FreeDATA/test/test_callback_rx.py

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

@author: DJ2LS
"""

import argparse
import ctypes
import sys
import time

import numpy as np
import pyaudio

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_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.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(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:
                    sys.exit()

            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 = ctypes.cast(codec2.api.freedv_open(self.MODE), ctypes.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 = ctypes.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()