diff --git a/test/test_callback_tx.py b/test/test_callback_tx.py new file mode 100644 index 00000000..e1d4c04a --- /dev/null +++ b/test/test_callback_tx.py @@ -0,0 +1,230 @@ +#!/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('--debug', dest="DEBUGGING_MODE", action="store_true") +parser.add_argument('--timeout', dest="TIMEOUT", default=10, type=int, help="Timeout (seconds) before test ends") +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.DEBUGGING_MODE = args.DEBUGGING_MODE + self.TIMEOUT = args.TIMEOUT + 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) + + + while self.transmit: + time.sleep(0.1) + + + 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"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()