#!/usr/bin/env python3 # -*- coding: utf-8 -*- import argparse import ctypes import sys import time import numpy as np import pyaudio sys.path.insert(0, "..") from tnc import codec2 def test_mm_tx(): # 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 args = parse_arguments() if args.LIST: p_audio = pyaudio.PyAudio() for dev in range(p_audio.get_device_count()): print("audiodev: ", dev, p_audio.get_device_info_by_index(dev)["name"]) sys.exit() 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 resampler = codec2.resampler() # Data binary string data_out = b"HELLO WORLD!" modes = [ codec2.api.FREEDV_MODE_DATAC0, codec2.api.FREEDV_MODE_DATAC1, codec2.api.FREEDV_MODE_DATAC3, ] if AUDIO_OUTPUT_DEVICE != -1: p_audio = pyaudio.PyAudio() # Auto search for loopback devices if AUDIO_OUTPUT_DEVICE == -2: loopback_list = [ dev for dev in range(p_audio.get_device_count()) if "Loopback: PCM" in p_audio.get_device_info_by_index(dev)["name"] ] 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: sys.exit() # pyaudio init stream_tx = p_audio.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, ) for mode in modes: freedv = ctypes.cast(codec2.api.freedv_open(mode), 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(data_out)] = data_out # Generate CRC16 crc = ctypes.c_ushort( codec2.api.freedv_gen_crc16(bytes(buffer), payload_per_frame) ) # 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 brst 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 frm in range(1, N_FRAMES_PER_BURST + 1): data = (ctypes.c_ubyte * bytes_per_frame).from_buffer_copy(buffer) # Modulate DATA and save it into mod_out pointer codec2.api.freedv_rawdatatx(freedv, mod_out, data) txbuffer += bytes(mod_out) print( f"TX BURST: {brst}/{N_BURSTS} FRAME: {frm}/{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 = ctypes.create_string_buffer(samples_delay * 2) txbuffer += bytes(mod_out_silence) # Resample up to 48k (resampler works on np.int16) audio_buffer = np.frombuffer(txbuffer, dtype=np.int16) txbuffer_48k = resampler.resample8_to_48(audio_buffer) # 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 opened 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_audio.terminate() def parse_arguments(): # 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_known_args() return args if __name__ == "__main__": test_mm_tx()