- 根目录:
- drivers
- isdn
- mISDN
- dsp_audio.c
/*
* Audio support data for mISDN_dsp.
*
* Copyright 2002/2003 by Andreas Eversberg (jolly@eversberg.eu)
* Rewritten by Peter
*
* This software may be used and distributed according to the terms
* of the GNU General Public License, incorporated herein by reference.
*
*/
#include <linux/delay.h>
#include <linux/mISDNif.h>
#include <linux/mISDNdsp.h>
#include "core.h"
#include "dsp.h"
/* ulaw[unsigned char] -> signed 16-bit */
s32 dsp_audio_ulaw_to_s32[256];
/* alaw[unsigned char] -> signed 16-bit */
s32 dsp_audio_alaw_to_s32[256];
s32 *dsp_audio_law_to_s32;
EXPORT_SYMBOL(dsp_audio_law_to_s32);
/* signed 16-bit -> law */
u8 dsp_audio_s16_to_law[65536];
EXPORT_SYMBOL(dsp_audio_s16_to_law);
/* alaw -> ulaw */
u8 dsp_audio_alaw_to_ulaw[256];
/* ulaw -> alaw */
static u8 dsp_audio_ulaw_to_alaw[256];
u8 dsp_silence;
/*****************************************************
* generate table for conversion of s16 to alaw/ulaw *
*****************************************************/
#define AMI_MASK 0x55
static inline unsigned char linear2alaw(short int linear)
{
int mask;
int seg;
int pcm_val;
static int seg_end[8] = {
0xFF, 0x1FF, 0x3FF, 0x7FF, 0xFFF, 0x1FFF, 0x3FFF, 0x7FFF
};
pcm_val = linear;
if (pcm_val >= 0) {
/* Sign (7th) bit = 1 */
mask = AMI_MASK | 0x80;
} else {
/* Sign bit = 0 */
mask = AMI_MASK;
pcm_val = -pcm_val;
}
/* Convert the scaled magnitude to segment number. */
for (seg = 0; seg < 8; seg++) {
if (pcm_val <= seg_end[seg])
break;
}
/* Combine the sign, segment, and quantization bits. */
return ((seg << 4) |
((pcm_val >> ((seg) ? (seg + 3) : 4)) & 0x0F)) ^ mask;
}
static inline short int alaw2linear(unsigned char alaw)
{
int i;
int seg;
alaw ^= AMI_MASK;
i = ((alaw & 0x0F) << 4) + 8 /* rounding error */;
seg = (((int) alaw & 0x70) >> 4);
if (seg)
i = (i + 0x100) << (seg - 1);
return (short int) ((alaw & 0x80) ? i : -i);
}
static inline short int ulaw2linear(unsigned char ulaw)
{
short mu, e, f, y;
static short etab[] = {0, 132, 396, 924, 1980, 4092, 8316, 16764};
mu = 255 - ulaw;
e = (mu & 0x70) / 16;
f = mu & 0x0f;
y = f * (1 << (e + 3));
y += etab[e];
if (mu & 0x80)
y = -y;
return y;
}
#define BIAS 0x84 /*!< define the add-in bias for 16 bit samples */
static unsigned char linear2ulaw(short sample)
{
static int exp_lut[256] = {
0, 0, 1, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3,
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7};
int sign, exponent, mantissa;
unsigned char ulawbyte;
/* Get the sample into sign-magnitude. */
sign = (sample >> 8) & 0x80; /* set aside the sign */
if (sign != 0)
sample = -sample; /* get magnitude */
/* Convert from 16 bit linear to ulaw. */
sample = sample + BIAS;
exponent = exp_lut[(sample >> 7) & 0xFF];
mantissa = (sample >> (exponent + 3)) & 0x0F;
ulawbyte = ~(sign | (exponent << 4) | mantissa);
return ulawbyte;
}
static int reverse_bits(int i)
{
int z, j;
z = 0;
for (j = 0; j < 8; j++) {
if ((i & (1 << j)) != 0)
z |= 1 << (7 - j);
}
return z;
}
void dsp_audio_generate_law_tables(void)
{
int i;
for (i = 0; i < 256; i++)
dsp_audio_alaw_to_s32[i] = alaw2linear(reverse_bits(i));
for (i = 0; i < 256; i++)
dsp_audio_ulaw_to_s32[i] = ulaw2linear(reverse_bits(i));
for (i = 0; i < 256; i++) {
dsp_audio_alaw_to_ulaw[i] =
linear2ulaw(dsp_audio_alaw_to_s32[i]);
dsp_audio_ulaw_to_alaw[i] =
linear2alaw(dsp_audio_ulaw_to_s32[i]);
}
}
void
dsp_audio_generate_s2law_table(void)
{
int i;
if (dsp_options & DSP_OPT_ULAW) {
/* generating ulaw-table */
for (i = -32768; i < 32768; i++) {
dsp_audio_s16_to_law[i & 0xffff] =
reverse_bits(linear2ulaw(i));
}
} else {
/* generating alaw-table */
for (i = -32768; i < 32768; i++) {
dsp_audio_s16_to_law[i & 0xffff] =
reverse_bits(linear2alaw(i));
}
}
}
/*
* the seven bit sample is the number of every second alaw-sample ordered by
* aplitude. 0x00 is negative, 0x7f is positive amplitude.
*/
u8 dsp_audio_seven2law[128];
u8 dsp_audio_law2seven[256];
/********************************************************************
* generate table for conversion law from/to 7-bit alaw-like sample *
********************************************************************/
void
dsp_audio_generate_seven(void)
{
int i, j, k;
u8 spl;
u8 sorted_alaw[256];
/* generate alaw table, sorted by the linear value */
for (i = 0; i < 256; i++) {
j = 0;
for (k = 0; k < 256; k++) {
if (dsp_audio_alaw_to_s32[k]
< dsp_audio_alaw_to_s32[i])
j++;
}
sorted_alaw[j] = i;
}
/* generate tabels */
for (i = 0; i < 256; i++) {
/* spl is the source: the law-sample (converted to alaw) */
spl = i;
if (dsp_options & DSP_OPT_ULAW)
spl = dsp_audio_ulaw_to_alaw[i];
/* find the 7-bit-sample */
for (j = 0; j < 256; j++) {
if (sorted_alaw[j] == spl)
break;
}
/* write 7-bit audio value */
dsp_audio_law2seven[i] = j >> 1;
}
for (i = 0; i < 128; i++) {
spl = sorted_alaw[i << 1];
if (dsp_options & DSP_OPT_ULAW)
spl = dsp_audio_alaw_to_ulaw[spl];
dsp_audio_seven2law[i] = spl;
}
}
/* mix 2*law -> law */
u8 dsp_audio_mix_law[65536];
/******************************************************
* generate mix table to mix two law samples into one *
******************************************************/
void
dsp_audio_generate_mix_table(void)
{
int i, j;
s32 sample;
i = 0;
while (i < 256) {
j = 0;
while (j < 256) {
sample = dsp_audio_law_to_s32[i];
sample += dsp_audio_law_to_s32[j];
if (sample > 32767)
sample = 32767;
if (sample < -32768)
sample = -32768;
dsp_audio_mix_law[(i<<8)|j] =
dsp_audio_s16_to_law[sample & 0xffff];
j++;
}
i++;
}
}
/*************************************
* generate different volume changes *
*************************************/
static u8 dsp_audio_reduce8[256];
static u8 dsp_audio_reduce7[256];
static u8 dsp_audio_reduce6[256];
static u8 dsp_audio_reduce5[256];
static u8 dsp_audio_reduce4[256];
static u8 dsp_audio_reduce3[256];
static u8 dsp_audio_reduce2[256];
static u8 dsp_audio_reduce1[256];
static u8 dsp_audio_increase1[256];
static u8 dsp_audio_increase2[256];
static u8 dsp_audio_increase3[256];
static u8 dsp_audio_increase4[256];
static u8 dsp_audio_increase5[256];
static u8 dsp_audio_increase6[256];
static u8 dsp_audio_increase7[256];
static u8 dsp_audio_increase8[256];
static u8 *dsp_audio_volume_change[16] = {
dsp_audio_reduce8,
dsp_audio_reduce7,
dsp_audio_reduce6,
dsp_audio_reduce5,
dsp_audio_reduce4,
dsp_audio_reduce3,
dsp_audio_reduce2,
dsp_audio_reduce1,
dsp_audio_increase1,
dsp_audio_increase2,
dsp_audio_increase3,
dsp_audio_increase4,
dsp_audio_increase5,
dsp_audio_increase6,
dsp_audio_increase7,
dsp_audio_increase8,
};
void
dsp_audio_generate_volume_changes(void)
{
register s32 sample;
int i;
int num[] = { 110, 125, 150, 175, 200, 300, 400, 500 };
int denum[] = { 100, 100, 100, 100, 100, 100, 100, 100 };
i = 0;
while (i < 256) {
dsp_audio_reduce8[i] = dsp_audio_s16_to_law[
(dsp_audio_law_to_s32[i] * denum[7] / num[7]) & 0xffff];
dsp_audio_reduce7[i] = dsp_audio_s16_to_law[
(dsp_audio_law_to_s32[i] * denum[6] / num[6]) & 0xffff];
dsp_audio_reduce6[i] = dsp_audio_s16_to_law[
(dsp_audio_law_to_s32[i] * denum[5] / num[5]) & 0xffff];
dsp_audio_reduce5[i] = dsp_audio_s16_to_law[
(dsp_audio_law_to_s32[i] * denum[4] / num[4]) & 0xffff];
dsp_audio_reduce4[i] = dsp_audio_s16_to_law[
(dsp_audio_law_to_s32[i] * denum[3] / num[3]) & 0xffff];
dsp_audio_reduce3[i] = dsp_audio_s16_to_law[
(dsp_audio_law_to_s32[i] * denum[2] / num[2]) & 0xffff];
dsp_audio_reduce2[i] = dsp_audio_s16_to_law[
(dsp_audio_law_to_s32[i] * denum[1] / num[1]) & 0xffff];
dsp_audio_reduce1[i] = dsp_audio_s16_to_law[
(dsp_audio_law_to_s32[i] * denum[0] / num[0]) & 0xffff];
sample = dsp_audio_law_to_s32[i] * num[0] / denum[0];
if (sample < -32768)
sample = -32768;
else if (sample > 32767)
sample = 32767;
dsp_audio_increase1[i] = dsp_audio_s16_to_law[sample & 0xffff];
sample = dsp_audio_law_to_s32[i] * num[1] / denum[1];
if (sample < -32768)
sample = -32768;
else if (sample > 32767)
sample = 32767;
dsp_audio_increase2[i] = dsp_audio_s16_to_law[sample & 0xffff];
sample = dsp_audio_law_to_s32[i] * num[2] / denum[2];
if (sample < -32768)
sample = -32768;
else if (sample > 32767)
sample = 32767;
dsp_audio_increase3[i] = dsp_audio_s16_to_law[sample & 0xffff];
sample = dsp_audio_law_to_s32[i] * num[3] / denum[3];
if (sample < -32768)
sample = -32768;
else if (sample > 32767)
sample = 32767;
dsp_audio_increase4[i] = dsp_audio_s16_to_law[sample & 0xffff];
sample = dsp_audio_law_to_s32[i] * num[4] / denum[4];
if (sample < -32768)
sample = -32768;
else if (sample > 32767)
sample = 32767;
dsp_audio_increase5[i] = dsp_audio_s16_to_law[sample & 0xffff];
sample = dsp_audio_law_to_s32[i] * num[5] / denum[5];
if (sample < -32768)
sample = -32768;
else if (sample > 32767)
sample = 32767;
dsp_audio_increase6[i] = dsp_audio_s16_to_law[sample & 0xffff];
sample = dsp_audio_law_to_s32[i] * num[6] / denum[6];
if (sample < -32768)
sample = -32768;
else if (sample > 32767)
sample = 32767;
dsp_audio_increase7[i] = dsp_audio_s16_to_law[sample & 0xffff];
sample = dsp_audio_law_to_s32[i] * num[7] / denum[7];
if (sample < -32768)
sample = -32768;
else if (sample > 32767)
sample = 32767;
dsp_audio_increase8[i] = dsp_audio_s16_to_law[sample & 0xffff];
i++;
}
}
/**************************************
* change the volume of the given skb *
**************************************/
/* this is a helper function for changing volume of skb. the range may be
* -8 to 8, which is a shift to the power of 2. 0 == no volume, 3 == volume*8
*/
void
dsp_change_volume(struct sk_buff *skb, int volume)
{
u8 *volume_change;
int i, ii;
u8 *p;
int shift;
if (volume == 0)
return;
/* get correct conversion table */
if (volume < 0) {
shift = volume + 8;
if (shift < 0)
shift = 0;
} else {
shift = volume + 7;
if (shift > 15)
shift = 15;
}
volume_change = dsp_audio_volume_change[shift];
i = 0;
ii = skb->len;
p = skb->data;
/* change volume */
while (i < ii) {
*p = volume_change[*p];
p++;
i++;
}
}