--- /dev/null
+/*
+ * 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 <linux/export.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++;
+ }
+}
Code Review / kvmfornfv.git / blobdiff