FreeDATA/test/test_callback_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

184 lines
7.4 KiB
Python

#!/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 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('--mode', dest="FREEDV_MODE", type=str, choices=['datac0', 'datac1', 'datac3'])
parser.add_argument('--audiodev', dest="AUDIO_INPUT_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")
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.N_BURSTS = args.N_BURSTS
self.N_FRAMES_PER_BURST = args.N_FRAMES_PER_BURST
self.AUDIO_INPUT_DEVICE = args.AUDIO_INPUT_DEVICE
self.MODE = codec2.FREEDV_MODE[args.FREEDV_MODE].value
self.DEBUGGING_MODE = args.DEBUGGING_MODE
self.TIMEOUT = args.TIMEOUT
# AUDIO PARAMETERS
self.AUDIO_FRAMES_PER_BUFFER = 2400*2 # <- consider increasing if you get nread_exceptions > 0
self.MODEM_SAMPLE_RATE = codec2.api.FREEDV_FS_8000
self.AUDIO_SAMPLE_RATE_RX = 48000
# make sure our resampler will work
assert (self.AUDIO_SAMPLE_RATE_RX / self.MODEM_SAMPLE_RATE) == codec2.api.FDMDV_OS_48
# check if we want to use an audio device then do an pyaudio init
if self.AUDIO_INPUT_DEVICE != -1:
self.p = pyaudio.PyAudio()
# auto search for loopback devices
if self.AUDIO_INPUT_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_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: {self.AUDIO_INPUT_DEVICE} DEVICE: {self.p.get_device_info_by_index(self.AUDIO_INPUT_DEVICE)['name']} \
AUDIO SAMPLE RATE: {self.AUDIO_SAMPLE_RATE_RX}", file=sys.stderr)
self.stream_rx = self.p.open(format=pyaudio.paInt16,
channels=1,
rate=self.AUDIO_SAMPLE_RATE_RX,
frames_per_buffer=self.AUDIO_FRAMES_PER_BUFFER,
input=True,
output=False,
input_device_index=self.AUDIO_INPUT_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)
self.total_n_bytes = 0
self.rx_total_frames = 0
self.rx_frames = 0
self.rx_bursts = 0
self.rx_errors = 0
self.nread_exceptions = 0
self.timeout = time.time() + self.TIMEOUT
self.receive = True
self.audio_buffer = codec2.audio_buffer(self.AUDIO_FRAMES_PER_BUFFER*2)
self.resampler = codec2.resampler()
# 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.frx = open("rx48_callback.raw", mode='wb')
def callback(self, data_in48k, frame_count, time_info, status):
x = np.frombuffer(data_in48k, dtype=np.int16)
x.tofile(self.frx)
x = self.resampler.resample48_to_8(x)
self.audio_buffer.push(x)
# when we have enough samples call FreeDV Rx
nin = codec2.api.freedv_nin(self.freedv)
while self.audio_buffer.nbuffer >= nin:
# demodulate audio
nbytes = codec2.api.freedv_rawdatarx(self.freedv, self.bytes_out, self.audio_buffer.buffer.ctypes)
self.audio_buffer.pop(nin)
# call me on every loop!
nin = codec2.api.freedv_nin(self.freedv)
rx_status = codec2.api.freedv_get_rx_status(self.freedv)
if rx_status & codec2.api.FREEDV_RX_BIT_ERRORS:
self.rx_errors = self.rx_errors + 1
if self.DEBUGGING_MODE:
rx_status = codec2.api.rx_sync_flags_to_text[rx_status]
print("nin: %5d rx_status: %4s naudio_buffer: %4d" % \
(nin,rx_status,self.audio_buffer.nbuffer), file=sys.stderr)
if nbytes:
self.total_n_bytes = self.total_n_bytes + nbytes
if nbytes == self.bytes_per_frame:
self.rx_total_frames = self.rx_total_frames + 1
self.rx_frames = self.rx_frames + 1
if self.rx_frames == self.N_FRAMES_PER_BURST:
self.rx_frames = 0
self.rx_bursts = self.rx_bursts + 1
if self.rx_bursts == self.N_BURSTS:
self.receive = False
return (None, pyaudio.paContinue)
def run_audio(self):
try:
print(f"starting pyaudio callback", file=sys.stderr)
self.stream_rx.start_stream()
except Exception as e:
print(f"pyAudio error: {e}", file=sys.stderr)
while self.receive and time.time() < self.timeout:
time.sleep(1)
if time.time() >= self.timeout:
print("TIMEOUT REACHED")
if self.nread_exceptions:
print("nread_exceptions %d - receive audio lost! Consider increasing Pyaudio frames_per_buffer..." % \
self.nread_exceptions, file=sys.stderr)
print(f"RECEIVED BURSTS: {self.rx_bursts} RECEIVED FRAMES: {self.rx_total_frames} RX_ERRORS: {self.rx_errors}", file=sys.stderr)
self.frx.close()
# cloese pyaudio instance
self.stream_rx.close()
self.p.terminate()
test = Test()
test.run_audio()