MODE.datac4.value, None)
|
|
|
|
|
|
# FSK LDPC - 0
|
|
self.fsk_ldpc_freedv_0, \
|
|
self.fsk_ldpc_bytes_per_frame_0, \
|
|
self.fsk_ldpc_bytes_out_0, \
|
|
self.fsk_ldpc_buffer_0, \
|
|
self.fsk_ldpc_nin_0 = \
|
|
self.init_codec2_mode(
|
|
codec2.FREEDV_MODE.fsk_ldpc.value,
|
|
codec2.api.FREEDV_MODE_FSK_LDPC_0_ADV
|
|
)
|
|
|
|
# FSK LDPC - 1
|
|
self.fsk_ldpc_freedv_1, \
|
|
self.fsk_ldpc_bytes_per_frame_1, \
|
|
self.fsk_ldpc_bytes_out_1, \
|
|
self.fsk_ldpc_buffer_1, \
|
|
self.fsk_ldpc_nin_1 = \
|
|
self.init_codec2_mode(
|
|
codec2.FREEDV_MODE.fsk_ldpc.value,
|
|
codec2.api.FREEDV_MODE_FSK_LDPC_1_ADV
|
|
)
|
|
|
|
# INIT TX MODES - here we need all modes.
|
|
self.freedv_datac0_tx = open_codec2_instance(codec2.FREEDV_MODE.datac0.value)
|
|
self.freedv_datac1_tx = open_codec2_instance(codec2.FREEDV_MODE.datac1.value)
|
|
self.freedv_datac3_tx = open_codec2_instance(codec2.FREEDV_MODE.datac3.value)
|
|
self.freedv_datac4_tx = open_codec2_instance(codec2.FREEDV_MODE.datac4.value)
|
|
self.freedv_datac13_tx = open_codec2_instance(codec2.FREEDV_MODE.datac13.value)
|
|
self.freedv_ldpc0_tx = open_codec2_instance(codec2.FREEDV_MODE.fsk_ldpc_0.value)
|
|
self.freedv_ldpc1_tx = open_codec2_instance(codec2.FREEDV_MODE.fsk_ldpc_1.value)
|
|
|
|
def init_codec2_mode(self, mode, adv):
|
|
"""
|
|
Init codec2 and return some important parameters
|
|
|
|
Args:
|
|
self:
|
|
mode:
|
|
adv:
|
|
|
|
Returns:
|
|
c2instance, bytes_per_frame, bytes_out, audio_buffer, nin
|
|
"""
|
|
if adv:
|
|
# FSK Long-distance Parity Code 1 - data frames
|
|
c2instance = ctypes.cast(
|
|
codec2.api.freedv_open_advanced(
|
|
codec2.FREEDV_MODE.fsk_ldpc.value,
|
|
ctypes.byref(adv),
|
|
),
|
|
ctypes.c_void_p,
|
|
)
|
|
else:
|
|
|
|
# create codec2 instance
|
|
c2instance = ctypes.cast(
|
|
codec2.api.freedv_open(mode), ctypes.c_void_p
|
|
)
|
|
|
|
# set tuning range
|
|
codec2.api.freedv_set_tuning_range(
|
|
c2instance,
|
|
ctypes.c_float(float(self.tuning_range_fmin)),
|
|
ctypes.c_float(float(self.tuning_range_fmax)),
|
|
)
|
|
|
|
# get bytes per frame
|
|
bytes_per_frame = int(
|
|
codec2.api.freedv_get_bits_per_modem_frame(c2instance) / 8
|
|
)
|
|
|
|
# create byte out buffer
|
|
bytes_out = ctypes.create_string_buffer(bytes_per_frame)
|
|
|
|
# set initial frames per burst
|
|
codec2.api.freedv_set_frames_per_burst(c2instance, 1)
|
|
|
|
# init audio buffer
|
|
audio_buffer = codec2.audio_buffer(2 * self.AUDIO_FRAMES_PER_BUFFER_RX)
|
|
|
|
# get initial nin
|
|
nin = codec2.api.freedv_nin(c2instance)
|
|
|
|
# Additional Datac0-specific information - these are not referenced anywhere else.
|
|
# self.sig0_datac0_payload_per_frame = self.sig0_datac0_bytes_per_frame - 2
|
|
# self.sig0_datac0_n_nom_modem_samples = codec2.api.freedv_get_n_nom_modem_samples(
|
|
# self.sig0_datac0_freedv
|
|
# )
|
|
# self.sig0_datac0_n_tx_modem_samples = codec2.api.freedv_get_n_tx_modem_samples(
|
|
# self.sig0_datac0_freedv
|
|
# )
|
|
# self.sig0_datac0_n_tx_preamble_modem_samples = (
|
|
# codec2.api.freedv_get_n_tx_preamble_modem_samples(self.sig0_datac0_freedv)
|
|
# )
|
|
# self.sig0_datac0_n_tx_postamble_modem_samples = (
|
|
# codec2.api.freedv_get_n_tx_postamble_modem_samples(self.sig0_datac0_freedv)
|
|
# )
|
|
|
|
# return values
|
|
return c2instance, bytes_per_frame, bytes_out, audio_buffer, nin
|
|
|
|
def audio_sig0_datac13(self) -> None:
|
|
"""Receive data encoded with datac13 - 0"""
|
|
self.sig0_datac13_nin = self.demodulate_audio(
|
|
self.sig0_datac13_buffer,
|
|
self.sig0_datac13_nin,
|
|
self.sig0_datac13_freedv,
|
|
self.sig0_datac13_bytes_out,
|
|
self.sig0_datac13_bytes_per_frame,
|
|
SIG0_DATAC13_STATE,
|
|
"sig0-datac13"
|
|
)
|
|
|
|
def audio_sig1_datac13(self) -> None:
|
|
"""Receive data encoded with datac13 - 1"""
|
|
self.sig1_datac13_nin = self.demodulate_audio(
|
|
self.sig1_datac13_buffer,
|
|
self.sig1_datac13_nin,
|
|
self.sig1_datac13_freedv,
|
|
self.sig1_datac13_bytes_out,
|
|
self.sig1_datac13_bytes_per_frame,
|
|
SIG1_DATAC13_STATE,
|
|
"sig1-datac13"
|
|
)
|
|
|
|
def audio_dat0_datac4(self) -> None:
|
|
"""Receive data encoded with datac4"""
|
|
self.dat0_datac4_nin = self.demodulate_audio(
|
|
self.dat0_datac4_buffer,
|
|
self.dat0_datac4_nin,
|
|
self.dat0_datac4_freedv,
|
|
self.dat0_datac4_bytes_out,
|
|
self.dat0_datac4_bytes_per_frame,
|
|
DAT0_DATAC4_STATE,
|
|
"dat0-datac4"
|
|
)
|
|
|
|
def audio_dat0_datac1(self) -> None:
|
|
"""Receive data encoded with datac1"""
|
|
self.dat0_datac1_nin = self.demodulate_audio(
|
|
self.dat0_datac1_buffer,
|
|
self.dat0_datac1_nin,
|
|
self.dat0_datac1_freedv,
|
|
self.dat0_datac1_bytes_out,
|
|
self.dat0_datac1_bytes_per_frame,
|
|
DAT0_DATAC1_STATE,
|
|
"dat0-datac1"
|
|
)
|
|
|
|
def audio_dat0_datac3(self) -> None:
|
|
"""Receive data encoded with datac3"""
|
|
self.dat0_datac3_nin = self.demodulate_audio(
|
|
self.dat0_datac3_buffer,
|
|
self.dat0_datac3_nin,
|
|
self.dat0_datac3_freedv,
|
|
self.dat0_datac3_bytes_out,
|
|
self.dat0_datac3_bytes_per_frame,
|
|
DAT0_DATAC3_STATE,
|
|
"dat0-datac3"
|
|
)
|
|
|
|
def audio_fsk_ldpc_0(self) -> None:
|
|
"""Receive data encoded with FSK + LDPC0"""
|
|
self.fsk_ldpc_nin_0 = self.demodulate_audio(
|
|
self.fsk_ldpc_buffer_0,
|
|
self.fsk_ldpc_nin_0,
|
|
self.fsk_ldpc_freedv_0,
|
|
self.fsk_ldpc_bytes_out_0,
|
|
self.fsk_ldpc_bytes_per_frame_0,
|
|
FSK_LDPC0_STATE,
|
|
"fsk_ldpc0",
|
|
)
|
|
|
|
def audio_fsk_ldpc_1(self) -> None:
|
|
"""Receive data encoded with FSK + LDPC1"""
|
|
self.fsk_ldpc_nin_1 = self.demodulate_audio(
|
|
self.fsk_ldpc_buffer_1,
|
|
self.fsk_ldpc_nin_1,
|
|
self.fsk_ldpc_freedv_1,
|
|
self.fsk_ldpc_bytes_out_1,
|
|
self.fsk_ldpc_bytes_per_frame_1,
|
|
FSK_LDPC1_STATE,
|
|
"fsk_ldpc1",
|
|
)
|
|
|
|
def init_data_threads(self):
|
|
# self.log.debug("[MDM] Starting worker_receive")
|
|
worker_received = threading.Thread(
|
|
target=self.worker_received, name="WORKER_THREAD", daemon=True
|
|
)
|
|
worker_received.start()
|
|
|
|
worker_transmit = threading.Thread(
|
|
target=self.worker_transmit, name="WORKER_THREAD", daemon=True
|
|
)
|
|
worker_transmit.start()
|
|
|
|
def worker_transmit(self) -> None:
|
|
"""Worker for FIFO queue for processing frames to be transmitted"""
|
|
while True:
|
|
# print queue size for debugging purposes
|
|
# TODO Lets check why we have several frames in our transmit queue which causes sometimes a double transmission
|
|
# we could do a cleanup after a transmission so theres no reason sending twice
|
|
queuesize = self.modem_transmit_queue.qsize()
|
|
self.log.debug("[MDM] self.modem_transmit_queue", qsize=queuesize)
|
|
data = self.modem_transmit_queue.get()
|
|
|
|
if data[0] in ["morse"]:
|
|
self.transmit_morse(repeats=data[1], repeat_delay=data[2], frames=data[3])
|
|
else:
|
|
self.transmit(
|
|
mode=data[0], repeats=data[1], repeat_delay=data[2], frames=data[3]
|
|
)
|
|
# self.modem_transmit_queue.task_done()
|
|
|
|
def worker_received(self) -> None:
|
|
"""Worker for FIFO queue for processing received frames"""
|
|
while True:
|
|
data = self.modem_received_queue.get()
|
|
self.log.debug("[MDM] worker_received: received data!")
|
|
# data[0] = bytes_out
|
|
# data[1] = freedv session
|
|
# data[2] = bytes_per_frame
|
|
# data[3] = snr
|
|
DATA_QUEUE_RECEIVED.put([data[0], data[1], data[2], data[3]])
|
|
self.modem_received_queue.task_done()
|
|
|
|
def get_frequency_offset(self, freedv: ctypes.c_void_p) -> float:
|
|
"""
|
|
Ask codec2 for the calculated (audio) frequency offset of the received signal.
|
|
|
|
:param freedv: codec2 instance to query
|
|
:type freedv: ctypes.c_void_p
|
|
:return: Offset of audio frequency in Hz
|
|
:rtype: float
|
|
"""
|
|
modemStats = codec2.MODEMSTATS()
|
|
codec2.api.freedv_get_modem_extended_stats(freedv, ctypes.byref(modemStats))
|
|
offset = round(modemStats.foff) * (-1)
|
|
return offset
|
|
|
|
def get_scatter(self, freedv: ctypes.c_void_p) -> None:
|
|
"""
|
|
Ask codec2 for data about the received signal and calculate the scatter plot.
|
|
|
|
:param freedv: codec2 instance to query
|
|
:type freedv: ctypes.c_void_p
|
|
"""
|
|
if not self.enable_scatter:
|
|
return
|
|
|
|
modemStats = codec2.MODEMSTATS()
|
|
ctypes.cast(
|
|
codec2.api.freedv_get_modem_extended_stats(freedv, ctypes.byref(modemStats)),
|
|
ctypes.c_void_p,
|
|
)
|
|
|
|
scatterdata = []
|
|
# original function before itertool
|
|
# for i in range(codec2.MODEM_STATS_NC_MAX):
|
|
# for j in range(1, codec2.MODEM_STATS_NR_MAX, 2):
|
|
# # print(f"{modemStats.rx_symbols[i][j]} - {modemStats.rx_symbols[i][j]}")
|
|
# xsymbols = round(modemStats.rx_symbols[i][j - 1] // 1000)
|
|
# ysymbols = round(modemStats.rx_symbols[i][j] // 1000)
|
|
# if xsymbols != 0.0 and ysymbols != 0.0:
|
|
# scatterdata.append({"x": str(xsymbols), "y": str(ysymbols)})
|
|
|
|
for i, j in itertools.product(range(codec2.MODEM_STATS_NC_MAX), range(1, codec2.MODEM_STATS_NR_MAX, 2)):
|
|
# print(f"{modemStats.rx_symbols[i][j]} - {modemStats.rx_symbols[i][j]}")
|
|
xsymbols = round(modemStats.rx_symbols[i][j - 1] // 1000)
|
|
ysymbols = round(modemStats.rx_symbols[i][j] // 1000)
|
|
if xsymbols != 0.0 and ysymbols != 0.0:
|
|
scatterdata.append({"x": str(xsymbols), "y": str(ysymbols)})
|
|
|
|
# Send all the data if we have too-few samples, otherwise send a sampling
|
|
if 150 > len(scatterdata) > 0:
|
|
self.event_manager.send_scatter_change(scatterdata)
|
|
|
|
else:
|
|
# only take every tenth data point
|
|
self.event_manager.send_scatter_change(scatterdata[::10])
|
|
|
|
def calculate_snr(self, freedv: ctypes.c_void_p) -> float:
|
|
"""
|
|
Ask codec2 for data about the received signal and calculate
|
|
the signal-to-noise ratio.
|
|
|
|
:param freedv: codec2 instance to query
|
|
:type freedv: ctypes.c_void_p
|
|
:return: Signal-to-noise ratio of the decoded data
|
|
:rtype: float
|
|
"""
|
|
try:
|
|
modem_stats_snr = ctypes.c_float()
|
|
modem_stats_sync = ctypes.c_int()
|
|
|
|
codec2.api.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)
|
|
# snr = np.clip(
|
|
# snr, -127, 127
|
|
# ) # limit to max value of -128/128 as a possible fix of #188
|
|
return snr
|
|
except Exception as err:
|
|
self.log.error(f"[MDM] calculate_snr: Exception: {err}")
|
|
return 0
|
|
|
|
def init_rig_control(self):
|
|
# Check how we want to control the radio
|
|
if self.radiocontrol == "rigctld":
|
|
import rigctld as rig
|
|
elif self.radiocontrol == "tci":
|
|
self.radio = self.tci_module
|
|
else:
|
|
import rigdummy as rig
|
|
|
|
if not self.radiocontrol in ["tci"]:
|
|
self.radio = rig.radio()
|
|
self.radio.open_rig(
|
|
rigctld_ip=self.rigctld_ip,
|
|
rigctld_port=self.rigctld_port,
|
|
)
|
|
hamlib_thread = threading.Thread(
|
|
target=self.update_rig_data, name="HAMLIB_THREAD", daemon=True
|
|
)
|
|
hamlib_thread.start()
|
|
|
|
hamlib_set_thread = threading.Thread(
|
|
target=self.set_rig_data, name="HAMLIB_SET_THREAD", daemon=True
|
|
)
|
|
hamlib_set_thread.start()
|
|
|
|
def set_rig_data(self) -> None:
|
|
"""
|
|
Set rigctld parameters like frequency, mode
|
|
THis needs to be processed in a queue
|
|
"""
|
|
while True:
|
|
cmd = RIGCTLD_COMMAND_QUEUE.get()
|
|
if cmd[0] == "set_frequency":
|
|
# [1] = Frequency
|
|
self.radio.set_frequency(cmd[1])
|
|
if cmd[0] == "set_mode":
|
|
# [1] = Mode
|
|
self.radio.set_mode(cmd[1])
|
|
|
|
def update_rig_data(self) -> None:
|
|
"""
|
|
Request information about the current state of the radio via hamlib
|
|
"""
|
|
while True:
|
|
try:
|
|
# this looks weird, but is necessary for avoiding rigctld packet colission sock
|
|
threading.Event().wait(0.25)
|
|
self.states.set("radio_frequency", self.radio.get_frequency())
|
|
threading.Event().wait(0.1)
|
|
self.states.set("radio_mode", self.radio.get_mode())
|
|
threading.Event().wait(0.1)
|
|
self.states.set("radio_bandwidth", self.radio.get_bandwidth())
|
|
threading.Event().wait(0.1)
|
|
self.states.set("radio_status", self.radio.get_status())
|
|
threading.Event().wait(0.1)
|
|
if self.states.is_transmitting:
|
|
self.radio_alc = self.radio.get_alc()
|
|
threading.Event().wait(0.1)
|
|
self.states.set("radio_rf_power", self.radio.get_level())
|
|
# threading.Event().wait(0.1)
|
|
self.states.set("radio_strength", self.radio.get_strength())
|
|
|
|
except Exception as e:
|
|
self.log.warning(
|
|
"[MDM] error getting radio data",
|
|
e=e,
|
|
)
|
|
threading.Event().wait(1)
|
|
|
|
def calculate_fft(self, data) -> None:
|
|
"""
|
|
Calculate an average signal strength of the channel to assess
|
|
whether the channel is "busy."
|
|
"""
|
|
# Initialize dbfs counter
|
|
rms_counter = 0
|
|
|
|
# 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(data)
|
|
|
|
# Set value 0 to 1 to avoid division by zero
|
|
fftarray[fftarray == 0] = 1
|
|
dfft = 10.0 * np.log10(abs(fftarray))
|
|
|
|
# get average of dfft
|
|
avg = np.mean(dfft)
|
|
|
|
# Detect signals which are higher than the
|
|
# average + 10 (+10 smoothes the output).
|
|
# Data higher than the average must be a signal.
|
|
# Therefore we are setting it to 100 so it will be highlighted
|
|
# Have to do this when we are not transmitting so our
|
|
# own sending data will not affect this too much
|
|
if not self.states.is_transmitting:
|
|
dfft[dfft > avg + 15] = 100
|
|
|
|
# Calculate audio dbfs
|
|
# https://stackoverflow.com/a/9763652
|
|
# calculate dbfs every 50 cycles for reducing CPU load
|
|
rms_counter += 1
|
|
if rms_counter > 50:
|
|
d = np.frombuffer(data, np.int16).astype(np.float32)
|
|
# calculate RMS and then dBFS
|
|
# https://dsp.stackexchange.com/questions/8785/how-to-compute-dbfs
|
|
# try except for avoiding runtime errors by division/0
|
|
try:
|
|
rms = int(np.sqrt(np.max(d ** 2)))
|
|
if rms == 0:
|
|
raise ZeroDivisionError
|
|
audio_dbfs = 20 * np.log10(rms / 32768)
|
|
self.states.set("audio_dbfs", audio_dbfs)
|
|
except Exception as e:
|
|
self.states.set("audio_dbfs", -100)
|
|
|
|
|
|
rms_counter = 0
|
|
|
|
# Convert data to int to decrease size
|
|
dfft = dfft.astype(int)
|
|
|
|
# Create list of dfft
|
|
dfftlist = dfft.tolist()
|
|
|
|
# Reduce area where the busy detection is enabled
|
|
# We want to have this in correlation with mode bandwidth
|
|
# TODO This is not correctly and needs to be checked for correct maths
|
|
# dfftlist[0:1] = 10,15Hz
|
|
# Bandwidth[Hz] / 10,15
|
|
# narrowband = 563Hz = 56
|
|
# wideband = 1700Hz = 167
|
|
# 1500Hz = 148
|
|
# 2700Hz = 266
|
|
# 3200Hz = 315
|
|
|
|
# slot
|
|
slot = 0
|
|
slot1 = [0, 65]
|
|
slot2 = [65,120]
|
|
slot3 = [120, 176]
|
|
slot4 = [176, 231]
|
|
slot5 = [231, len(dfftlist)]
|
|
|
|
# Set to true if we should increment delay count; else false to decrement
|
|
addDelay=False
|
|
for range in [slot1, slot2, slot3, slot4, slot5]:
|
|
|
|
range_start = range[0]
|
|
range_end = range[1]
|
|
# define the area, we are detecting busy state
|
|
slotdfft = dfft[range_start:range_end]
|
|
# 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(slotdfft[slotdfft > avg + 15]) >= 200 and not self.states.is_transmitting:
|
|
addDelay=True
|
|
self.states.channel_busy_slot[slot] = True
|
|
else:
|
|
self.states.channel_busy_slot[slot] = False
|
|
# increment slot
|
|
slot += 1
|
|
if addDelay:
|
|
# Limit delay counter to a maximum of 200. The higher this value,
|
|
# the longer we will wait until releasing state
|
|
self.states.set("channel_busy", True)
|
|
self.channel_busy_delay = min(self.channel_busy_delay + 10, 200)
|
|
else:
|
|
# Decrement channel busy counter if no signal has been detected.
|
|
self.channel_busy_delay = max(self.channel_busy_delay - 1, 0)
|
|
# When our channel busy counter reaches 0, toggle state to False
|
|
if self.channel_busy_delay == 0:
|
|
self.states.set("channel_busy", False)
|
|
|
|
# erase queue if greater than 10
|
|
if self.fft_queue.qsize() >= 10:
|
|
self.fft_queue = queue.Queue()
|
|
|
|
self.fft_queue.put(dfftlist[:315]) # 315 --> bandwidth 3200
|
|
except Exception as err:
|
|
self.log.error(f"[MDM] calculate_fft: Exception: {err}")
|
|
self.log.debug("[MDM] Setting fft=0")
|
|
# else 0
|
|
self.fft_queue.put([0])
|
|
|
|
def set_frames_per_burst(self, frames_per_burst: int) -> None:
|
|
"""
|
|
Configure codec2 to send the configured number of frames per burst.
|
|
|
|
:param frames_per_burst: Number of frames per burst requested
|
|
:type frames_per_burst: int
|
|
"""
|
|
# Limit frames per burst to acceptable values
|
|
frames_per_burst = min(frames_per_burst, 1)
|
|
frames_per_burst = max(frames_per_burst, 5)
|
|
|
|
frames_per_burst = 1
|
|
|
|
codec2.api.freedv_set_frames_per_burst(self.dat0_datac1_freedv, frames_per_burst)
|
|
codec2.api.freedv_set_frames_per_burst(self.dat0_datac3_freedv, frames_per_burst)
|
|
codec2.api.freedv_set_frames_per_burst(self.dat0_datac4_freedv, frames_per_burst)
|
|
codec2.api.freedv_set_frames_per_burst(self.fsk_ldpc_freedv_0, frames_per_burst)
|
|
|
|
def reset_data_sync(self) -> None:
|
|
"""
|
|
reset sync state for data modes
|
|
|
|
:param frames_per_burst: Number of frames per burst requested
|
|
:type frames_per_burst: int
|
|
"""
|
|
|
|
codec2.api.freedv_set_sync(self.dat0_datac1_freedv, 0)
|
|
codec2.api.freedv_set_sync(self.dat0_datac3_freedv, 0)
|
|
codec2.api.freedv_set_sync(self.dat0_datac4_freedv, 0)
|
|
codec2.api.freedv_set_sync(self.fsk_ldpc_freedv_0, 0)
|
|
|
|
|
|
def open_codec2_instance(mode: int) -> ctypes.c_void_p:
|
|
"""
|
|
Return a codec2 instance of the type `mode`
|
|
|
|
:param mode: Type of codec2 instance to return
|
|
:type mode: Union[int, str]
|
|
:return: C-function of the requested codec2 instance
|
|
:rtype: ctypes.c_void_p
|
|
"""
|
|
if mode in [codec2.FREEDV_MODE.fsk_ldpc_0.value]:
|
|
return ctypes.cast(
|
|
codec2.api.freedv_open_advanced(
|
|
codec2.FREEDV_MODE.fsk_ldpc.value,
|
|
ctypes.byref(codec2.api.FREEDV_MODE_FSK_LDPC_0_ADV),
|
|
),
|
|
ctypes.c_void_p,
|
|
)
|
|
|
|
if mode in [codec2.FREEDV_MODE.fsk_ldpc_1.value]:
|
|
return ctypes.cast(
|
|
codec2.api.freedv_open_advanced(
|
|
codec2.FREEDV_MODE.fsk_ldpc.value,
|
|
ctypes.byref(codec2.api.FREEDV_MODE_FSK_LDPC_1_ADV),
|
|
),
|
|
ctypes.c_void_p,
|
|
)
|
|
|
|
return ctypes.cast(codec2.api.freedv_open(mode), ctypes.c_void_p)
|
|
|
|
def get_bytes_per_frame(mode: int) -> int:
|
|
"""
|
|
Provide bytes per frame information for accessing from data handler
|
|
|
|
:param mode: Codec2 mode to query
|
|
:type mode: int or str
|
|
:return: Bytes per frame of the supplied codec2 data mode
|
|
:rtype: int
|
|
"""
|
|
freedv = open_codec2_instance(mode)
|
|
# TODO add close session
|
|
# get number of bytes per frame for mode
|
|
return int(codec2.api.freedv_get_bits_per_modem_frame(freedv) / 8)
|
|
|
|
def set_audio_volume(datalist: np.ndarray, dB: float) -> np.ndarray:
|
|
"""
|
|
Scale values for the provided audio samples by dB.
|
|
|
|
:param datalist: Audio samples to scale
|
|
:type datalist: np.ndarray
|
|
:param dB: Decibels to scale samples, constrained to the range [-50, 50]
|
|
:type dB: float
|
|
:return: Scaled audio samples
|
|
:rtype: np.ndarray
|
|
"""
|
|
try:
|
|
dB = float(dB)
|
|
except ValueError as e:
|
|
print(f"[MDM] Changing audio volume failed with error: {e}")
|
|
dB = 0.0 # 0 dB means no change
|
|
|
|
# Clip dB value to the range [-50, 50]
|
|
dB = np.clip(dB, -30, 20)
|
|
|
|
# Ensure datalist is an np.ndarray
|
|
if not isinstance(datalist, np.ndarray):
|
|
print("[MDM] Invalid data type for datalist. Expected np.ndarray.")
|
|
return datalist
|
|
|
|
# Convert dB to linear scale
|
|
scale_factor = 10 ** (dB / 20)
|
|
|
|
# Scale samples
|
|
scaled_data = datalist * scale_factor
|
|
|
|
# Clip values to int16 range and convert data type
|
|
return np.clip(scaled_data, -32768, 32767).astype(np.int16)
|
|
|
|
def get_modem_error_state():
|
|
"""
|
|
get current state buffer and return True of contains 10
|
|
|
|
"""
|
|
|
|
if RECEIVE_DATAC1 and 10 in DAT0_DATAC1_STATE:
|
|
DAT0_DATAC1_STATE.clear()
|
|
return True
|
|
if RECEIVE_DATAC3 and 10 in DAT0_DATAC3_STATE:
|
|
DAT0_DATAC3_STATE.clear()
|
|
return True
|
|
if RECEIVE_DATAC4 and 10 in DAT0_DATAC4_STATE:
|
|
DAT0_DATAC4_STATE.clear()
|
|
return True
|
|
|
|
return False
|