FreeDATA/test/001_highsnr_stdio_audio/test_multimode_tx.py

150 lines
5.6 KiB
Python

#!/usr/bin/env python3
# -*- coding: utf-8 -*-
import ctypes
from ctypes import *
import pathlib
import pyaudio
import time
import threading
import audioop
import argparse
import sys
sys.path.insert(0,'..')
import codec2
# GET PARAMETER INPUTS
parser = argparse.ArgumentParser(description='FreeDATA TEST')
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('--delay', dest="DELAY_BETWEEN_BURSTS", default=500, type=int)
parser.add_argument('--audiodev', dest="AUDIO_OUTPUT_DEVICE", default=-1, type=int, help="audio output device number to use")
parser.add_argument('--list', dest="LIST", action="store_true", help="list audio devices by number and exit")
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
DELAY_BETWEEN_BURSTS = args.DELAY_BETWEEN_BURSTS/1000
AUDIO_OUTPUT_DEVICE = args.AUDIO_OUTPUT_DEVICE
# AUDIO PARAMETERS
AUDIO_FRAMES_PER_BUFFER = 2048
MODEM_SAMPLE_RATE = codec2.api.FREEDV_FS_8000
AUDIO_SAMPLE_RATE_TX = 8000
assert (AUDIO_SAMPLE_RATE_TX % MODEM_SAMPLE_RATE) == 0
if AUDIO_OUTPUT_DEVICE != -1:
p = pyaudio.PyAudio()
# auto search for loopback devices
if AUDIO_OUTPUT_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_OUTPUT_DEVICE = loopback_list[1] #0 = RX 1 = TX
print(f"loopback_list tx: {loopback_list}", file=sys.stderr)
else:
quit()
# pyaudio init
stream_tx = p.open(format=pyaudio.paInt16,
channels=1,
rate=AUDIO_SAMPLE_RATE_TX,
frames_per_buffer=AUDIO_FRAMES_PER_BUFFER, #n_nom_modem_samples
output=True,
output_device_index=AUDIO_OUTPUT_DEVICE,
)
modes = [codec2.api.FREEDV_MODE_DATAC0, codec2.api.FREEDV_MODE_DATAC1, codec2.api.FREEDV_MODE_DATAC3]
for m in modes:
freedv = cast(codec2.api.freedv_open(m), c_void_p)
n_tx_modem_samples = codec2.api.freedv_get_n_tx_modem_samples(freedv)
mod_out = create_string_buffer(2*n_tx_modem_samples)
n_tx_preamble_modem_samples = codec2.api.freedv_get_n_tx_preamble_modem_samples(freedv)
mod_out_preamble = create_string_buffer(2*n_tx_preamble_modem_samples)
n_tx_postamble_modem_samples = codec2.api.freedv_get_n_tx_postamble_modem_samples(freedv)
mod_out_postamble = create_string_buffer(2*n_tx_postamble_modem_samples)
bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(freedv) / 8)
payload_per_frame = bytes_per_frame - 2
# data binary string
data_out = b'HELLO WORLD!'
buffer = bytearray(payload_per_frame)
# set buffersize to length of data which will be send
buffer[:len(data_out)] = data_out
crc = ctypes.c_ushort(codec2.api.freedv_gen_crc16(bytes(buffer), payload_per_frame)) # generate CRC16
# convert crc to 2 byte hex string
crc = crc.value.to_bytes(2, byteorder='big')
buffer += crc # append crc16 to buffer
data = (ctypes.c_ubyte * bytes_per_frame).from_buffer_copy(buffer)
for i in range(1,N_BURSTS+1):
# write preamble to txbuffer
codec2.api.freedv_rawdatapreambletx(freedv, mod_out_preamble)
txbuffer = bytes(mod_out_preamble)
# create modulaton for N = FRAMESPERBURST and append it to txbuffer
for n in range(1,N_FRAMES_PER_BURST+1):
data = (ctypes.c_ubyte * bytes_per_frame).from_buffer_copy(buffer)
codec2.api.freedv_rawdatatx(freedv,mod_out,data) # modulate DATA and save it into mod_out pointer
txbuffer += bytes(mod_out)
# append postamble to txbuffer
codec2.api.freedv_rawdatapostambletx(freedv, mod_out_postamble)
txbuffer += bytes(mod_out_postamble)
# append a delay between bursts as audio silence
samples_delay = int(MODEM_SAMPLE_RATE*DELAY_BETWEEN_BURSTS)
mod_out_silence = create_string_buffer(samples_delay*2)
txbuffer += bytes(mod_out_silence)
# check if we want to use an audio device or stdout
if AUDIO_OUTPUT_DEVICE != -1:
# sample rate conversion from 8000Hz to 48000Hz
#audio = audioop.ratecv(txbuffer,2,1,MODEM_SAMPLE_RATE, AUDIO_SAMPLE_RATE_TX, None)
stream_tx.write(txbuffer)
else:
# this test needs a lot of time, so we are having a look at times...
starttime = time.time()
# print data to terminal for piping the output to other programs
sys.stdout.buffer.write(txbuffer)
sys.stdout.flush()
# and at least print the needed time to see which time we needed
timeneeded = time.time()-starttime
#print(f"time: {timeneeded} buffer: {len(txbuffer)}", file=sys.stderr)
# and at last check if we had an openend pyaudio instance and close it
if AUDIO_OUTPUT_DEVICE != -1:
time.sleep(stream_tx.get_output_latency())
stream_tx.stop_stream()
stream_tx.close()
p.terminate()