mirror of
https://github.com/DJ2LS/FreeDATA
synced 2024-05-14 08:04:33 +00:00
256 lines
8.8 KiB
Python
256 lines
8.8 KiB
Python
#!/usr/bin/env python3
|
|
# -*- coding: utf-8 -*-
|
|
"""
|
|
Created on Wed Dec 23 07:04:24 2020
|
|
|
|
@author: DJ2LS
|
|
"""
|
|
|
|
import argparse
|
|
import ctypes
|
|
import queue
|
|
import sys
|
|
import time
|
|
|
|
import numpy as np
|
|
import pyaudio
|
|
|
|
sys.path.insert(0, "..")
|
|
from modem import codec2
|
|
|
|
# --------------------------------------------GET PARAMETER INPUTS
|
|
parser = argparse.ArgumentParser(description="FreeDATA audio 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",
|
|
)
|
|
parser.add_argument(
|
|
"--testframes",
|
|
dest="TESTFRAMES",
|
|
action="store_true",
|
|
default=False,
|
|
help="generate testframes",
|
|
)
|
|
|
|
args, _ = parser.parse_known_args()
|
|
|
|
if args.LIST:
|
|
p = pyaudio.PyAudio()
|
|
for dev in range(p.get_device_count()):
|
|
print("audiodev: ", dev, p.get_device_info_by_index(dev)["name"])
|
|
sys.exit()
|
|
|
|
|
|
class Test:
|
|
def __init__(self):
|
|
|
|
self.dataqueue = queue.Queue()
|
|
self.N_BURSTS = args.N_BURSTS
|
|
self.N_FRAMES_PER_BURST = args.N_FRAMES_PER_BURST
|
|
self.AUDIO_OUTPUT_DEVICE = args.AUDIO_OUTPUT_DEVICE
|
|
self.DELAY_BETWEEN_BURSTS = args.DELAY_BETWEEN_BURSTS / 1000
|
|
|
|
# AUDIO PARAMETERS
|
|
# v-- consider increasing if you get nread_exceptions > 0
|
|
self.AUDIO_FRAMES_PER_BUFFER = 2400
|
|
self.MODEM_SAMPLE_RATE = codec2.api.FREEDV_FS_8000
|
|
self.AUDIO_SAMPLE_RATE_TX = 48000
|
|
|
|
# make sure our resampler will work
|
|
assert (
|
|
self.AUDIO_SAMPLE_RATE_TX / self.MODEM_SAMPLE_RATE
|
|
) == codec2.api.FDMDV_OS_48
|
|
|
|
self.transmit = True
|
|
|
|
self.resampler = codec2.resampler()
|
|
|
|
# check if we want to use an audio device then do a pyaudio init
|
|
if self.AUDIO_OUTPUT_DEVICE != -1:
|
|
self.p = pyaudio.PyAudio()
|
|
# auto search for loopback devices
|
|
if self.AUDIO_OUTPUT_DEVICE == -2:
|
|
loopback_list = [
|
|
dev
|
|
for dev in range(self.p.get_device_count())
|
|
if "Loopback: PCM" in self.p.get_device_info_by_index(dev)["name"]
|
|
]
|
|
|
|
if len(loopback_list) >= 2:
|
|
self.AUDIO_OUTPUT_DEVICE = loopback_list[0] # 0 = RX 1 = TX
|
|
print(f"loopback_list rx: {loopback_list}", file=sys.stderr)
|
|
else:
|
|
sys.exit()
|
|
|
|
print(
|
|
f"AUDIO OUTPUT DEVICE: {self.AUDIO_OUTPUT_DEVICE} "
|
|
f"DEVICE: {self.p.get_device_info_by_index(self.AUDIO_OUTPUT_DEVICE)['name']} "
|
|
f"AUDIO SAMPLE RATE: {self.AUDIO_SAMPLE_RATE_TX}",
|
|
file=sys.stderr,
|
|
)
|
|
|
|
self.stream_tx = self.p.open(
|
|
format=pyaudio.paInt16,
|
|
channels=1,
|
|
rate=self.AUDIO_SAMPLE_RATE_TX,
|
|
frames_per_buffer=self.AUDIO_FRAMES_PER_BUFFER,
|
|
input=False,
|
|
output=True,
|
|
output_device_index=self.AUDIO_OUTPUT_DEVICE,
|
|
stream_callback=self.callback,
|
|
)
|
|
else:
|
|
print("test_callback_multimode_tx: Not written for STDOUT usage.")
|
|
print("Exiting.")
|
|
sys.exit()
|
|
|
|
# Copy received 48 kHz to a file. Listen to this file with:
|
|
# aplay -r 48000 -f S16_LE rx48_callback.raw
|
|
# Corruption of this file is a good way to detect audio card issues
|
|
self.ftx = open("tx48_callback.raw", mode="wb")
|
|
|
|
# data binary string
|
|
if args.TESTFRAMES:
|
|
self.data_out = bytearray(14)
|
|
self.data_out[:1] = bytes([255])
|
|
self.data_out[1:2] = bytes([1])
|
|
self.data_out[2:] = b"HELLO WORLD"
|
|
|
|
else:
|
|
self.data_out = b"HELLO WORLD!"
|
|
|
|
def callback(self, data_in48k, frame_count, time_info, status):
|
|
|
|
data_out48k = self.dataqueue.get()
|
|
return (data_out48k, pyaudio.paContinue)
|
|
|
|
def run_audio(self):
|
|
try:
|
|
print("starting pyaudio callback", file=sys.stderr)
|
|
self.stream_tx.start_stream()
|
|
except Exception as e:
|
|
print(f"pyAudio error: {e}", file=sys.stderr)
|
|
|
|
sheeps = 0
|
|
while self.transmit:
|
|
time.sleep(1)
|
|
sheeps = sheeps + 1
|
|
print(f"counting sheeps...{sheeps}")
|
|
|
|
self.ftx.close()
|
|
|
|
# close pyaudio instance
|
|
self.stream_tx.close()
|
|
self.p.terminate()
|
|
|
|
def create_modulation(self):
|
|
|
|
modes = [
|
|
codec2.FREEDV_MODE.datac13.value,
|
|
codec2.FREEDV_MODE.datac1.value,
|
|
codec2.FREEDV_MODE.datac3.value,
|
|
]
|
|
for m in modes:
|
|
|
|
freedv = ctypes.cast(codec2.api.freedv_open(m), ctypes.c_void_p)
|
|
|
|
n_tx_modem_samples = codec2.api.freedv_get_n_tx_modem_samples(freedv)
|
|
mod_out = ctypes.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 = ctypes.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 = ctypes.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
|
|
|
|
buffer = bytearray(payload_per_frame)
|
|
# set buffer size to length of data which will be sent
|
|
buffer[: len(self.data_out)] = self.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, self.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, self.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)
|
|
print(
|
|
f"GENERATING TX BURST: {i}/{self.N_BURSTS} FRAME: {n}/{self.N_FRAMES_PER_BURST}",
|
|
file=sys.stderr,
|
|
)
|
|
|
|
# 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(self.MODEM_SAMPLE_RATE * self.DELAY_BETWEEN_BURSTS)
|
|
mod_out_silence = ctypes.create_string_buffer(samples_delay * 2)
|
|
txbuffer += bytes(mod_out_silence)
|
|
|
|
# resample up to 48k (resampler works on np.int16)
|
|
x = np.frombuffer(txbuffer, dtype=np.int16)
|
|
txbuffer_48k = self.resampler.resample8_to_48(x)
|
|
|
|
# split modulated audio to chunks
|
|
# https://newbedev.com/how-to-split-a-byte-string-into-separate-bytes-in-python
|
|
txbuffer_48k = bytes(txbuffer_48k)
|
|
chunk = [
|
|
txbuffer_48k[i : i + self.AUDIO_FRAMES_PER_BUFFER * 2]
|
|
for i in range(
|
|
0, len(txbuffer_48k), self.AUDIO_FRAMES_PER_BUFFER * 2
|
|
)
|
|
]
|
|
# add modulated chunks to fifo buffer
|
|
for c in chunk:
|
|
# if data is shorter than the expcected audio frames per buffer we need to append 0
|
|
# to prevent the callback from stucking/crashing
|
|
if len(c) < self.AUDIO_FRAMES_PER_BUFFER * 2:
|
|
c += bytes(self.AUDIO_FRAMES_PER_BUFFER * 2 - len(c))
|
|
self.dataqueue.put(c)
|
|
|
|
|
|
test = Test()
|
|
test.create_modulation()
|
|
test.run_audio()
|