1 files changed, 0 insertions, 433 deletions
diff --git a/ANDROID_3.4.5/drivers/isdn/mISDN/dsp_audio.c b/ANDROID_3.4.5/drivers/isdn/mISDN/dsp_audio.c
deleted file mode 100644
index 06022952..00000000
--- a/ANDROID_3.4.5/drivers/isdn/mISDN/dsp_audio.c
+++ /dev/null
@@ -1,433 +0,0 @@
-/*
- * 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++;
-	}
-}