FreeDATA/modem/data cemetery/test/util_callback_multimode_tx.py

#!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""
Created on Wed Dec 23 07:04:24 2020

@author: DJ2LS
"""

import argparse
import ctypes
import queue
import sys
import time

import numpy as np
import pyaudio

sys.path.insert(0, "..")
from modem 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(
    "--audiodev",
    dest="AUDIO_OUTPUT_DEVICE",
    default=-1,
    type=int,
    help="audio output device number to use",
)
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_known_args()

if args.LIST:
    p = pyaudio.PyAudio()
    for dev in range(p.get_device_count()):
        print("audiodev: ", dev, p.get_device_info_by_index(dev)["name"])
    sys.exit()


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.DELAY_BETWEEN_BURSTS = args.DELAY_BETWEEN_BURSTS / 1000

        # AUDIO PARAMETERS
        # v-- consider increasing if you get nread_exceptions > 0
        self.AUDIO_FRAMES_PER_BUFFER = 2400
        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 a 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 = [
                    dev
                    for dev in range(self.p.get_device_count())
                    if "Loopback: PCM" in self.p.get_device_info_by_index(dev)["name"]
                ]

                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:
                    sys.exit()

            print(
                f"AUDIO OUTPUT DEVICE: {self.AUDIO_OUTPUT_DEVICE} "
                f"DEVICE: {self.p.get_device_info_by_index(self.AUDIO_OUTPUT_DEVICE)['name']}  "
                f"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,
            )
        else:
            print("test_callback_multimode_tx: Not written for STDOUT usage.")
            print("Exiting.")
            sys.exit()

        # 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("starting pyaudio callback", file=sys.stderr)
            self.stream_tx.start_stream()
        except Exception as e:
            print(f"pyAudio error: {e}", file=sys.stderr)

        sheeps = 0
        while self.transmit:
            time.sleep(1)
            sheeps = sheeps + 1
            print(f"counting sheeps...{sheeps}")

        self.ftx.close()

        # close pyaudio instance
        self.stream_tx.close()
        self.p.terminate()

    def create_modulation(self):

        modes = [
            codec2.FREEDV_MODE.datac13.value,
            codec2.FREEDV_MODE.datac1.value,
            codec2.FREEDV_MODE.datac3.value,
        ]
        for m in modes:

            freedv = ctypes.cast(codec2.api.freedv_open(m), ctypes.c_void_p)

            n_tx_modem_samples = codec2.api.freedv_get_n_tx_modem_samples(freedv)
            mod_out = ctypes.create_string_buffer(2 * n_tx_modem_samples)

            n_tx_preamble_modem_samples = (
                codec2.api.freedv_get_n_tx_preamble_modem_samples(freedv)
            )
            mod_out_preamble = ctypes.create_string_buffer(
                2 * n_tx_preamble_modem_samples
            )

            n_tx_postamble_modem_samples = (
                codec2.api.freedv_get_n_tx_postamble_modem_samples(freedv)
            )
            mod_out_postamble = ctypes.create_string_buffer(
                2 * n_tx_postamble_modem_samples
            )

            bytes_per_frame = int(
                codec2.api.freedv_get_bits_per_modem_frame(freedv) / 8
            )
            payload_per_frame = bytes_per_frame - 2

            buffer = bytearray(payload_per_frame)
            # set buffer size to length of data which will be sent
            buffer[: len(self.data_out)] = self.data_out

            crc = ctypes.c_ushort(
                codec2.api.freedv_gen_crc16(bytes(buffer), payload_per_frame)
            )  # generate CRC16
            # convert crc to 2 byte hex string
            crc = crc.value.to_bytes(2, byteorder="big")
            buffer += crc  # append crc16 to buffer
            data = (ctypes.c_ubyte * bytes_per_frame).from_buffer_copy(buffer)

            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"GENERATING 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 = ctypes.create_string_buffer(samples_delay * 2)
                txbuffer += bytes(mod_out_silence)

                # 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 modulated 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()