mirror of
https://github.com/DJ2LS/FreeDATA
synced 2024-05-14 08:04:33 +00:00
199 lines
6.8 KiB
Python
199 lines
6.8 KiB
Python
#!/usr/bin/env python3
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# -*- coding: utf-8 -*-
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"""
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Send-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. Tests all three codec2 data frames simultaneously.
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Invoked from CMake, test_highsnr_stdio_P_P_multi.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 pyaudio
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sys.path.insert(0, "..")
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from modem import codec2
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def test_mm_tx():
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MODEM_SAMPLE_RATE = codec2.api.FREEDV_FS_8000
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AUDIO_SAMPLE_RATE_TX = 48000
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assert (AUDIO_SAMPLE_RATE_TX % MODEM_SAMPLE_RATE) == 0
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args = parse_arguments()
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if args.LIST:
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p_audio = pyaudio.PyAudio()
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for dev in range(p_audio.get_device_count()):
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print("audiodev: ", dev, p_audio.get_device_info_by_index(dev)["name"])
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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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DELAY_BETWEEN_BURSTS = args.DELAY_BETWEEN_BURSTS / 1000
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AUDIO_OUTPUT_DEVICE = args.AUDIO_OUTPUT_DEVICE
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resampler = codec2.resampler()
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# Data binary string
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data_out = b"HELLO WORLD!"
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modes = [
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codec2.FREEDV_MODE.datac13.value,
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codec2.FREEDV_MODE.datac1.value,
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codec2.FREEDV_MODE.datac3.value,
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]
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if AUDIO_OUTPUT_DEVICE != -1:
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p_audio = pyaudio.PyAudio()
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# Auto search for loopback devices
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if AUDIO_OUTPUT_DEVICE == -2:
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loopback_list = [
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dev
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for dev in range(p_audio.get_device_count())
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if "Loopback: PCM" in p_audio.get_device_info_by_index(dev)["name"]
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]
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if len(loopback_list) >= 2:
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AUDIO_OUTPUT_DEVICE = loopback_list[1] # 0 = RX 1 = TX
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print(f"loopback_list tx: {loopback_list}", file=sys.stderr)
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else:
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sys.exit()
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# AUDIO PARAMETERS
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AUDIO_FRAMES_PER_BUFFER = 2400
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# pyaudio init
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stream_tx = p_audio.open(
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format=pyaudio.paInt16,
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channels=1,
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rate=AUDIO_SAMPLE_RATE_TX,
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frames_per_buffer=AUDIO_FRAMES_PER_BUFFER, # n_nom_modem_samples
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output=True,
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output_device_index=AUDIO_OUTPUT_DEVICE,
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)
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for mode in modes:
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freedv = ctypes.cast(codec2.api.freedv_open(mode), ctypes.c_void_p)
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n_tx_modem_samples = codec2.api.freedv_get_n_tx_modem_samples(freedv)
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mod_out = ctypes.create_string_buffer(2 * n_tx_modem_samples)
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n_tx_preamble_modem_samples = codec2.api.freedv_get_n_tx_preamble_modem_samples(
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freedv
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)
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mod_out_preamble = ctypes.create_string_buffer(2 * n_tx_preamble_modem_samples)
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n_tx_postamble_modem_samples = (
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codec2.api.freedv_get_n_tx_postamble_modem_samples(freedv)
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)
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mod_out_postamble = ctypes.create_string_buffer(
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2 * n_tx_postamble_modem_samples
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)
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bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(freedv) / 8)
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payload_per_frame = bytes_per_frame - 2
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buffer = bytearray(payload_per_frame)
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# Set buffer size to length of data which will be sent
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buffer[: len(data_out)] = data_out
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# Generate CRC16
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crc = ctypes.c_ushort(
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codec2.api.freedv_gen_crc16(bytes(buffer), payload_per_frame)
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)
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# Convert CRC to 2 byte hex string
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crc = crc.value.to_bytes(2, byteorder="big")
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buffer += crc # Append crc16 to buffer
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data = (ctypes.c_ubyte * bytes_per_frame).from_buffer_copy(buffer)
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for brst in range(1, N_BURSTS + 1):
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# Write preamble to txbuffer
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codec2.api.freedv_rawdatapreambletx(freedv, mod_out_preamble)
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txbuffer = bytes(mod_out_preamble)
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# Create modulaton for N = FRAMESPERBURST and append it to txbuffer
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for frm in range(1, N_FRAMES_PER_BURST + 1):
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data = (ctypes.c_ubyte * bytes_per_frame).from_buffer_copy(buffer)
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# Modulate DATA and save it into mod_out pointer
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codec2.api.freedv_rawdatatx(freedv, mod_out, data)
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txbuffer += bytes(mod_out)
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print(
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f"TX BURST: {brst}/{N_BURSTS} FRAME: {frm}/{N_FRAMES_PER_BURST}",
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file=sys.stderr,
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)
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# Append postamble to txbuffer
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codec2.api.freedv_rawdatapostambletx(freedv, mod_out_postamble)
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txbuffer += bytes(mod_out_postamble)
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# Append a delay between bursts as audio silence
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samples_delay = int(MODEM_SAMPLE_RATE * DELAY_BETWEEN_BURSTS)
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mod_out_silence = ctypes.create_string_buffer(samples_delay * 2)
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txbuffer += bytes(mod_out_silence)
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# Resample up to 48k (resampler works on np.int16)
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audio_buffer = np.frombuffer(txbuffer, dtype=np.int16)
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txbuffer_48k = resampler.resample8_to_48(audio_buffer)
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# Check if we want to use an audio device or stdout
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if AUDIO_OUTPUT_DEVICE != -1:
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stream_tx.write(txbuffer_48k.tobytes())
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else:
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# This test needs a lot of time, so we are having a look at times...
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starttime = time.time()
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# Print data to terminal for piping the output to other programs
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sys.stdout.buffer.write(txbuffer_48k)
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sys.stdout.flush()
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# and at least print the needed time to see which time we needed
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timeneeded = time.time() - starttime
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# print(f"time: {timeneeded} buffer: {len(txbuffer)}", file=sys.stderr)
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# and at last check if we had an opened pyaudio instance and close it
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if AUDIO_OUTPUT_DEVICE != -1:
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time.sleep(stream_tx.get_output_latency())
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stream_tx.stop_stream()
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stream_tx.close()
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p_audio.terminate()
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def parse_arguments():
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# GET PARAMETER INPUTS
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parser = argparse.ArgumentParser(description="FreeDATA TEST")
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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("--delay", dest="DELAY_BETWEEN_BURSTS", default=500, type=int)
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parser.add_argument(
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"--audiodev",
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dest="AUDIO_OUTPUT_DEVICE",
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default=-1,
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type=int,
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help="audio output device number to use",
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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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test_mm_tx()
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