mirror of
https://github.com/DJ2LS/FreeDATA
synced 2024-05-14 08:04:33 +00:00
259 lines
8.1 KiB
Python
259 lines
8.1 KiB
Python
#!/usr/bin/env python3
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# -*- coding: utf-8 -*-
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"""
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Receive-side station emulator for test frame tests over a high quality audio channel
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using a physical sound card or STDIO.
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Legacy test for sending / receiving connection test frames through the codec2 and
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back through on the other station. Data injection initiates directly through
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the codec2 API.
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Invoked from CMake, test_highsnr_stdio_{P_C, P_P}_datacx.py, and many test_virtual[1-3]*.sh.
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@author: DJ2LS
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"""
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import argparse
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import ctypes
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import sys
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import time
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import numpy as np
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import sounddevice as sd
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# pylint: disable=wrong-import-position
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sys.path.insert(0, "..")
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sys.path.insert(0, "../tnc")
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from tnc import codec2
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def util_rx():
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args = parse_arguments()
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if args.LIST:
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devices = sd.query_devices(device=None, kind=None)
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for index, device in enumerate(devices):
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print(f"{index} {device['name']}")
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index += 1
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# pylint: disable=protected-access
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sd._terminate()
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sys.exit()
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N_BURSTS = args.N_BURSTS
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N_FRAMES_PER_BURST = args.N_FRAMES_PER_BURST
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AUDIO_INPUT_DEVICE = args.AUDIO_INPUT_DEVICE
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MODE = codec2.FREEDV_MODE[args.FREEDV_MODE].value
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DEBUGGING_MODE = args.DEBUGGING_MODE
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MAX_TIME = args.TIMEOUT
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# AUDIO PARAMETERS
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# v-- consider increasing if you get nread_exceptions > 0
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AUDIO_FRAMES_PER_BUFFER = 2400 * 2
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MODEM_SAMPLE_RATE = codec2.api.FREEDV_FS_8000
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AUDIO_SAMPLE_RATE_RX = 48000
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# make sure our resampler will work
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assert (AUDIO_SAMPLE_RATE_RX / MODEM_SAMPLE_RATE) == codec2.api.FDMDV_OS_48 # type: ignore
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# check if we want to use an audio device then do a pyaudio init
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if AUDIO_INPUT_DEVICE != -1:
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# auto search for loopback devices
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if AUDIO_INPUT_DEVICE == -2:
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loopback_list = []
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devices = sd.query_devices(device=None, kind=None)
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for index, device in enumerate(devices):
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if "Loopback: PCM" in device["name"]:
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print(index)
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loopback_list.append(index)
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if loopback_list:
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# 0 = RX 1 = TX
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AUDIO_INPUT_DEVICE = loopback_list[0]
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print(f"loopback_list tx: {loopback_list}", file=sys.stderr)
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else:
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print("not enough audio loopback devices ready...")
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print("you should wait about 30 seconds...")
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sd._terminate()
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sys.exit()
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print(f"AUDIO INPUT DEVICE: {AUDIO_INPUT_DEVICE}", file=sys.stderr)
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# audio stream init
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stream_rx = sd.RawStream(
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channels=1,
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dtype="int16",
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device=AUDIO_INPUT_DEVICE,
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samplerate=AUDIO_SAMPLE_RATE_RX,
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blocksize=4800,
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)
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stream_rx.start()
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# ----------------------------------------------------------------
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# DATA CHANNEL INITIALISATION
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# open codec2 instance
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freedv = ctypes.cast(codec2.api.freedv_open(MODE), ctypes.c_void_p)
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# get number of bytes per frame for mode
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bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(freedv) / 8)
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payload_bytes_per_frame = bytes_per_frame - 2
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n_max_modem_samples = codec2.api.freedv_get_n_max_modem_samples(freedv)
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bytes_out = ctypes.create_string_buffer(bytes_per_frame)
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codec2.api.freedv_set_frames_per_burst(freedv, N_FRAMES_PER_BURST)
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total_n_bytes = 0
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rx_total_frames = 0
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rx_frames = 0
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rx_bursts = 0
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rx_errors = 0
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nread_exceptions = 0
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timeout = time.time() + MAX_TIME
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receive = True
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audio_buffer = codec2.audio_buffer(AUDIO_FRAMES_PER_BUFFER * 2)
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resampler = codec2.resampler()
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# time meassurement
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time_start = 0
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time_end = 0
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# Copy received 48 kHz to a file. Listen to this file with:
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# aplay -r 48000 -f S16_LE rx48.raw
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# Corruption of this file is a good way to detect audio card issues
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frx = open("rx48.raw", mode="wb")
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# initial number of samples we need
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nin = codec2.api.freedv_nin(freedv)
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while receive and time.time() < timeout:
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if AUDIO_INPUT_DEVICE != -1:
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try:
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# data_in48k = stream_rx.read(AUDIO_FRAMES_PER_BUFFER, exception_on_overflow = True)
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data_in48k, overflowed = stream_rx.read(AUDIO_FRAMES_PER_BUFFER) # type: ignore
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except OSError as err:
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print(err, file=sys.stderr)
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# if str(err).find("Input overflowed") != -1:
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# nread_exceptions += 1
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# if str(err).find("Stream closed") != -1:
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# print("Ending...")
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# receive = False
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else:
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data_in48k = sys.stdin.buffer.read(AUDIO_FRAMES_PER_BUFFER * 2)
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# insert samples in buffer
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x = np.frombuffer(data_in48k, dtype=np.int16) # type: ignore
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# print(x)
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# x = data_in48k
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x.tofile(frx)
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if len(x) != AUDIO_FRAMES_PER_BUFFER:
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receive = False
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x = resampler.resample48_to_8(x)
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audio_buffer.push(x)
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# when we have enough samples call FreeDV Rx
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while audio_buffer.nbuffer >= nin:
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# start time measurement
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time_start = time.time()
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# demodulate audio
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nbytes = codec2.api.freedv_rawdatarx(
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freedv, bytes_out, audio_buffer.buffer.ctypes
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)
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time_end = time.time()
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audio_buffer.pop(nin)
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# call me on every loop!
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nin = codec2.api.freedv_nin(freedv)
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rx_status = codec2.api.freedv_get_rx_status(freedv)
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if rx_status & codec2.api.FREEDV_RX_BIT_ERRORS:
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rx_errors = rx_errors + 1
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if DEBUGGING_MODE:
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rx_status = codec2.api.rx_sync_flags_to_text[rx_status] # type: ignore
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time_needed = time_end - time_start
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print(
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f"nin: {nin:5d} rx_status: {rx_status:4s} "
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f"naudio_buffer: {audio_buffer.nbuffer:4d} time: {time_needed:4f}",
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file=sys.stderr,
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)
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if nbytes:
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total_n_bytes += nbytes
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if nbytes == bytes_per_frame:
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rx_total_frames += 1
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rx_frames += 1
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if rx_frames == N_FRAMES_PER_BURST:
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rx_frames = 0
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rx_bursts += 1
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if rx_bursts == N_BURSTS:
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receive = False
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if time.time() >= timeout:
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print("TIMEOUT REACHED")
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time.sleep(0.01)
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if nread_exceptions:
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print(
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f"nread_exceptions {nread_exceptions:d} - receive audio lost! "
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"Consider increasing Pyaudio frames_per_buffer...",
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file=sys.stderr,
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)
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print(
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f"RECEIVED BURSTS: {rx_bursts} "
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f"RECEIVED FRAMES: {rx_total_frames} "
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f"RX_ERRORS: {rx_errors}",
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file=sys.stderr,
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)
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frx.close()
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# and at last check if we had an opened audio instance and close it
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if AUDIO_INPUT_DEVICE != -1:
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sd._terminate()
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def parse_arguments():
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# --------------------------------------------GET PARAMETER INPUTS
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parser = argparse.ArgumentParser(description="Simons TEST TNC")
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parser.add_argument("--bursts", dest="N_BURSTS", default=1, type=int)
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parser.add_argument(
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"--framesperburst", dest="N_FRAMES_PER_BURST", default=1, type=int
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)
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parser.add_argument(
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"--mode", dest="FREEDV_MODE", type=str, choices=["datac0", "datac1", "datac3"]
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)
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parser.add_argument(
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"--audiodev",
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dest="AUDIO_INPUT_DEVICE",
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default=-1,
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type=int,
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help="audio device number to use, use -2 to automatically select a loopback device",
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)
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parser.add_argument("--debug", dest="DEBUGGING_MODE", action="store_true")
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parser.add_argument(
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"--timeout",
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dest="TIMEOUT",
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default=10,
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type=int,
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help="Timeout (seconds) before test ends",
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)
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parser.add_argument(
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"--list",
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dest="LIST",
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action="store_true",
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help="list audio devices by number and exit",
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)
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args, _ = parser.parse_known_args()
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return args
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if __name__ == "__main__":
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util_rx()
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