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transmitting in callback mode

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used a FIFO queue to store modulation...crazy somehow. But its working
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dj2ls 2021-12-22 10:31:21 +01:00
parent 05e65018d8
commit 24f46204ea
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test/test_callback_tx.py Normal file
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#!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""
Created on Wed Dec 23 07:04:24 2020
@author: DJ2LS
"""
import ctypes
from ctypes import *
import pathlib
import pyaudio
import sys
import logging
import time
import threading
import sys
import argparse
import queue
import numpy as np
sys.path.insert(0,'..')
from tnc 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('--mode', dest="FREEDV_MODE", type=str, choices=['datac0', 'datac1', 'datac3'])
parser.add_argument('--audiodev', dest="AUDIO_OUTPUT_DEVICE", default=-1, type=int,
help="audio device number to use, use -2 to automatically select a loopback device")
parser.add_argument('--debug', dest="DEBUGGING_MODE", action="store_true")
parser.add_argument('--timeout', dest="TIMEOUT", default=10, type=int, help="Timeout (seconds) before test ends")
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_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()
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.MODE = codec2.FREEDV_MODE[args.FREEDV_MODE].value
self.DEBUGGING_MODE = args.DEBUGGING_MODE
self.TIMEOUT = args.TIMEOUT
self.DELAY_BETWEEN_BURSTS = args.DELAY_BETWEEN_BURSTS/1000
# AUDIO PARAMETERS
self.AUDIO_FRAMES_PER_BUFFER = 2400 # <- consider increasing if you get nread_exceptions > 0
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 an 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 = []
for dev in range(0,self.p.get_device_count()):
if 'Loopback: PCM' in self.p.get_device_info_by_index(dev)["name"]:
loopback_list.append(dev)
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:
quit()
print(f"AUDIO OUTPUT DEVICE: {self.AUDIO_OUTPUT_DEVICE} DEVICE: {self.p.get_device_info_by_index(self.AUDIO_OUTPUT_DEVICE)['name']} \
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
)
# open codec2 instance
self.freedv = cast(codec2.api.freedv_open(self.MODE), c_void_p)
# get number of bytes per frame for mode
self.bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(self.freedv)/8)
self.bytes_out = create_string_buffer(self.bytes_per_frame)
codec2.api.freedv_set_frames_per_burst(self.freedv,self.N_FRAMES_PER_BURST)
# 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(f"starting pyaudio callback", file=sys.stderr)
self.stream_tx.start_stream()
except Exception as e:
print(f"pyAudio error: {e}", file=sys.stderr)
while self.transmit:
time.sleep(0.1)
self.ftx.close()
# close pyaudio instance
self.stream_tx.close()
self.p.terminate()
def create_modulation(self):
# open codec2 instance
freedv = cast(codec2.api.freedv_open(self.MODE), c_void_p)
# get number of bytes per frame for mode
bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(freedv)/8)
payload_bytes_per_frame = bytes_per_frame -2
# init buffer for data
n_tx_modem_samples = codec2.api.freedv_get_n_tx_modem_samples(freedv)
mod_out = create_string_buffer(n_tx_modem_samples * 2)
# init buffer for preample
n_tx_preamble_modem_samples = codec2.api.freedv_get_n_tx_preamble_modem_samples(freedv)
mod_out_preamble = create_string_buffer(n_tx_preamble_modem_samples * 2)
# init buffer for postamble
n_tx_postamble_modem_samples = codec2.api.freedv_get_n_tx_postamble_modem_samples(freedv)
mod_out_postamble = create_string_buffer(n_tx_postamble_modem_samples * 2)
# create buffer for data
buffer = bytearray(payload_bytes_per_frame) # use this if CRC16 checksum is required ( DATA1-3)
buffer[:len(self.data_out)] = self.data_out # set buffersize to length of data which will be send
# create crc for data frame - we are using the crc function shipped with codec2 to avoid
# crc algorithm incompatibilities
crc = ctypes.c_ushort(codec2.api.freedv_gen_crc16(bytes(buffer), payload_bytes_per_frame)) # generate CRC16
crc = crc.value.to_bytes(2, byteorder='big') # convert crc to 2 byte hex string
buffer += crc # append crc16 to buffer
print(f"TOTAL BURSTS: {self.N_BURSTS} TOTAL FRAMES_PER_BURST: {self.N_FRAMES_PER_BURST}", file=sys.stderr)
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"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 = create_string_buffer(samples_delay*2)
txbuffer += bytes(mod_out_silence)
print(f"samples_delay: {samples_delay} DELAY_BETWEEN_BURSTS: {self.DELAY_BETWEEN_BURSTS}", file=sys.stderr)
# 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 modualted 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()