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 000000000000..1c2dd5694773
--- /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++;
+        }
+}
+