mirror of
https://github.com/DJ2LS/FreeDATA
synced 2024-05-14 08:04:33 +00:00
05e65018d8
not sure if we really need this
250 lines
11 KiB
Python
250 lines
11 KiB
Python
#!/usr/bin/env python3
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# -*- coding: utf-8 -*-
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"""
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Created on Wed Dec 23 07:04:24 2020
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@author: DJ2LS
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"""
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import ctypes
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from ctypes import *
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import pathlib
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import pyaudio
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import sys
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import logging
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import time
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import threading
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import sys
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import argparse
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import numpy as np
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sys.path.insert(0,'..')
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from tnc import codec2
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#--------------------------------------------GET PARAMETER INPUTS
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parser = argparse.ArgumentParser(description='FreeDATA audio test')
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parser.add_argument('--bursts', dest="N_BURSTS", default=1, type=int)
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parser.add_argument('--framesperburst', dest="N_FRAMES_PER_BURST", default=1, type=int)
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parser.add_argument('--audiodev', dest="AUDIO_INPUT_DEVICE", default=-1, type=int, help="audio device number to use")
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parser.add_argument('--debug', dest="DEBUGGING_MODE", action="store_true")
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parser.add_argument('--list', dest="LIST", action="store_true", help="list audio devices by number and exit")
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parser.add_argument('--timeout', dest="TIMEOUT", default=10, type=int, help="Timeout (seconds) before test ends")
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args = parser.parse_args()
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if args.LIST:
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p = pyaudio.PyAudio()
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for dev in range(0,p.get_device_count()):
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print("audiodev: ", dev, p.get_device_info_by_index(dev)["name"])
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quit()
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class Test():
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def __init__(self):
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self.N_BURSTS = args.N_BURSTS
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self.N_FRAMES_PER_BURST = args.N_FRAMES_PER_BURST
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self.AUDIO_INPUT_DEVICE = args.AUDIO_INPUT_DEVICE
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self.DEBUGGING_MODE = args.DEBUGGING_MODE
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self.TIMEOUT = args.TIMEOUT
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# AUDIO PARAMETERS
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self.AUDIO_FRAMES_PER_BUFFER = 2400*2 # <- consider increasing if you get nread_exceptions > 0
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self.MODEM_SAMPLE_RATE = codec2.api.FREEDV_FS_8000
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self.AUDIO_SAMPLE_RATE_RX = 48000
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# make sure our resampler will work
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assert (self.AUDIO_SAMPLE_RATE_RX / self.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
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if self.AUDIO_INPUT_DEVICE != -1:
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self.p = pyaudio.PyAudio()
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# auto search for loopback devices
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if self.AUDIO_INPUT_DEVICE == -2:
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loopback_list = []
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for dev in range(0,self.p.get_device_count()):
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if 'Loopback: PCM' in self.p.get_device_info_by_index(dev)["name"]:
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loopback_list.append(dev)
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if len(loopback_list) >= 2:
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self.AUDIO_INPUT_DEVICE = loopback_list[0] #0 = RX 1 = TX
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print(f"loopback_list rx: {loopback_list}", file=sys.stderr)
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else:
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quit()
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print(f"AUDIO INPUT DEVICE: {self.AUDIO_INPUT_DEVICE} DEVICE: {self.p.get_device_info_by_index(self.AUDIO_INPUT_DEVICE)['name']} \
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AUDIO SAMPLE RATE: {self.AUDIO_SAMPLE_RATE_RX}", file=sys.stderr)
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self.stream_rx = self.p.open(format=pyaudio.paInt16,
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channels=1,
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rate=self.AUDIO_SAMPLE_RATE_RX,
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frames_per_buffer=self.AUDIO_FRAMES_PER_BUFFER,
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input=True,
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output=False,
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input_device_index=self.AUDIO_INPUT_DEVICE,
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stream_callback=self.callback
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)
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# open codec2 instance
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self.datac0_freedv = cast(codec2.api.freedv_open(codec2.api.FREEDV_MODE_DATAC0), c_void_p)
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self.datac0_bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(self.datac0_freedv)/8)
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self.datac0_bytes_out = create_string_buffer(self.datac0_bytes_per_frame)
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codec2.api.freedv_set_frames_per_burst(self.datac0_freedv,self.N_FRAMES_PER_BURST)
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self.datac0_buffer = codec2.audio_buffer(2*self.AUDIO_FRAMES_PER_BUFFER)
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self.datac1_freedv = cast(codec2.api.freedv_open(codec2.api.FREEDV_MODE_DATAC1), c_void_p)
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self.datac1_bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(self.datac1_freedv)/8)
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self.datac1_bytes_out = create_string_buffer(self.datac1_bytes_per_frame)
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codec2.api.freedv_set_frames_per_burst(self.datac1_freedv,self.N_FRAMES_PER_BURST)
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self.datac1_buffer = codec2.audio_buffer(2*self.AUDIO_FRAMES_PER_BUFFER)
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self.datac3_freedv = cast(codec2.api.freedv_open(codec2.api.FREEDV_MODE_DATAC3), c_void_p)
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self.datac3_bytes_per_frame = int(codec2.api.freedv_get_bits_per_modem_frame(self.datac3_freedv)/8)
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self.datac3_bytes_out = create_string_buffer(self.datac3_bytes_per_frame)
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codec2.api.freedv_set_frames_per_burst(self.datac3_freedv,self.N_FRAMES_PER_BURST)
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self.datac3_buffer = codec2.audio_buffer(2*self.AUDIO_FRAMES_PER_BUFFER)
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# SET COUNTERS
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self.rx_total_frames_datac0 = 0
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self.rx_frames_datac0 = 0
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self.rx_bursts_datac0 = 0
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self.rx_total_frames_datac1 = 0
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self.rx_frames_datac1 = 0
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self.rx_bursts_datac1 = 0
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self.rx_total_frames_datac3 = 0
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self.rx_frames_datac3 = 0
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self.rx_bursts_datac3 = 0
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self.rx_errors = 0
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self.nread_exceptions = 0
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self.timeout = time.time() + self.TIMEOUT
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self.receive = True
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self.resampler = codec2.resampler()
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# Copy received 48 kHz to a file. Listen to this file with:
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# aplay -r 48000 -f S16_LE rx48_callback.raw
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# Corruption of this file is a good way to detect audio card issues
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self.frx = open("rx48_callback_multimode.raw", mode='wb')
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# initial nin values
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self.datac0_nin = codec2.api.freedv_nin(self.datac0_freedv)
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self.datac1_nin = codec2.api.freedv_nin(self.datac1_freedv)
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self.datac3_nin = codec2.api.freedv_nin(self.datac3_freedv)
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self.LOGGER_THREAD = threading.Thread(target=self.print_stats, name="LOGGER_THREAD")
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self.LOGGER_THREAD.start()
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def callback(self, data_in48k, frame_count, time_info, status):
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x = np.frombuffer(data_in48k, dtype=np.int16)
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x.tofile(self.frx)
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x = self.resampler.resample48_to_8(x)
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self.datac0_buffer.push(x)
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self.datac1_buffer.push(x)
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self.datac3_buffer.push(x)
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while self.datac0_buffer.nbuffer >= self.datac0_nin:
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# demodulate audio
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nbytes = codec2.api.freedv_rawdatarx(self.datac0_freedv, self.datac0_bytes_out, self.datac0_buffer.buffer.ctypes)
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self.datac0_buffer.pop(self.datac0_nin)
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self.datac0_nin = codec2.api.freedv_nin(self.datac0_freedv)
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if nbytes == self.datac0_bytes_per_frame:
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self.rx_total_frames_datac0 = self.rx_total_frames_datac0 + 1
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self.rx_frames_datac0 = self.rx_frames_datac0 + 1
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if self.rx_frames_datac0 == self.N_FRAMES_PER_BURST:
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self.rx_frames_datac0 = 0
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self.rx_bursts_datac0 = self.rx_bursts_datac0 + 1
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while self.datac1_buffer.nbuffer >= self.datac1_nin:
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# demodulate audio
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nbytes = codec2.api.freedv_rawdatarx(self.datac1_freedv, self.datac1_bytes_out, self.datac1_buffer.buffer.ctypes)
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self.datac1_buffer.pop(self.datac1_nin)
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self.datac1_nin = codec2.api.freedv_nin(self.datac1_freedv)
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if nbytes == self.datac1_bytes_per_frame:
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self.rx_total_frames_datac1 = self.rx_total_frames_datac1 + 1
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self.rx_frames_datac1 = self.rx_frames_datac1 + 1
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if self.rx_frames_datac1 == self.N_FRAMES_PER_BURST:
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self.rx_frames_datac1 = 0
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self.rx_bursts_datac1 = self.rx_bursts_datac1 + 1
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while self.datac3_buffer.nbuffer >= self.datac3_nin:
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# demodulate audio
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nbytes = codec2.api.freedv_rawdatarx(self.datac3_freedv, self.datac3_bytes_out, self.datac3_buffer.buffer.ctypes)
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self.datac3_buffer.pop(self.datac3_nin)
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self.datac3_nin = codec2.api.freedv_nin(self.datac3_freedv)
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if nbytes == self.datac3_bytes_per_frame:
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self.rx_total_frames_datac3 = self.rx_total_frames_datac3 + 1
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self.rx_frames_datac3 = self.rx_frames_datac3 + 1
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if self.rx_frames_datac3 == self.N_FRAMES_PER_BURST:
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self.rx_frames_datac3 = 0
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self.rx_bursts_datac3 = self.rx_bursts_datac3 + 1
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if (self.rx_bursts_datac0 and self.rx_bursts_datac1 and self.rx_bursts_datac3) == self.N_BURSTS:
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self.receive = False
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return (None, pyaudio.paContinue)
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def print_stats(self):
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while self.receive:
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time.sleep(0.01)
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if self.DEBUGGING_MODE:
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self.datac0_rxstatus = codec2.api.freedv_get_rx_status(self.datac0_freedv)
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self.datac0_rxstatus = codec2.api.rx_sync_flags_to_text[self.datac0_rxstatus]
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self.datac1_rxstatus = codec2.api.freedv_get_rx_status(self.datac1_freedv)
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self.datac1_rxstatus = codec2.api.rx_sync_flags_to_text[self.datac1_rxstatus]
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self.datac3_rxstatus = codec2.api.freedv_get_rx_status(self.datac3_freedv)
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self.datac3_rxstatus = codec2.api.rx_sync_flags_to_text[self.datac3_rxstatus]
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print("NIN0: %5d RX_STATUS0: %4s NIN1: %5d RX_STATUS1: %4s NIN3: %5d RX_STATUS3: %4s" % \
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(self.datac0_nin, self.datac0_rxstatus, self.datac1_nin, self.datac1_rxstatus, self.datac3_nin, self.datac3_rxstatus),
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file=sys.stderr)
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def run_audio(self):
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try:
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print(f"starting pyaudio callback", file=sys.stderr)
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self.stream_rx.start_stream()
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except Exception as e:
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print(f"pyAudio error: {e}", file=sys.stderr)
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while self.receive and time.time() < self.timeout:
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time.sleep(1)
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if time.time() >= self.timeout and self.stream_rx.is_active():
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print("TIMEOUT REACHED")
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self.receive = False
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if self.nread_exceptions:
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print("nread_exceptions %d - receive audio lost! Consider increasing Pyaudio frames_per_buffer..." % \
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self.nread_exceptions, file=sys.stderr)
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print(f"DATAC0: {self.rx_bursts_datac0}/{self.rx_total_frames_datac0} DATAC1: {self.rx_bursts_datac1}/{self.rx_total_frames_datac1} DATAC3: {self.rx_bursts_datac3}/{self.rx_total_frames_datac3}", file=sys.stderr)
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self.frx.close()
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# cloese pyaudio instance
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self.stream_rx.close()
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self.p.terminate()
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test = Test()
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test.run_audio()
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