Add mISDN DSP

Enable support for digital audio processing capability.
This module may be used for special applications that require
cross connecting of bchannels, conferencing, dtmf decoding
echo cancelation, tone generation, and Blowfish encryption and
decryption.
It may use hardware features if available.

Signed-off-by: Karsten Keil <kkeil@suse.de>

1 files changed, 434 insertions, 0 deletions

diff --git a/drivers/isdn/mISDN/dsp_audio.c b/drivers/isdn/mISDN/dsp_audio.c
new file mode 100644
index 00000000000..1c2dd569477
--- /dev/null
+++ b/drivers/isdn/mISDN/dsp_audio.c
@@ -0,0 +1,434 @@
+/*
+ * 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 */
+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++;
+        }
+}
+