#!/usr/bin/env python3 # -*- coding: utf-8 -*- """ Created on Wed Dec 23 07:04:24 2020 @author: DJ2LS """ import argparse import ctypes import sys import time import numpy as np import sounddevice as sd sys.path.insert(0, "..") from tnc import codec2 def test_rx(): args = parse_arguments() if args.LIST: devices = sd.query_devices(device=None, kind=None) index = 0 for device in devices: print(f"{index} {device['name']}") index += 1 sd._terminate() sys.exit() N_BURSTS = args.N_BURSTS N_FRAMES_PER_BURST = args.N_FRAMES_PER_BURST AUDIO_INPUT_DEVICE = args.AUDIO_INPUT_DEVICE MODE = codec2.FREEDV_MODE[args.FREEDV_MODE].value DEBUGGING_MODE = args.DEBUGGING_MODE MAX_TIME = args.TIMEOUT # AUDIO PARAMETERS # v-- consider increasing if you get nread_exceptions > 0 AUDIO_FRAMES_PER_BUFFER = 2400 * 2 MODEM_SAMPLE_RATE = codec2.api.FREEDV_FS_8000 AUDIO_SAMPLE_RATE_RX = 48000 # make sure our resampler will work assert (AUDIO_SAMPLE_RATE_RX / MODEM_SAMPLE_RATE) == codec2.api.FDMDV_OS_48 # check if we want to use an audio device then do an pyaudio init if AUDIO_INPUT_DEVICE != -1: # auto search for loopback devices if AUDIO_INPUT_DEVICE == -2: loopback_list = [] devices = sd.query_devices(device=None, kind=None) for index, device in enumerate(devices): if "Loopback: PCM" in device["name"]: print(index) loopback_list.append(index) if loopback_list: # 0 = RX 1 = TX AUDIO_INPUT_DEVICE = loopback_list[0] print(f"loopback_list tx: {loopback_list}", file=sys.stderr) else: print("not enough audio loopback devices ready...") print("you should wait about 30 seconds...") sd._terminate() sys.exit() print(f"AUDIO INPUT DEVICE: {AUDIO_INPUT_DEVICE}", file=sys.stderr) # audio stream init stream_rx = sd.RawStream( channels=1, dtype="int16", device=AUDIO_INPUT_DEVICE, samplerate=AUDIO_SAMPLE_RATE_RX, blocksize=4800, ) stream_rx.start() # ---------------------------------------------------------------- # DATA CHANNEL INITIALISATION # open codec2 instance freedv = ctypes.cast(codec2.api.freedv_open(MODE), ctypes.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 n_max_modem_samples = codec2.api.freedv_get_n_max_modem_samples(freedv) bytes_out = ctypes.create_string_buffer(bytes_per_frame) codec2.api.freedv_set_frames_per_burst(freedv, N_FRAMES_PER_BURST) total_n_bytes = 0 rx_total_frames = 0 rx_frames = 0 rx_bursts = 0 rx_errors = 0 nread_exceptions = 0 timeout = time.time() + MAX_TIME receive = True audio_buffer = codec2.audio_buffer(AUDIO_FRAMES_PER_BUFFER * 2) resampler = codec2.resampler() # time meassurement time_start = 0 time_end = 0 # Copy received 48 kHz to a file. Listen to this file with: # aplay -r 48000 -f S16_LE rx48.raw # Corruption of this file is a good way to detect audio card issues frx = open("rx48.raw", mode="wb") # initial number of samples we need nin = codec2.api.freedv_nin(freedv) while receive and time.time() < timeout: if AUDIO_INPUT_DEVICE != -1: try: # data_in48k = stream_rx.read(AUDIO_FRAMES_PER_BUFFER, exception_on_overflow = True) data_in48k, overflowed = stream_rx.read(AUDIO_FRAMES_PER_BUFFER) except OSError as err: print(err, file=sys.stderr) # if str(err).find("Input overflowed") != -1: # nread_exceptions += 1 # if str(err).find("Stream closed") != -1: # print("Ending...") # receive = False else: data_in48k = sys.stdin.buffer.read(AUDIO_FRAMES_PER_BUFFER * 2) # insert samples in buffer x = np.frombuffer(data_in48k, dtype=np.int16) # print(x) # x = data_in48k x.tofile(frx) if len(x) != AUDIO_FRAMES_PER_BUFFER: receive = False x = resampler.resample48_to_8(x) audio_buffer.push(x) # when we have enough samples call FreeDV Rx while audio_buffer.nbuffer >= nin: # start time measurement time_start = time.time() # demodulate audio nbytes = codec2.api.freedv_rawdatarx( freedv, bytes_out, audio_buffer.buffer.ctypes ) time_end = time.time() audio_buffer.pop(nin) # call me on every loop! nin = codec2.api.freedv_nin(freedv) rx_status = codec2.api.freedv_get_rx_status(freedv) if rx_status & codec2.api.FREEDV_RX_BIT_ERRORS: rx_errors = rx_errors + 1 if DEBUGGING_MODE: rx_status = codec2.api.rx_sync_flags_to_text[rx_status] time_needed = time_end - time_start print( f"nin: {nin:5d} rx_status: {rx_status:4s} " f"naudio_buffer: {audio_buffer.nbuffer:4d} time: {time_needed:4f}", file=sys.stderr, ) if nbytes: total_n_bytes += nbytes if nbytes == bytes_per_frame: rx_total_frames += 1 rx_frames += 1 if rx_frames == N_FRAMES_PER_BURST: rx_frames = 0 rx_bursts += 1 if rx_bursts == N_BURSTS: receive = False if time.time() >= timeout: print("TIMEOUT REACHED") if nread_exceptions: print( f"nread_exceptions {nread_exceptions:d} - receive audio lost! " "Consider increasing Pyaudio frames_per_buffer...", file=sys.stderr, ) print( f"RECEIVED BURSTS: {rx_bursts} " f"RECEIVED FRAMES: {rx_total_frames} " f"RX_ERRORS: {rx_errors}", file=sys.stderr, ) frx.close() # and at last check if we had an opened audio instance and close it if AUDIO_INPUT_DEVICE != -1: sd._terminate() def parse_arguments(): # --------------------------------------------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( "--mode", dest="FREEDV_MODE", type=str, choices=["datac0", "datac1", "datac3"] ) parser.add_argument( "--audiodev", dest="AUDIO_INPUT_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", ) args, _ = parser.parse_known_args() return args if __name__ == "__main__": test_rx()