mirror of
https://github.com/DJ2LS/FreeDATA
synced 2024-05-14 10:04:33 +02:00
215 lines
8.4 KiB
Python
215 lines
8.4 KiB
Python
#!/usr/bin/env python3
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# -*- coding: utf-8 -*-
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import ctypes
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from ctypes import *
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import pathlib
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import pyaudio
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import time
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import argparse
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import sys
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sys.path.insert(0,'..')
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from tnc import codec2
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import numpy as np
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# GET PARAMETER INPUTS
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parser = argparse.ArgumentParser(description='Simons TEST TNC')
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parser.add_argument('--bursts', dest="N_BURSTS", default=1, type=int)
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parser.add_argument('--framesperburst', dest="N_FRAMES_PER_BURST", default=1, type=int)
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parser.add_argument('--delay', dest="DELAY_BETWEEN_BURSTS", default=500, type=int,
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help="delay between bursts in ms")
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parser.add_argument('--mode', dest="FREEDV_MODE", type=str, choices=['datac0', 'datac1', 'datac3', 'fsk_ldpc'])
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parser.add_argument('--audiodev', dest="AUDIO_OUTPUT_DEVICE", default=-1, type=int,
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help="audio output device number to use, use -2 to automatically select a loopback device")
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parser.add_argument('--list', dest="LIST", action="store_true", help="list audio devices by number and exit")
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parser.add_argument('--testframes', dest="TESTFRAMES", action="store_true", default=False, help="list audio devices by number and exit")
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args = parser.parse_args()
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if args.LIST:
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p = pyaudio.PyAudio()
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for dev in range(0,p.get_device_count()):
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print("audiodev: ", dev, p.get_device_info_by_index(dev)["name"])
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quit()
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N_BURSTS = args.N_BURSTS
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N_FRAMES_PER_BURST = args.N_FRAMES_PER_BURST
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DELAY_BETWEEN_BURSTS = args.DELAY_BETWEEN_BURSTS/1000
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AUDIO_OUTPUT_DEVICE = args.AUDIO_OUTPUT_DEVICE
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MODE = codec2.FREEDV_MODE[args.FREEDV_MODE].value
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# AUDIO PARAMETERS
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AUDIO_FRAMES_PER_BUFFER = 2400
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MODEM_SAMPLE_RATE = codec2.api.FREEDV_FS_8000
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AUDIO_SAMPLE_RATE_TX = 48000
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assert (AUDIO_SAMPLE_RATE_TX % MODEM_SAMPLE_RATE) == 0
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# check if we want to use an audio device then do an pyaudio init
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if AUDIO_OUTPUT_DEVICE != -1:
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p = pyaudio.PyAudio()
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# auto search for loopback devices
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if AUDIO_OUTPUT_DEVICE == -2:
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loopback_list = []
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for dev in range(0,p.get_device_count()):
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if 'Loopback: PCM' in 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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AUDIO_OUTPUT_DEVICE = loopback_list[1] #0 = RX 1 = TX
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print(f"loopback_list tx: {loopback_list}", file=sys.stderr)
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else:
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quit()
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print(f"AUDIO OUTPUT DEVICE: {AUDIO_OUTPUT_DEVICE} DEVICE: {p.get_device_info_by_index(AUDIO_OUTPUT_DEVICE)['name']} \
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AUDIO SAMPLE RATE: {AUDIO_SAMPLE_RATE_TX}", file=sys.stderr)
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# pyaudio init
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stream_tx = p.open(format=pyaudio.paInt16,
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channels=1,
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rate=AUDIO_SAMPLE_RATE_TX,
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frames_per_buffer=AUDIO_FRAMES_PER_BUFFER, #n_nom_modem_samples
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output=True,
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output_device_index=AUDIO_OUTPUT_DEVICE
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)
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resampler = codec2.resampler()
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# data binary string
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if args.TESTFRAMES:
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data_out = bytearray(14)
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data_out[:1] = bytes([255])
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data_out[1:2] = bytes([1])
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data_out[2:] = b'HELLO WORLD'
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else:
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data_out = b'HELLO WORLD!'
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# ----------------------------------------------------------------
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class ADVANCED(ctypes.Structure):
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""" """
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_fields_ = [
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("interleave_frames", ctypes.c_int),
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("M", ctypes.c_int),
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("Rs", ctypes.c_int),
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("Fs", ctypes.c_int),
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("first_tone", ctypes.c_int),
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("tone_spacing", ctypes.c_int),
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("codename", ctypes.c_char_p),
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]
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adv = ADVANCED()
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adv.interleave_frames = 0 # max amplitude
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adv.M = 2 # number of fsk tones 2/4
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adv.Rs = 100 # symbol rate
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adv.Fs = 8000 # sample rate
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adv.first_tone = 1500 # first tone freq
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adv.tone_spacing = 200 # shift between tones
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'''
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HRA_112_112 rate 0.50 (224,112) # BPF: 14
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HRA_56_56 rate 0.50 (112,56) # BPF: 7, geht nicht
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H_2064_516_sparse rate 0.80 (2580,2064) #BPF: 258,
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HRAb_396_504 rate 0.79 (504,396) # BPF: 49, geht nicht payload !== %8
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H_256_768_22 rate 0.33 (768,256) # BPF: 32, geht --> geht gut
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H_256_512_4 rate 0.50 (512,256) # BPF: 32, geht
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HRAa_1536_512 rate 0.75 (2048,1536) #BPF: 192, geht nicht
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H_128_256_5 rate 0.50 (256,128) # BPF: 16, geht nicht
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H_4096_8192_3d rate 0.50 (8192,4096) # BPF: 512, geht nicht
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H_16200_9720 rate 0.60 (16200,9720) # BPF: 1215
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H_1024_2048_4f rate 0.50 (2048,1024) # BPF: 128, geht nicht
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'''
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adv.codename = 'H_256_512_4'.encode('utf-8') # code word
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if MODE == 9:
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# open codec2 advanced instance for ldpc codes
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print(f"adv.interleave_frames {adv.interleave_frames}")
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print(f"adv.M {adv.M}")
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print(f"adv.Rs {adv.Rs}")
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print(f"adv.Fs {adv.Fs}")
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print(f"adv.first_tone {adv.first_tone}")
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print(f"adv.tone_spacing {adv.tone_spacing}")
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print(f"adv.codename {adv.codename}")
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#freedv = codec2.api.freedv_open_advanced(MODE, pointer(adv))
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freedv = cast(codec2.api.freedv_open_advanced(MODE, ctypes.byref(adv)), c_void_p)
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else:
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# open codec2 instance
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freedv = cast(codec2.api.freedv_open(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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# 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(data_out)] = data_out # 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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print(f"TOTAL BURSTS: {N_BURSTS} TOTAL FRAMES_PER_BURST: {N_FRAMES_PER_BURST}", file=sys.stderr)
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for i in range(1,N_BURSTS+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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txbuffer = bytes()
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# create modulaton for N = FRAMESPERBURST and append it to txbuffer
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for n in range(1,N_FRAMES_PER_BURST+1):
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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
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txbuffer += bytes(mod_out)
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print(f"TX BURST: {i}/{N_BURSTS} FRAME: {n}/{N_FRAMES_PER_BURST}", file=sys.stderr)
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# append postamble to txbuffer
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codec2.api.freedv_rawdatapostambletx(freedv, mod_out_postamble)
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#txbuffer += bytes(mod_out_postamble)
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# append a delay between bursts as audio silence
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samples_delay = int(MODEM_SAMPLE_RATE*DELAY_BETWEEN_BURSTS)
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mod_out_silence = create_string_buffer(samples_delay*2)
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txbuffer += bytes(mod_out_silence)
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#print(f"samples_delay: {samples_delay} DELAY_BETWEEN_BURSTS: {DELAY_BETWEEN_BURSTS}", file=sys.stderr)
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# resample up to 48k (resampler works on np.int16)
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x = np.frombuffer(txbuffer, dtype=np.int16)
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txbuffer_48k = resampler.resample8_to_48(x)
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# check if we want to use an audio device or stdout
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if AUDIO_OUTPUT_DEVICE != -1:
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# Gotcha: we have to convert from np.int16 to Python "bytes"
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stream_tx.write(txbuffer_48k.tobytes())
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else:
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# print data to terminal for piping the output to other programs
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sys.stdout.buffer.write(txbuffer_48k)
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sys.stdout.flush()
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# and at last check if we had an opened pyaudio instance and close it
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if AUDIO_OUTPUT_DEVICE != -1:
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time.sleep(stream_tx.get_output_latency())
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stream_tx.stop_stream()
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stream_tx.close()
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p.terminate()
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