diff options
Diffstat (limited to 'drivers/isdn/mISDN/dsp_audio.c')
| -rw-r--r-- | drivers/isdn/mISDN/dsp_audio.c | 434 |
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++; + } +} + |