mirror of
https://github.com/DJ2LS/FreeDATA
synced 2024-05-14 08:04:33 +00:00
213 lines
8.6 KiB
Python
Executable file
213 lines
8.6 KiB
Python
Executable file
#!/usr/bin/env python3
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# -*- coding: utf-8 -*-
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import pyaudio
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import audioop
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import time
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import argparse
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import sys
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import ctypes
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from ctypes import *
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import pathlib
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sys.path.insert(0,'..')
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from tnc import codec2
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import numpy as np
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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('--framesperburst', dest="N_FRAMES_PER_BURST", default=1, type=int)
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parser.add_argument('--audiodev', dest="AUDIO_INPUT_DEVICE", default=-1, type=int, help="audio device number to use")
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parser.add_argument('--debug', dest="DEBUGGING_MODE", action="store_true")
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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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parser.add_argument('--timeout', dest="TIMEOUT", default=10, type=int, help="Timeout (seconds) before test ends")
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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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AUDIO_INPUT_DEVICE = args.AUDIO_INPUT_DEVICE
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DEBUGGING_MODE = args.DEBUGGING_MODE
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TIMEOUT = args.TIMEOUT
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# AUDIO PARAMETERS
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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
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# SET COUNTERS
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rx_total_frames_datac0 = 0
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rx_frames_datac0 = 0
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rx_bursts_datac0 = 0
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rx_total_frames_datac1 = 0
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rx_frames_datac1 = 0
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rx_bursts_datac1 = 0
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rx_total_frames_datac3 = 0
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rx_frames_datac3 = 0
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rx_bursts_datac3 = 0
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# open codec2 instance
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datac0_freedv = cast(codec2.api.freedv_open(codec2.api.FREEDV_MODE_DATAC0), c_void_p)
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datac0_bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(datac0_freedv)/8)
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datac0_bytes_out = create_string_buffer(datac0_bytes_per_frame)
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codec2.api.freedv_set_frames_per_burst(datac0_freedv,N_FRAMES_PER_BURST)
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datac0_buffer = codec2.audio_buffer(2*AUDIO_FRAMES_PER_BUFFER)
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datac1_freedv = cast(codec2.api.freedv_open(codec2.api.FREEDV_MODE_DATAC1), c_void_p)
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datac1_bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(datac1_freedv)/8)
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datac1_bytes_out = create_string_buffer(datac1_bytes_per_frame)
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codec2.api.freedv_set_frames_per_burst(datac1_freedv,N_FRAMES_PER_BURST)
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datac1_buffer = codec2.audio_buffer(2*AUDIO_FRAMES_PER_BUFFER)
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datac3_freedv = cast(codec2.api.freedv_open(codec2.api.FREEDV_MODE_DATAC3), c_void_p)
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datac3_bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(datac3_freedv)/8)
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datac3_bytes_out = create_string_buffer(datac3_bytes_per_frame)
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codec2.api.freedv_set_frames_per_burst(datac3_freedv,N_FRAMES_PER_BURST)
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datac3_buffer = codec2.audio_buffer(2*AUDIO_FRAMES_PER_BUFFER)
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resampler = codec2.resampler()
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# check if we want to use an audio device then do an pyaudio init
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if AUDIO_INPUT_DEVICE != -1:
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p = pyaudio.PyAudio()
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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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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_INPUT_DEVICE = loopback_list[0] #0 = RX 1 = TX
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print(f"loopback_list rx: {loopback_list}", file=sys.stderr)
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else:
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quit()
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print(f"AUDIO INPUT DEVICE: {AUDIO_INPUT_DEVICE} DEVICE: {p.get_device_info_by_index(AUDIO_INPUT_DEVICE)['name']} AUDIO SAMPLE RATE: {AUDIO_SAMPLE_RATE_RX}", file=sys.stderr)
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stream_rx = p.open(format=pyaudio.paInt16,
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channels=1,
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rate=AUDIO_SAMPLE_RATE_RX,
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frames_per_buffer=AUDIO_FRAMES_PER_BUFFER,
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input=True,
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input_device_index=AUDIO_INPUT_DEVICE
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)
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timeout = time.time() + TIMEOUT
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print(time.time(),TIMEOUT, timeout)
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receive = True
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nread_exceptions = 0
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# initial nin values
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datac0_nin = codec2.api.freedv_nin(datac0_freedv)
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datac1_nin = codec2.api.freedv_nin(datac1_freedv)
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datac3_nin = codec2.api.freedv_nin(datac3_freedv)
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def print_stats():
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if DEBUGGING_MODE:
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datac0_rxstatus = codec2.api.freedv_get_rx_status(datac0_freedv)
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datac0_rxstatus = codec2.api.rx_sync_flags_to_text[datac0_rxstatus]
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datac1_rxstatus = codec2.api.freedv_get_rx_status(datac1_freedv)
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datac1_rxstatus = codec2.api.rx_sync_flags_to_text[datac1_rxstatus]
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datac3_rxstatus = codec2.api.freedv_get_rx_status(datac3_freedv)
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datac3_rxstatus = codec2.api.rx_sync_flags_to_text[datac3_rxstatus]
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print("NIN0: %5d RX_STATUS0: %4s NIN1: %5d RX_STATUS1: %4s NIN3: %5d RX_STATUS3: %4s" % \
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(datac0_nin, datac0_rxstatus, datac1_nin, datac1_rxstatus, datac3_nin, datac3_rxstatus),
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file=sys.stderr)
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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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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)
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if len(x) != AUDIO_FRAMES_PER_BUFFER:
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print("len(x)",len(x))
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receive = False
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x = resampler.resample48_to_8(x)
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datac0_buffer.push(x)
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datac1_buffer.push(x)
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datac3_buffer.push(x)
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print_something = False
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while datac0_buffer.nbuffer >= datac0_nin:
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# demodulate audio
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nbytes = codec2.api.freedv_rawdatarx(datac0_freedv, datac0_bytes_out, datac0_buffer.buffer.ctypes)
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datac0_buffer.pop(datac0_nin)
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datac0_nin = codec2.api.freedv_nin(datac0_freedv)
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if nbytes == datac0_bytes_per_frame:
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rx_total_frames_datac0 = rx_total_frames_datac0 + 1
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rx_frames_datac0 = rx_frames_datac0 + 1
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if rx_frames_datac0 == N_FRAMES_PER_BURST:
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rx_frames_datac0 = 0
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rx_bursts_datac0 = rx_bursts_datac0 + 1
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print_stats()
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while datac1_buffer.nbuffer >= datac1_nin:
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# demodulate audio
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nbytes = codec2.api.freedv_rawdatarx(datac1_freedv, datac1_bytes_out, datac1_buffer.buffer.ctypes)
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datac1_buffer.pop(datac1_nin)
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datac1_nin = codec2.api.freedv_nin(datac1_freedv)
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if nbytes == datac1_bytes_per_frame:
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rx_total_frames_datac1 = rx_total_frames_datac1 + 1
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rx_frames_datac1 = rx_frames_datac1 + 1
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if rx_frames_datac1 == N_FRAMES_PER_BURST:
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rx_frames_datac1 = 0
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rx_bursts_datac1 = rx_bursts_datac1 + 1
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print_stats()
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while datac3_buffer.nbuffer >= datac3_nin:
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# demodulate audio
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nbytes = codec2.api.freedv_rawdatarx(datac3_freedv, datac3_bytes_out, datac3_buffer.buffer.ctypes)
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datac3_buffer.pop(datac3_nin)
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datac3_nin = codec2.api.freedv_nin(datac3_freedv)
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if nbytes == datac3_bytes_per_frame:
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rx_total_frames_datac3 = rx_total_frames_datac3 + 1
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rx_frames_datac3 = rx_frames_datac3 + 1
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if rx_frames_datac3 == N_FRAMES_PER_BURST:
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rx_frames_datac3 = 0
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rx_bursts_datac3 = rx_bursts_datac3 + 1
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print_stats()
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if rx_bursts_datac0 == N_BURSTS and rx_bursts_datac1 == N_BURSTS and rx_bursts_datac3 == N_BURSTS:
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receive = False
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if nread_exceptions:
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print("nread_exceptions %d - receive audio lost! Consider increasing Pyaudio frames_per_buffer..." % \
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nread_exceptions, file=sys.stderr)
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# INFO IF WE REACHED TIMEOUT
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if time.time() > timeout:
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print(f"TIMEOUT REACHED", file=sys.stderr)
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print(f"DATAC0: {rx_bursts_datac0}/{rx_total_frames_datac0} DATAC1: {rx_bursts_datac1}/{rx_total_frames_datac1} DATAC3: {rx_bursts_datac3}/{rx_total_frames_datac3}", file=sys.stderr)
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if AUDIO_INPUT_DEVICE != -1:
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stream_rx.close()
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p.terminate()
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