/* sparc/audio.c vi:ts=3 sw=3: */ /* $Id: audio.c,v 4.3 1995/02/01 16:43:47 espie Exp $ * $Log: audio.c,v $ * Revision 4.3 1995/02/01 16:43:47 espie * 23 bit samples. * * Revision 4.3 1995/02/01 16:43:47 espie * 23 bit samples. * * Revision 4.2 1994/01/13 09:19:08 espie * Forgotten something. * * Revision 4.0 1994/01/11 18:16:36 espie * New release. * * Revision 1.2 1993/12/26 18:54:21 Espie * Handle errors better. * * Revision 1.1 1993/12/26 00:55:53 Espie * Initial revision * * Revision 3.14 1993/12/04 16:12:50 espie * BOOL -> boolean. * * Revision 3.13 1993/12/02 15:45:33 espie * Merged ss10/solaris. * * Revision 3.12 1993/11/27 17:07:33 espie * Merged support for solaris together. * * Revision 3.11 1993/11/17 15:31:16 espie * *** empty log message *** * * * Revision 3.9 1993/07/14 16:33:41 espie * Fixed /16 bug. * * Revision 3.8 1993/05/09 14:06:03 espie * Corrected mix problem. * * Revision 3.6 1992/12/03 15:00:50 espie * restore stty. * * Revision 3.4 1992/11/24 10:51:19 espie * Sync pseudo call. * * Revision 3.3 1992/11/22 17:20:01 espie * Added update_frequency call, mostly unchecked * * Revision 3.2 1992/11/20 14:53:32 espie * Added finetune. * * Revision 3.1 1992/11/19 20:44:47 espie * Protracker commands. * * Revision 3.0 1992/11/18 16:08:05 espie * New release. * * Revision 1.3 1992/11/17 15:38:00 espie * discard_buffer() call for snappier interface calls. * - Unified support for all sparcs. * - moved down to level 2 io. */ #include "defs.h" #include "extern.h" #ifdef SOLARIS #include #else #include #endif #include #include #include /* things that aren't defined in all sun/audioio.h */ #ifndef AUDIO_ENCODING_LINEAR #define AUDIO_ENCODING_LINEAR (3) #endif #ifndef AUDIO_GETDEV #define AUDIO_GETDEV _IOR(A, 4, int) #endif #ifndef AUDIO_DEV_UNKNOWN #define AUDIO_DEV_UNKNOWN (0) #endif #ifndef AUDIO_DEV_AMD #define AUDIO_DEV_AMD (1) #endif ID("$Id: audio.c,v 4.3 1995/02/01 16:43:47 espie Exp $") LOCAL int audio; LOCAL struct audio_info ainfo; LOCAL char *buffer; LOCAL short *sbuffer; LOCAL int index; LOCAL int dsize; LOCAL int stereo; LOCAL int primary, secondary; void set_mix(percent) int percent; { percent *= 256; percent /= 100; primary = percent; secondary = 512 - percent; } #define abs(x) ((x) < 0 ? -(x) : (x)) LOCAL int available(f) int f; { static int possible[] = { 8000, 9600, 11025, 16000, 18900, 22050, 32000, 37800, 44100, 48000, 0}; int best = 8000; int i; for (i = 0; possible[i]; i++) if (abs(possible[i] - f) < abs(best - f)) best = possible[i]; return best; } int open_audio(f, s) int f; int s; { int type; int basic; #ifdef SOLARIS audio_device_t dev; audio = open("/dev/audio", O_WRONLY); #else audio = open("/dev/audio", O_WRONLY|O_NDELAY); #endif basic = 0; if (audio == -1) end_all("Error: could not open audio"); if (f == 0) f = 22050; /* round frequency to acceptable value */ f = available(f); /* check whether we know about AUDIO_ENCODING_LINEAR */ #ifdef SOLARIS ioctl(audio, AUDIO_GETDEV, &dev); if (strcmp(dev.name, "SUNW,dbri") != 0) #else if (ioctl(audio, AUDIO_GETDEV, &type) || type == AUDIO_DEV_UNKNOWN || type == AUDIO_DEV_AMD || basic) #endif { /* not a new ss5/10/20 -> revert to base quality audio */ stereo = 0; dsize = 1; ainfo.play.sample_rate = 8000; ainfo.play.encoding = AUDIO_ENCODING_ULAW; ainfo.play.channels = 1; puts("Basic"); } else { /* tentative set up */ stereo = s; AUDIO_INITINFO(&ainfo); ainfo.play.sample_rate = f; ainfo.play.precision = 16; dsize = 2; if (stereo) { ainfo.play.channels = 2; } else ainfo.play.channels = 1; /* try it */ ainfo.play.encoding = AUDIO_ENCODING_LINEAR; if (ioctl(audio, AUDIO_SETINFO, &ainfo) != 0) /* didn't work: fatal problem */ end_all("Error: AUDIO_SETINFO"); } index = 0; buffer = (char *)malloc(dsize * ainfo.play.channels * ainfo.play.sample_rate); sbuffer = (short *) buffer; if (!buffer) end_all("Error: could not allocate buffer"); return ainfo.play.sample_rate; } void set_synchro(s) int s; { } int update_frequency() { int oldfreq; oldfreq = ainfo.play.sample_rate; if (ioctl(audio, AUDIO_GETINFO, &ainfo) == 0) { if (oldfreq != ainfo.play.sample_rate) { buffer = realloc(buffer, dsize * ainfo.play.channels * ainfo.play.sample_rate); sbuffer = (short *)buffer; return ainfo.play.sample_rate; } } return 0; } LOCAL int sign(x) unsigned char x; { return x; } /************************************************************************/ /* For routine 'cvt' only */ /************************************************************************/ /* Copyright 1989 by Rich Gopstein and Harris Corporation */ /************************************************************************/ LOCAL unsigned int cvt(ch) int ch; { int mask; if (ch < 0) { ch = -ch; mask = 0x7f; } else mask = 0xff; if (ch < 32) { ch = 0xF0 | 15 - (ch / 2); } else if (ch < 96) { ch = 0xE0 | 15 - (ch - 32) / 4; } else if (ch < 224) { ch = 0xD0 | 15 - (ch - 96) / 8; } else if (ch < 480) { ch = 0xC0 | 15 - (ch - 224) / 16; } else if (ch < 992) { ch = 0xB0 | 15 - (ch - 480) / 32; } else if (ch < 2016) { ch = 0xA0 | 15 - (ch - 992) / 64; } else if (ch < 4064) { ch = 0x90 | 15 - (ch - 2016) / 128; } else if (ch < 8160) { ch = 0x80 | 15 - (ch - 4064) / 256; } else { ch = 0x80; } return (mask & ch); } void output_samples(left, right) int left, right; { if (stereo) { sbuffer[index++] = (left * primary + right * secondary)/65536; sbuffer[index++] = (right * primary + left * secondary)/65536; } else switch(ainfo.play.encoding) { case AUDIO_ENCODING_LINEAR: sbuffer[index++] = (left + right)/256; break; case AUDIO_ENCODING_ULAW: buffer[index++] = cvt((left + right) /1024); break; } } void flush_buffer() { int actual; actual = write(audio, buffer, dsize * index); if (actual == -1) notice("Write to audio failed"); else if (actual != dsize * index) notice("Short write to audio"); index = 0; } void discard_buffer() { ioctl(audio, I_FLUSH, FLUSHW); } void close_audio() { free(buffer); close(audio); }