mirror of
https://github.com/DJ2LS/FreeDATA
synced 2024-05-14 10:04:33 +02:00
257 lines
8.8 KiB
Python
257 lines
8.8 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 argparse
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import ctypes
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import queue
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import sys
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import time
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import numpy as np
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import pyaudio
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sys.path.insert(0, "..")
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from modem 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("--delay", dest="DELAY_BETWEEN_BURSTS", default=500, type=int)
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parser.add_argument(
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"--audiodev",
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dest="AUDIO_OUTPUT_DEVICE",
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default=-1,
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type=int,
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help="audio output device number to use",
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)
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parser.add_argument(
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"--list",
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dest="LIST",
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action="store_true",
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help="list audio devices by number and exit",
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)
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parser.add_argument(
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"--testframes",
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dest="TESTFRAMES",
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action="store_true",
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default=False,
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help="generate testframes",
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)
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args, _ = parser.parse_known_args()
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if args.LIST:
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p = pyaudio.PyAudio()
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for dev in range(p.get_device_count()):
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print("audiodev: ", dev, p.get_device_info_by_index(dev)["name"])
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sys.exit()
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class Test:
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def __init__(self):
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self.dataqueue = queue.Queue()
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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_OUTPUT_DEVICE = args.AUDIO_OUTPUT_DEVICE
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self.DELAY_BETWEEN_BURSTS = args.DELAY_BETWEEN_BURSTS / 1000
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# AUDIO PARAMETERS
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# v-- consider increasing if you get nread_exceptions > 0
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self.AUDIO_FRAMES_PER_BUFFER = 2400
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self.MODEM_SAMPLE_RATE = codec2.api.FREEDV_FS_8000
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self.AUDIO_SAMPLE_RATE_TX = 48000
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# make sure our resampler will work
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assert (
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self.AUDIO_SAMPLE_RATE_TX / self.MODEM_SAMPLE_RATE
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) == codec2.api.FDMDV_OS_48
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self.transmit = True
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self.resampler = codec2.resampler()
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# check if we want to use an audio device then do a pyaudio init
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if self.AUDIO_OUTPUT_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_OUTPUT_DEVICE == -2:
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loopback_list = [
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dev
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for dev in range(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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]
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if len(loopback_list) >= 2:
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self.AUDIO_OUTPUT_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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sys.exit()
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print(
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f"AUDIO OUTPUT DEVICE: {self.AUDIO_OUTPUT_DEVICE} "
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f"DEVICE: {self.p.get_device_info_by_index(self.AUDIO_OUTPUT_DEVICE)['name']} "
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f"AUDIO SAMPLE RATE: {self.AUDIO_SAMPLE_RATE_TX}",
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file=sys.stderr,
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)
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self.stream_tx = self.p.open(
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format=pyaudio.paInt16,
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channels=1,
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rate=self.AUDIO_SAMPLE_RATE_TX,
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frames_per_buffer=self.AUDIO_FRAMES_PER_BUFFER,
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input=False,
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output=True,
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output_device_index=self.AUDIO_OUTPUT_DEVICE,
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stream_callback=self.callback,
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)
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else:
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print("test_callback_multimode_tx: Not written for STDOUT usage.")
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print("Exiting.")
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sys.exit()
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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.ftx = open("tx48_callback.raw", mode="wb")
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# data binary string
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if args.TESTFRAMES:
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self.data_out = bytearray(14)
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self.data_out[:1] = bytes([255])
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self.data_out[1:2] = bytes([1])
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self.data_out[2:] = b"HELLO WORLD"
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else:
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self.data_out = b"HELLO WORLD!"
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def callback(self, data_in48k, frame_count, time_info, status):
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data_out48k = self.dataqueue.get()
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return (data_out48k, pyaudio.paContinue)
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def run_audio(self):
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try:
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print("starting pyaudio callback", file=sys.stderr)
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self.stream_tx.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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sheeps = 0
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while self.transmit:
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time.sleep(1)
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sheeps = sheeps + 1
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print(f"counting sheeps...{sheeps}")
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self.ftx.close()
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# close pyaudio instance
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self.stream_tx.close()
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self.p.terminate()
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def create_modulation(self):
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modes = [
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codec2.FREEDV_MODE.datac13.value,
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codec2.FREEDV_MODE.datac1.value,
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codec2.FREEDV_MODE.datac3.value,
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]
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for m in modes:
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freedv = ctypes.cast(codec2.api.freedv_open(m), ctypes.c_void_p)
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n_tx_modem_samples = codec2.api.freedv_get_n_tx_modem_samples(freedv)
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mod_out = ctypes.create_string_buffer(2 * n_tx_modem_samples)
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n_tx_preamble_modem_samples = (
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codec2.api.freedv_get_n_tx_preamble_modem_samples(freedv)
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)
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mod_out_preamble = ctypes.create_string_buffer(
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2 * n_tx_preamble_modem_samples
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)
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n_tx_postamble_modem_samples = (
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codec2.api.freedv_get_n_tx_postamble_modem_samples(freedv)
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)
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mod_out_postamble = ctypes.create_string_buffer(
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2 * n_tx_postamble_modem_samples
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)
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bytes_per_frame = int(
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codec2.api.freedv_get_bits_per_modem_frame(freedv) / 8
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)
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payload_per_frame = bytes_per_frame - 2
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buffer = bytearray(payload_per_frame)
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# set buffer size to length of data which will be sent
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buffer[: len(self.data_out)] = self.data_out
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crc = ctypes.c_ushort(
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codec2.api.freedv_gen_crc16(bytes(buffer), payload_per_frame)
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) # generate CRC16
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# convert crc to 2 byte hex string
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crc = crc.value.to_bytes(2, byteorder="big")
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buffer += crc # append crc16 to buffer
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data = (ctypes.c_ubyte * bytes_per_frame).from_buffer_copy(buffer)
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for i in range(1, self.N_BURSTS + 1):
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# write preamble to txbuffer
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codec2.api.freedv_rawdatapreambletx(freedv, mod_out_preamble)
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txbuffer = bytes(mod_out_preamble)
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# create modulaton for N = FRAMESPERBURST and append it to txbuffer
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for n in range(1, self.N_FRAMES_PER_BURST + 1):
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data = (ctypes.c_ubyte * bytes_per_frame).from_buffer_copy(buffer)
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codec2.api.freedv_rawdatatx(
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freedv, mod_out, data
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) # modulate DATA and save it into mod_out pointer
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txbuffer += bytes(mod_out)
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print(
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f"GENERATING TX BURST: {i}/{self.N_BURSTS} FRAME: {n}/{self.N_FRAMES_PER_BURST}",
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file=sys.stderr,
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)
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# append postamble to txbuffer
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codec2.api.freedv_rawdatapostambletx(freedv, mod_out_postamble)
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txbuffer += bytes(mod_out_postamble)
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# append a delay between bursts as audio silence
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samples_delay = int(self.MODEM_SAMPLE_RATE * self.DELAY_BETWEEN_BURSTS)
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mod_out_silence = ctypes.create_string_buffer(samples_delay * 2)
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txbuffer += bytes(mod_out_silence)
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# resample up to 48k (resampler works on np.int16)
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x = np.frombuffer(txbuffer, dtype=np.int16)
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txbuffer_48k = self.resampler.resample8_to_48(x)
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# split modulated audio to chunks
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# https://newbedev.com/how-to-split-a-byte-string-into-separate-bytes-in-python
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txbuffer_48k = bytes(txbuffer_48k)
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chunk = [
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txbuffer_48k[i : i + self.AUDIO_FRAMES_PER_BUFFER * 2]
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for i in range(
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0, len(txbuffer_48k), self.AUDIO_FRAMES_PER_BUFFER * 2
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)
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]
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# add modulated chunks to fifo buffer
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for c in chunk:
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# if data is shorter than the expcected audio frames per buffer we need to append 0
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# to prevent the callback from stucking/crashing
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if len(c) < self.AUDIO_FRAMES_PER_BUFFER * 2:
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c += bytes(self.AUDIO_FRAMES_PER_BUFFER * 2 - len(c))
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self.dataqueue.put(c)
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test = Test()
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test.create_modulation()
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test.run_audio()
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