mirror of
https://github.com/DJ2LS/FreeDATA
synced 2024-05-14 08:04:33 +00:00
151 lines
5.7 KiB
Python
151 lines
5.7 KiB
Python
#!/usr/bin/env python3
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# -*- coding: utf-8 -*-
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import ctypes
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from ctypes import *
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import pathlib
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import pyaudio
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import time
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import threading
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import audioop
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import argparse
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import sys
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sys.path.insert(0,'..')
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import codec2
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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('--framesperburst', dest="N_FRAMES_PER_BURST", default=1, type=int)
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parser.add_argument('--delay', dest="DELAY_BETWEEN_BURSTS", default=500, type=int)
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parser.add_argument('--audiodev', dest="AUDIO_OUTPUT_DEVICE", default=-1, type=int, help="audio output device number to use")
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parser.add_argument('--list', dest="LIST", action="store_true", help="list audio devices by number and exit")
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args = parser.parse_args()
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if args.LIST:
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p = pyaudio.PyAudio()
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for dev in range(0,p.get_device_count()):
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print("audiodev: ", dev, p.get_device_info_by_index(dev)["name"])
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quit()
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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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# AUDIO PARAMETERS
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AUDIO_FRAMES_PER_BUFFER = 2048
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MODEM_SAMPLE_RATE = codec2.api.FREEDV_FS_8000
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AUDIO_SAMPLE_RATE_TX = 8000
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assert (AUDIO_SAMPLE_RATE_TX % MODEM_SAMPLE_RATE) == 0
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if AUDIO_OUTPUT_DEVICE != -1:
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p = 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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for dev in range(0,p.get_device_count()):
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if 'Loopback: PCM' in p.get_device_info_by_index(dev)["name"]:
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loopback_list.append(dev)
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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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quit()
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# pyaudio init
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stream_tx = p.open(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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modes = [codec2.api.FREEDV_MODE_DATAC0, codec2.api.FREEDV_MODE_DATAC1, codec2.api.FREEDV_MODE_DATAC3]
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for m in modes:
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freedv = cast(codec2.api.freedv_open(m), 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 = 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(freedv)
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mod_out_preamble = create_string_buffer(2*n_tx_preamble_modem_samples)
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n_tx_postamble_modem_samples = codec2.api.freedv_get_n_tx_postamble_modem_samples(freedv)
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mod_out_postamble = create_string_buffer(2*n_tx_postamble_modem_samples)
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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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# data binary string
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data_out = b'HELLO WORLD!'
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buffer = bytearray(payload_per_frame)
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# set buffersize to length of data which will be send
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buffer[:len(data_out)] = data_out
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crc = ctypes.c_ushort(codec2.api.freedv_gen_crc16(bytes(buffer), payload_per_frame)) # generate CRC16
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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 i 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 n 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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codec2.api.freedv_rawdatatx(freedv,mod_out,data) # modulate DATA and save it into mod_out pointer
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txbuffer += bytes(mod_out)
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print(f"TX BURST: {i}/{N_BURSTS} FRAME: {n}/{N_FRAMES_PER_BURST}", file=sys.stderr)
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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 = create_string_buffer(samples_delay*2)
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txbuffer += bytes(mod_out_silence)
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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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# sample rate conversion from 8000Hz to 48000Hz
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#audio = audioop.ratecv(txbuffer,2,1,MODEM_SAMPLE_RATE, AUDIO_SAMPLE_RATE_TX, None)
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stream_tx.write(txbuffer)
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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)
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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 openend 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.terminate()
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