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/*
* FILE: audio_util.c
* PROGRAM: RAT
* AUTHOR: Orion Hodson / Isidor Kouvelas / Colin Perkins
*
* Copyright (c) 1995-2001 University College London
* All rights reserved.
*/
#ifndef HIDE_SOURCE_STRINGS
static const char cvsid[] =
"$Id: audio_util.c,v 1.24 2001/01/08 20:29:51 ucaccsp Exp $";
#endif /* HIDE_SOURCE_STRINGS */
#include "config_unix.h"
#include "config_win32.h"
#include "audio_types.h"
#include "audio_util.h"
#include "debug.h"
/* Bias control code */
typedef struct s_bias_ctl {
sample lta;
uint16_t step;
uint16_t freq;
uint32_t age;
} bias_ctl;
bias_ctl *
bias_ctl_create(int channels, int freq)
{
bias_ctl *bc = (bias_ctl*)xmalloc(channels*sizeof(bias_ctl));
memset(bc, 0, channels*sizeof(bias_ctl));
bc->step = channels;
bc->freq = freq;
return bc;
}
void
bias_ctl_destroy(bias_ctl *bc)
{
xfree(bc);
}
static void
remove_lta(bias_ctl *bc, sample *buf, int len, int step)
{
int i, samples;
int32_t m = 0;
/* NB m is buffer mean value. It's 32 bits long. During
* calculation we first let m equal sum of samples and then
* divide. Sum of few thousand samples (limit of len) will
* not exceed range of m. */
samples = len / step;
if (bc->age == 0) {
/* On first pass do expensive task of calculating and
* removing bias in two steps.
*/
for (i = 0; i < len; i += step) {
m += buf[i];
}
m /= samples;
bc->lta = (sample)m;
bc->age++;
for (i = 0; i < len; i += step) {
buf[i] -= bc->lta;
}
} else {
for (i = 0; i < len; i += step) {
m += buf[i];
buf[i] -= bc->lta;
}
m /= samples;
bc->lta -= ((bc->lta - (sample)m) / 8);
}
}
void
bias_remove(bias_ctl *bc, sample *buf, int len)
{
if (bc->step == 1) {
remove_lta(bc, buf, len, 1);
} else {
remove_lta(bc , buf , len / 2, 2);
remove_lta(bc+1, buf+1, len / 2, 2);
}
}
/* Audio processing utility functions */
void
audio_zero(sample *buf, int len, deve_e type)
{
assert(len>=0);
switch(type) {
case DEV_PCMU:
memset(buf, PCMU_AUDIO_ZERO, len);
break;
case DEV_PCMA:
memset(buf, PCMA_AUDIO_ZERO, len);
break;
case DEV_S8:
memset(buf, 0, len);
break;
case DEV_S16:
memset(buf, 0, 2*len);
break;
default:
fprintf(stderr, "%s:%d Type not recognized", __FILE__, __LINE__);
break;
}
}
void
audio_mix(sample *dst, sample *src, int len)
{
int i, tmp;
for(i = 0; i < len; i++) {
tmp = dst[i] + src[i];
if (tmp > 32767) {
tmp = 32767;
} else if (tmp < -32768) {
tmp = -32768;
}
dst[i] = tmp;
}
}
#ifdef WIN32
/* mmx_present is (C) Intel 1998 */
BOOL
mmx_present(void)
{
BOOL retval = TRUE;
DWORD RegEDX;
__try {
_asm {
mov eax, 1 // set up CPUID to return processor version and features
// 0 = vendor string, 1 = version info, 2 = cache info
CPUID // code bytes = 0fh, 0a2h
mov RegEDX, edx // features returned in edx
}
} __except(EXCEPTION_EXECUTE_HANDLER) { retval = FALSE; }
if (retval == FALSE)
return FALSE; // processor does not support CPUID
if (RegEDX & 0x800000) // bit 23 is set for MMX technology
{
__try { _asm emms } // try executing the MMX instruction "emms"
__except(EXCEPTION_EXECUTE_HANDLER) { retval = FALSE; }
} else {
return FALSE; // processor supports CPUID but does not support MMX technology
}
// if retval == 0 here, it means the processor has MMX technology but
// floating-point emulation is on; so MMX technology is unavailable
return retval;
}
/* audio_mix_mmx is a trivial bit of mmx code but it is six times
* quicker than best optimized C compiled that msvc offers, and msvc
* is marginally better than the x86 assembler generated by my hand [oh].
*/
void
audio_mix_mmx(sample *dst, sample *src, int len)
{
int tmp, endq, i;
/* The end of where we can do quad sample addition */
endq = ((len * sizeof(short)) / 8 - 1) * 8 ;
if (endq > 0) {
/* Order of these instructions is crucial for performance */
__asm {
mov esi, 0
mov eax, dst
mov ebx, src
jmp START
LOOP_L1:
add esi, 8
START:
movq mm0, [eax + esi]
paddsw mm0, [ebx + esi]
movq [eax + esi], mm0
cmp esi, endq
jb LOOP_L1
emms
}
endq += 8;
} else {
endq = 0;
}
for (i = endq / 2; i < len; i++) {
tmp = src[i] + dst[i];
if (tmp > 32767) {
tmp = 32767;
} else if (tmp < -32767) {
tmp = -32767;
}
dst[i] = (short)tmp;
}
}
#endif /* WIN32 */
#define ENERGY_CALC_STEP 1
uint16_t
audio_avg_energy(sample *buf, uint32_t samples, uint32_t channels)
{
register uint32_t e1, e2;
register sample *buf_end = buf + samples;
assert (channels > 0);
e1 = e2 = 0;
switch (channels) {
case 1:
while(buf < buf_end) {
e1 += abs(*buf);
buf += ENERGY_CALC_STEP;
}
break;
case 2:
/* SIMD would improve this */
while(buf < buf_end) {
e1 += abs(*buf++);
e2 += abs(*buf);
buf += ENERGY_CALC_STEP*channels - 1;
}
e1 = max(e1, e2);
samples /= channels; /* No. of samples to no. of sampling intervals */
}
/* Return mean sampled energy:
* no. of sampling points = samples/ENERGY_CALC_STEP;
*/
return (uint16_t)(e1*ENERGY_CALC_STEP/samples);
}
sample
audio_abs_max(sample *buf, uint32_t samples)
{
uint32_t i;
sample max;
max = 0;
for(i = 0; i < samples; i++) {
if (abs(buf[i]) > max) {
max = abs(buf[i]);
}
}
return max;
}
void
audio_scale_buffer(sample *buf, int len, double scale)
{
int i;
for(i = 0; i < len; i++) {
buf[i] = (sample)((double)buf[i] * scale);
}
}
void
audio_blend(sample *from, sample *to, sample *out, int samples, int channels)
{
int i;
int32_t tmp, sf, m;
m = samples / channels;
for (i = 0; i < samples; i++) {
sf = i / channels;
tmp = from[i] * (m - sf) + to[i] * sf;
tmp /= m;
out[i] = (sample)tmp;
}
}
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