/*************************************************************************** * Copyright (C) 2008 by Paul Lutus * * lutusp@arachnoid.com * * * * This program is free software; you can redistribute it and/or modify * * it under the terms of the GNU General Public License as published by * * the Free Software Foundation; either version 2 of the License, or * * (at your option) any later version. * * * * This program is distributed in the hope that it will be useful, * * but WITHOUT ANY WARRANTY; without even the implied warranty of * * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * * GNU General Public License for more details. * * * * You should have received a copy of the GNU General Public License * * along with this program; if not, write to the * * Free Software Foundation, Inc., * * 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. * ***************************************************************************/ #include #include #include #include #include #include #include #include #include #include #include "mulaw_converter.h" using namespace std; class SoundSource { string input_source; int format; unsigned long array_size,samples_per_second; static const int data_formats[]; public: SoundSource() { input_source = "/dev/dsp"; array_size = 2048; samples_per_second = 1000; format = 0; } bool decode_comline(int argc,char **argv) { for(int i = 1;i < argc;i++) { string arg = (string) argv[i]; if(arg.length() == 2 && arg[0] == '-') { char c = arg[1]; switch (c) { case 'h': cerr << "Usage: [-i(nput, default " << input_source << ")]" << endl; cerr << " [-f(ormat: 0=unsigned 16-bit, 1=signed 16-bit, 2=muLaw, default " << format << ")]" << endl; cerr << " [-a(rray_size, must be a power of 2, default " << array_size << ")]" << endl; cerr << " [-s(amples_per_second, default " << samples_per_second << ")]" << endl; return true; case 'a': sscanf(argv[++i],"%lud",&array_size); break; case 's': sscanf(argv[++i],"%lud",&samples_per_second); break; case 'f': sscanf(argv[++i],"%d",&format); break; case 'i': input_source = (string) argv[++i]; break; } } } return false; } template void scan_string(T& t,char* s) { stringstream ss(s); ss >> t; } long convert_pwr2(long v) { double q = floor(log(v) / log(2)); q = pow(2.0,q); return (long)q; } void set_options(int audio) { int fmt = data_formats[format]; int r = ioctl ( audio, SNDCTL_DSP_SETFMT, &fmt ); unsigned long v = samples_per_second; r = ioctl ( audio, SNDCTL_DSP_SPEED, &v ); // did the DSP accept our sample rate? if ( v != samples_per_second ) { // if not, accept the reply samples_per_second = v; cerr << "sound_source error: forced sample rate: " << samples_per_second << endl; } // this produces more reliable performance int val = ( int ) ( ( 2 << 16 ) + log2 ( 512 ) ); r = ioctl ( audio, SNDCTL_DSP_SETFRAGMENT, &val ); } void process_stream(int audio) { double volume_constant = 1.0/1024.0; double v; unsigned short s; unsigned char c; // continuous output while(true) { cout << array_size << endl; cout << samples_per_second << endl; for(unsigned long t = 0;t < array_size;t++) { if(format < 2) { if(read ( audio,&s,2) == 2 ) { if(format == 0) { v = ( ( double ) s-32768.0 ) * volume_constant; } else { v = ( double ) s * volume_constant; } cout << v << " " << 0 << endl; } } else { if(read ( audio,&c,1) == 1) { v = MulawConverter::mu2linfast ( c ) * volume_constant; cout << v << " " << 0 << endl; } } } } } void process(int argc,char **argv) { if(decode_comline(argc,argv)) { return; } // force entry to be a power of 2 array_size = convert_pwr2(array_size); int audio = open (input_source.c_str(),O_RDONLY); set_options(audio); process_stream(audio); } }; const int SoundSource::data_formats[] = { AFMT_U16_LE, AFMT_S16_LE,AFMT_MU_LAW,0 }; int main(int argc, char **argv) { SoundSource ss; ss.process(argc,argv); return 0; }