FreeDATA/test/001_highsnr_stdio_audio/test_rx.py

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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 numpy as np
sys.path.insert(0,'../..')
from tnc import codec2
#--------------------------------------------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('--mode', dest="FREEDV_MODE", type=str, choices=['datac0', 'datac1', 'datac3'])
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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")
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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()
N_BURSTS = args.N_BURSTS
N_FRAMES_PER_BURST = args.N_FRAMES_PER_BURST
AUDIO_INPUT_DEVICE = args.AUDIO_INPUT_DEVICE
MODE = codec2.FREEDV_MODE[args.FREEDV_MODE].value
DEBUGGING_MODE = args.DEBUGGING_MODE
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TIMEOUT = args.TIMEOUT
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# AUDIO PARAMETERS
AUDIO_FRAMES_PER_BUFFER = 2400*2 # <- consider increasing if you get nread_exceptions > 0
MODEM_SAMPLE_RATE = codec2.api.FREEDV_FS_8000
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AUDIO_SAMPLE_RATE_RX = 48000
# make sure our resampler will work
assert (AUDIO_SAMPLE_RATE_RX / MODEM_SAMPLE_RATE) == codec2.api.FDMDV_OS_48
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# check if we want to use an audio device then do an pyaudio init
if AUDIO_INPUT_DEVICE != -1:
p = pyaudio.PyAudio()
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# 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()
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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
)
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# ----------------------------------------------------------------
# DATA CHANNEL INITIALISATION
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# open codec2 instance
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freedv = cast(codec2.api.freedv_open(MODE), c_void_p)
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# get number of bytes per frame for mode
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bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(freedv)/8)
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payload_bytes_per_frame = bytes_per_frame -2
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n_max_modem_samples = codec2.api.freedv_get_n_max_modem_samples(freedv)
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bytes_out = create_string_buffer(bytes_per_frame * 2)
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codec2.api.freedv_set_frames_per_burst(freedv,N_FRAMES_PER_BURST)
total_n_bytes = 0
rx_total_frames = 0
rx_frames = 0
rx_bursts = 0
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rx_errors = 0
nread_exceptions = 0
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timeout = time.time() + TIMEOUT
receive = True
audio_buffer = codec2.audio_buffer(AUDIO_FRAMES_PER_BUFFER*2)
resampler = codec2.resampler()
# Copy received 48 kHz to a file. Listen to this file with:
# aplay -r 48000 -f S16_LE rx48.raw
# Corruption of this file is a good way to detect audio card issues
frx = open("rx48.raw", mode='wb')
# initial number of samples we need
nin = codec2.api.freedv_nin(freedv)
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:
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data_in48k = sys.stdin.buffer.read(AUDIO_FRAMES_PER_BUFFER*2)
# insert samples in buffer
x = np.frombuffer(data_in48k, dtype=np.int16)
x.tofile(frx)
if len(x) != AUDIO_FRAMES_PER_BUFFER:
receive = False
x = resampler.resample48_to_8(x)
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audio_buffer.push(x)
# when we have enough samples call FreeDV Rx
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while audio_buffer.nbuffer >= nin:
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# demodulate audio
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nbytes = codec2.api.freedv_rawdatarx(freedv, bytes_out, audio_buffer.buffer.ctypes)
audio_buffer.pop(nin)
# call me on every loop!
nin = codec2.api.freedv_nin(freedv)
rx_status = codec2.api.freedv_get_rx_status(freedv)
if rx_status & codec2.api.FREEDV_RX_BIT_ERRORS:
rx_errors = rx_errors + 1
if DEBUGGING_MODE:
rx_status = codec2.api.rx_sync_flags_to_text[rx_status]
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print("nin: %5d rx_status: %4s naudio_buffer: %4d" % \
(nin,rx_status,audio_buffer.nbuffer), file=sys.stderr)
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if nbytes:
total_n_bytes = total_n_bytes + nbytes
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if nbytes == bytes_per_frame:
rx_total_frames = rx_total_frames + 1
rx_frames = rx_frames + 1
if rx_frames == N_FRAMES_PER_BURST:
rx_frames = 0
rx_bursts = rx_bursts + 1
if rx_bursts == N_BURSTS:
receive = False
if time.time() >= timeout:
print("TIMEOUT REACHED")
if nread_exceptions:
print("nread_exceptions %d - receive audio lost! Consider increasing Pyaudio frames_per_buffer..." % \
nread_exceptions, file=sys.stderr)
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print(f"RECEIVED BURSTS: {rx_bursts} RECEIVED FRAMES: {rx_total_frames} RX_ERRORS: {rx_errors}", file=sys.stderr)
frx.close()
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# and at last check if we had an openend pyaudio instance and close it
if AUDIO_INPUT_DEVICE != -1:
stream_rx.close()
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p.terminate()