#!/usr/bin/env python3 # -*- coding: utf-8 -*- import ctypes from ctypes import * import pathlib import pyaudio import time import argparse import sys sys.path.insert(0,'..') from tnc import codec2 import numpy as np # GET PARAMETER INPUTS parser = argparse.ArgumentParser(description='Simons TEST TNC') parser.add_argument('--bursts', dest="N_BURSTS", default=1, type=int) parser.add_argument('--framesperburst', dest="N_FRAMES_PER_BURST", default=1, type=int) parser.add_argument('--delay', dest="DELAY_BETWEEN_BURSTS", default=500, type=int, help="delay between bursts in ms") parser.add_argument('--mode', dest="FREEDV_MODE", type=str, choices=['datac0', 'datac1', 'datac3']) parser.add_argument('--audiodev', dest="AUDIO_OUTPUT_DEVICE", default=-1, type=int, help="audio output device number to use, use -2 to automatically select a loopback device") parser.add_argument('--list', dest="LIST", action="store_true", help="list audio devices by number and exit") parser.add_argument('--testframes', dest="TESTFRAMES", action="store_true", default=False, help="list audio devices by number and exit") args = parser.parse_args() if args.LIST: p = pyaudio.PyAudio() for dev in range(0,p.get_device_count()): print("audiodev: ", dev, p.get_device_info_by_index(dev)["name"]) quit() N_BURSTS = args.N_BURSTS N_FRAMES_PER_BURST = args.N_FRAMES_PER_BURST DELAY_BETWEEN_BURSTS = args.DELAY_BETWEEN_BURSTS/1000 AUDIO_OUTPUT_DEVICE = args.AUDIO_OUTPUT_DEVICE MODE = codec2.FREEDV_MODE[args.FREEDV_MODE].value # AUDIO PARAMETERS AUDIO_FRAMES_PER_BUFFER = 2400 MODEM_SAMPLE_RATE = codec2.api.FREEDV_FS_8000 AUDIO_SAMPLE_RATE_TX = 48000 assert (AUDIO_SAMPLE_RATE_TX % MODEM_SAMPLE_RATE) == 0 # check if we want to use an audio device then do an pyaudio init if AUDIO_OUTPUT_DEVICE != -1: p = pyaudio.PyAudio() # auto search for loopback devices if AUDIO_OUTPUT_DEVICE == -2: loopback_list = [] for dev in range(0,p.get_device_count()): if 'Loopback: PCM' in p.get_device_info_by_index(dev)["name"]: loopback_list.append(dev) if len(loopback_list) >= 2: AUDIO_OUTPUT_DEVICE = loopback_list[1] #0 = RX 1 = TX print(f"loopback_list tx: {loopback_list}", file=sys.stderr) else: quit() print(f"AUDIO OUTPUT DEVICE: {AUDIO_OUTPUT_DEVICE} DEVICE: {p.get_device_info_by_index(AUDIO_OUTPUT_DEVICE)['name']} \ AUDIO SAMPLE RATE: {AUDIO_SAMPLE_RATE_TX}", file=sys.stderr) # pyaudio init stream_tx = p.open(format=pyaudio.paInt16, channels=1, rate=AUDIO_SAMPLE_RATE_TX, frames_per_buffer=AUDIO_FRAMES_PER_BUFFER, #n_nom_modem_samples output=True, output_device_index=AUDIO_OUTPUT_DEVICE ) resampler = codec2.resampler() # data binary string if args.TESTFRAMES: data_out = bytearray(14) data_out[:1] = bytes([255]) data_out[1:2] = bytes([1]) data_out[2:] = b'HELLO WORLD' else: data_out = b'HELLO WORLD!' # ---------------------------------------------------------------- # open codec2 instance freedv = cast(codec2.api.freedv_open(MODE), c_void_p) # get number of bytes per frame for mode bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(freedv)/8) payload_bytes_per_frame = bytes_per_frame -2 # init buffer for data n_tx_modem_samples = codec2.api.freedv_get_n_tx_modem_samples(freedv) mod_out = 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 = 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 = create_string_buffer(n_tx_postamble_modem_samples * 2) # create buffer for data buffer = bytearray(payload_bytes_per_frame) # use this if CRC16 checksum is required ( DATA1-3) buffer[:len(data_out)] = data_out # set buffersize to length of data which will be send # create crc for data frame - we are using the crc function shipped with codec2 to avoid # crc algorithm incompatibilities crc = ctypes.c_ushort(codec2.api.freedv_gen_crc16(bytes(buffer), payload_bytes_per_frame)) # generate CRC16 crc = crc.value.to_bytes(2, byteorder='big') # convert crc to 2 byte hex string buffer += crc # append crc16 to buffer print(f"TOTAL BURSTS: {N_BURSTS} TOTAL FRAMES_PER_BURST: {N_FRAMES_PER_BURST}", file=sys.stderr) for i in range(1,N_BURSTS+1): # write preamble to txbuffer codec2.api.freedv_rawdatapreambletx(freedv, mod_out_preamble) txbuffer = bytes(mod_out_preamble) # create modulaton for N = FRAMESPERBURST and append it to txbuffer for n in range(1,N_FRAMES_PER_BURST+1): data = (ctypes.c_ubyte * bytes_per_frame).from_buffer_copy(buffer) codec2.api.freedv_rawdatatx(freedv,mod_out,data) # modulate DATA and save it into mod_out pointer txbuffer += bytes(mod_out) print(f"TX BURST: {i}/{N_BURSTS} FRAME: {n}/{N_FRAMES_PER_BURST}", file=sys.stderr) # append postamble to txbuffer codec2.api.freedv_rawdatapostambletx(freedv, mod_out_postamble) txbuffer += bytes(mod_out_postamble) # append a delay between bursts as audio silence samples_delay = int(MODEM_SAMPLE_RATE*DELAY_BETWEEN_BURSTS) mod_out_silence = create_string_buffer(samples_delay*2) txbuffer += bytes(mod_out_silence) #print(f"samples_delay: {samples_delay} DELAY_BETWEEN_BURSTS: {DELAY_BETWEEN_BURSTS}", file=sys.stderr) # resample up to 48k (resampler works on np.int16) x = np.frombuffer(txbuffer, dtype=np.int16) txbuffer_48k = resampler.resample8_to_48(x) # check if we want to use an audio device or stdout if AUDIO_OUTPUT_DEVICE != -1: # Gotcha: we have to convert from np.int16 to Python "bytes" stream_tx.write(txbuffer_48k.tobytes()) else: # print data to terminal for piping the output to other programs sys.stdout.buffer.write(txbuffer_48k) sys.stdout.flush() # and at last check if we had an opened pyaudio instance and close it if AUDIO_OUTPUT_DEVICE != -1: time.sleep(stream_tx.get_output_latency()) stream_tx.stop_stream() stream_tx.close() p.terminate()