#!/usr/bin/env python3 # -*- coding: utf-8 -*- """ Created on Wed Dec 23 07:04:24 2020 @author: DJ2LS """ import argparse import ctypes import queue import sys import time import numpy as np import pyaudio 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( "--audiodev", dest="AUDIO_OUTPUT_DEVICE", default=-1, type=int, help="audio output device number to use", ) 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_known_args() if args.LIST: p = pyaudio.PyAudio() for dev in range(p.get_device_count()): print("audiodev: ", dev, p.get_device_info_by_index(dev)["name"]) sys.exit() 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.DELAY_BETWEEN_BURSTS = args.DELAY_BETWEEN_BURSTS / 1000 # AUDIO PARAMETERS # v-- consider increasing if you get nread_exceptions > 0 self.AUDIO_FRAMES_PER_BUFFER = 2400 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 a 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 = [ dev for dev in range(self.p.get_device_count()) if "Loopback: PCM" in self.p.get_device_info_by_index(dev)["name"] ] 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: sys.exit() print( f"AUDIO OUTPUT DEVICE: {self.AUDIO_OUTPUT_DEVICE} " f"DEVICE: {self.p.get_device_info_by_index(self.AUDIO_OUTPUT_DEVICE)['name']} " f"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, ) else: print("test_callback_multimode_tx: Not written for STDOUT usage.") print("Exiting.") sys.exit() # 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): modes = [ codec2.api.FREEDV_MODE_DATAC0, codec2.api.FREEDV_MODE_DATAC1, codec2.api.FREEDV_MODE_DATAC3, ] for m in modes: freedv = ctypes.cast(codec2.api.freedv_open(m), ctypes.c_void_p) n_tx_modem_samples = codec2.api.freedv_get_n_tx_modem_samples(freedv) mod_out = ctypes.create_string_buffer(2 * n_tx_modem_samples) n_tx_preamble_modem_samples = ( codec2.api.freedv_get_n_tx_preamble_modem_samples(freedv) ) mod_out_preamble = ctypes.create_string_buffer( 2 * n_tx_preamble_modem_samples ) n_tx_postamble_modem_samples = ( codec2.api.freedv_get_n_tx_postamble_modem_samples(freedv) ) mod_out_postamble = ctypes.create_string_buffer( 2 * n_tx_postamble_modem_samples ) bytes_per_frame = int( codec2.api.freedv_get_bits_per_modem_frame(freedv) / 8 ) payload_per_frame = bytes_per_frame - 2 buffer = bytearray(payload_per_frame) # set buffer size to length of data which will be sent buffer[: len(self.data_out)] = self.data_out crc = ctypes.c_ushort( codec2.api.freedv_gen_crc16(bytes(buffer), payload_per_frame) ) # generate CRC16 # convert crc to 2 byte hex string crc = crc.value.to_bytes(2, byteorder="big") buffer += crc # append crc16 to buffer data = (ctypes.c_ubyte * bytes_per_frame).from_buffer_copy(buffer) 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 = ctypes.create_string_buffer(samples_delay * 2) txbuffer += bytes(mod_out_silence) # 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 modulated 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()