FreeDATA/test/test_multimode_rx.py
dj2ls 82619ef098 reduced bytes_out buffer size factor *2
buffer was too large. this didnt affect the tests, but not that nice...
2021-12-20 15:57:32 +01:00

213 lines
8.6 KiB
Python
Executable file

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