#!/usr/bin/env python3 # -*- coding: utf-8 -*- """ Created on Wed Dec 23 07:04:24 2020 @author: DJ2LS """ import ctypes from ctypes import * import pathlib import pyaudio import sys import logging import time import threading import sys import argparse import queue import numpy as np sys.path.insert(0,'..') from tnc import codec2 #--------------------------------------------GET PARAMETER INPUTS parser = argparse.ArgumentParser(description='FreeDATA audio test') 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) 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 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="generate testframes") 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() class Test(): def __init__(self): self.dataqueue = queue.Queue() self.N_BURSTS = args.N_BURSTS self.N_FRAMES_PER_BURST = args.N_FRAMES_PER_BURST self.AUDIO_OUTPUT_DEVICE = args.AUDIO_OUTPUT_DEVICE self.MODE = codec2.FREEDV_MODE[args.FREEDV_MODE].value self.DELAY_BETWEEN_BURSTS = args.DELAY_BETWEEN_BURSTS/1000 # AUDIO PARAMETERS self.AUDIO_FRAMES_PER_BUFFER = 2400 # <- consider increasing if you get nread_exceptions > 0 self.MODEM_SAMPLE_RATE = codec2.api.FREEDV_FS_8000 self.AUDIO_SAMPLE_RATE_TX = 48000 # make sure our resampler will work assert (self.AUDIO_SAMPLE_RATE_TX / self.MODEM_SAMPLE_RATE) == codec2.api.FDMDV_OS_48 self.transmit = True self.resampler = codec2.resampler() # check if we want to use an audio device then do an pyaudio init if self.AUDIO_OUTPUT_DEVICE != -1: self.p = pyaudio.PyAudio() # auto search for loopback devices if self.AUDIO_OUTPUT_DEVICE == -2: loopback_list = [] for dev in range(0,self.p.get_device_count()): if 'Loopback: PCM' in self.p.get_device_info_by_index(dev)["name"]: loopback_list.append(dev) if len(loopback_list) >= 2: self.AUDIO_OUTPUT_DEVICE = loopback_list[0] #0 = RX 1 = TX print(f"loopback_list rx: {loopback_list}", file=sys.stderr) else: quit() print(f"AUDIO OUTPUT DEVICE: {self.AUDIO_OUTPUT_DEVICE} DEVICE: {self.p.get_device_info_by_index(self.AUDIO_OUTPUT_DEVICE)['name']} \ AUDIO SAMPLE RATE: {self.AUDIO_SAMPLE_RATE_TX}", file=sys.stderr) self.stream_tx = self.p.open(format=pyaudio.paInt16, channels=1, rate=self.AUDIO_SAMPLE_RATE_TX, frames_per_buffer=self.AUDIO_FRAMES_PER_BUFFER, input=False, output=True, output_device_index=self.AUDIO_OUTPUT_DEVICE, stream_callback=self.callback ) # open codec2 instance self.freedv = cast(codec2.api.freedv_open(self.MODE), c_void_p) # get number of bytes per frame for mode self.bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(self.freedv)/8) self.bytes_out = create_string_buffer(self.bytes_per_frame) codec2.api.freedv_set_frames_per_burst(self.freedv,self.N_FRAMES_PER_BURST) # Copy received 48 kHz to a file. Listen to this file with: # aplay -r 48000 -f S16_LE rx48_callback.raw # Corruption of this file is a good way to detect audio card issues self.ftx = open("tx48_callback.raw", mode='wb') # data binary string if args.TESTFRAMES: self.data_out = bytearray(14) self.data_out[:1] = bytes([255]) self.data_out[1:2] = bytes([1]) self.data_out[2:] = b'HELLO WORLD' else: self.data_out = b'HELLO WORLD!' def callback(self, data_in48k, frame_count, time_info, status): data_out48k = self.dataqueue.get() return (data_out48k, pyaudio.paContinue) def run_audio(self): try: print(f"starting pyaudio callback", file=sys.stderr) self.stream_tx.start_stream() except Exception as e: print(f"pyAudio error: {e}", file=sys.stderr) sheeps = 0 while self.transmit: time.sleep(1) sheeps = sheeps + 1 print(f"counting sheeps...{sheeps}") self.ftx.close() # close pyaudio instance self.stream_tx.close() self.p.terminate() def create_modulation(self): # open codec2 instance freedv = cast(codec2.api.freedv_open(self.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(self.data_out)] = self.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: {self.N_BURSTS} TOTAL FRAMES_PER_BURST: {self.N_FRAMES_PER_BURST}", file=sys.stderr) for i in range(1,self.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,self.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" GENERATING TX BURST: {i}/{self.N_BURSTS} FRAME: {n}/{self.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(self.MODEM_SAMPLE_RATE*self.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: {self.DELAY_BETWEEN_BURSTS}", file=sys.stderr) # resample up to 48k (resampler works on np.int16) x = np.frombuffer(txbuffer, dtype=np.int16) txbuffer_48k = self.resampler.resample8_to_48(x) # split modualted audio to chunks #https://newbedev.com/how-to-split-a-byte-string-into-separate-bytes-in-python txbuffer_48k = bytes(txbuffer_48k) chunk = [txbuffer_48k[i:i+self.AUDIO_FRAMES_PER_BUFFER*2] for i in range(0, len(txbuffer_48k), self.AUDIO_FRAMES_PER_BUFFER*2)] # add modulated chunks to fifo buffer for c in chunk: # if data is shorter than the expcected audio frames per buffer we need to append 0 # to prevent the callback from stucking/crashing if len(c) < self.AUDIO_FRAMES_PER_BUFFER*2: c += bytes(self.AUDIO_FRAMES_PER_BUFFER*2 - len(c)) self.dataqueue.put(c) test = Test() test.create_modulation() test.run_audio()