#!/usr/bin/env python3
# -*- coding: utf-8 -*-


import ctypes
from ctypes import *
import pathlib
import pyaudio
import time
import threading
import audioop
import argparse
import sys
sys.path.insert(0,'..')
from tnc import codec2
import numpy as np        

# GET PARAMETER INPUTS  
parser = argparse.ArgumentParser(description='FreeDATA 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")  

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
DELAY_BETWEEN_BURSTS = args.DELAY_BETWEEN_BURSTS/1000
AUDIO_OUTPUT_DEVICE = args.AUDIO_OUTPUT_DEVICE


# AUDIO PARAMETERS
AUDIO_FRAMES_PER_BUFFER = 2400 
MODEM_SAMPLE_RATE = codec2.api.FREEDV_FS_8000
AUDIO_SAMPLE_RATE_TX = 48000
assert (AUDIO_SAMPLE_RATE_TX % MODEM_SAMPLE_RATE) == 0

if AUDIO_OUTPUT_DEVICE != -1: 
    p = pyaudio.PyAudio()
    # auto search for loopback devices
    if AUDIO_OUTPUT_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_OUTPUT_DEVICE = loopback_list[1] #0  = RX   1 = TX
            print(f"loopback_list tx: {loopback_list}", file=sys.stderr)
        else:
            quit()        
    # pyaudio init
    stream_tx = p.open(format=pyaudio.paInt16,
                            channels=1,
                            rate=AUDIO_SAMPLE_RATE_TX,
                            frames_per_buffer=AUDIO_FRAMES_PER_BUFFER, #n_nom_modem_samples
                            output=True,
                            output_device_index=AUDIO_OUTPUT_DEVICE,  
                            )      

resampler = codec2.resampler()
modes = [codec2.api.FREEDV_MODE_DATAC0, codec2.api.FREEDV_MODE_DATAC1, codec2.api.FREEDV_MODE_DATAC3]
for m in modes:
    
    freedv = cast(codec2.api.freedv_open(m), c_void_p)        
            
    n_tx_modem_samples = codec2.api.freedv_get_n_tx_modem_samples(freedv)
    mod_out = 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 = 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 = 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

                                    
    # data binary string
    data_out = b'HELLO WORLD!'
        
    buffer = bytearray(payload_per_frame)
    # set buffersize to length of data which will be send
    buffer[:len(data_out)] = 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,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,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"TX BURST: {i}/{N_BURSTS} FRAME: {n}/{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(MODEM_SAMPLE_RATE*DELAY_BETWEEN_BURSTS)
        mod_out_silence = 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 = resampler.resample8_to_48(x)

        # check if we want to use an audio device or stdout
        if AUDIO_OUTPUT_DEVICE != -1: 
            stream_tx.write(txbuffer_48k.tobytes())
        else:
            # this test needs a lot of time, so we are having a look at times...
            starttime = time.time()            
 
            # print data to terminal for piping the output to other programs
            sys.stdout.buffer.write(txbuffer_48k)    
            sys.stdout.flush()

            # and at least print the needed time to see which time we needed
            timeneeded = time.time()-starttime
            #print(f"time: {timeneeded} buffer: {len(txbuffer)}", file=sys.stderr)  
       
            
# and at last check if we had an openend pyaudio instance and close it
if AUDIO_OUTPUT_DEVICE != -1: 
    time.sleep(stream_tx.get_output_latency())
    stream_tx.stop_stream()
    stream_tx.close()
    p.terminate()