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rt-thread-official/bsp/stm32_radio/libwma/wmadeci.c

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add libwma codec. git-svn-id: https://rt-thread.googlecode.com/svn/trunk@26 bbd45198-f89e-11dd-88c7-29a3b14d5316
2009-08-25 15:03:11 +08:00
/*
* WMA compatible decoder
* Copyright (c) 2002 The FFmpeg Project.
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
/**
* @file wmadec.c
* WMA compatible decoder.
*/
//#include <codecs.h>
#include "codeclib.h"
#include "asf.h"
#include "wmadec.h"
#include "wmafixed.h"
#include "wmabitstream.h"
#define VLCBITS 7 /*7 is the lowest without glitching*/
#define VLCMAX ((22+VLCBITS-1)/VLCBITS)
#define EXPVLCBITS 7
#define EXPMAX ((19+EXPVLCBITS-1)/EXPVLCBITS)
#define HGAINVLCBITS 9
#define HGAINMAX ((13+HGAINVLCBITS-1)/HGAINVLCBITS)
typedef struct CoefVLCTable
{
int n; /* total number of codes */
const uint32_t *huffcodes; /* VLC bit values */
const uint8_t *huffbits; /* VLC bit size */
const uint16_t *levels; /* table to build run/level tables */
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} CoefVLCTable;
add libwma codec. git-svn-id: https://rt-thread.googlecode.com/svn/trunk@26 bbd45198-f89e-11dd-88c7-29a3b14d5316
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static void wma_lsp_to_curve_init(WMADecodeContext *s, int frame_len);
fixed32 coefsarray[MAX_CHANNELS][BLOCK_MAX_SIZE] ;
/*decode and window into IRAM on targets with at least 80KB of codec IRAM*/
fixed32 frame_out_buf[MAX_CHANNELS][BLOCK_MAX_SIZE * 2] ;
//static variables that replace malloced stuff
fixed32 stat0[2048], stat1[1024], stat2[512], stat3[256], stat4[128]; //these are the MDCT reconstruction windows
uint16_t *runtabarray[2], *levtabarray[2]; //these are VLC lookup tables
uint16_t runtab0[1336], runtab1[1336], levtab0[1336], levtab1[1336]; //these could be made smaller since only one can be 1336
#define VLCBUF1SIZE 4598
#define VLCBUF2SIZE 3574
#define VLCBUF3SIZE 360
#define VLCBUF4SIZE 540
/*putting these in IRAM actually makes PP slower*/
VLC_TYPE vlcbuf1[VLCBUF1SIZE][2];
VLC_TYPE vlcbuf2[VLCBUF2SIZE][2];
VLC_TYPE vlcbuf3[VLCBUF3SIZE][2];
VLC_TYPE vlcbuf4[VLCBUF4SIZE][2];
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#include "wmadata.h"
add libwma codec. git-svn-id: https://rt-thread.googlecode.com/svn/trunk@26 bbd45198-f89e-11dd-88c7-29a3b14d5316
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/*
* Helper functions for wma_window.
*
*
*/
#ifdef CPU_ARM
static inline
void vector_fmul_add_add(fixed32 *dst, const fixed32 *data,
const fixed32 *window, int n)
{
/* Block sizes are always power of two */
asm volatile (
"0:"
"ldmia %[d]!, {r0, r1};"
"ldmia %[w]!, {r4, r5};"
/* consume the first data and window value so we can use those
* registers again */
"smull r8, r9, r0, r4;"
"ldmia %[dst], {r0, r4};"
"add r0, r0, r9, lsl #1;" /* *dst=*dst+(r9<<1)*/
"smull r8, r9, r1, r5;"
"add r1, r4, r9, lsl #1;"
"stmia %[dst]!, {r0, r1};"
"subs %[n], %[n], #2;"
"bne 0b;"
: [d] "+r" (data), [w] "+r" (window), [dst] "+r" (dst), [n] "+r" (n)
: : "r0", "r1", "r4", "r5", "r8", "r9", "memory", "cc");
}
static inline
void vector_fmul_reverse(fixed32 *dst, const fixed32 *src0, const fixed32 *src1,
int len)
{
/* Block sizes are always power of two */
asm volatile (
"add %[s1], %[s1], %[n], lsl #2;"
"0:"
"ldmia %[s0]!, {r0, r1};"
"ldmdb %[s1]!, {r4, r5};"
"smull r8, r9, r0, r5;"
"mov r0, r9, lsl #1;"
"smull r8, r9, r1, r4;"
"mov r1, r9, lsl #1;"
"stmia %[dst]!, {r0, r1};"
"subs %[n], %[n], #2;"
"bne 0b;"
: [s0] "+r" (src0), [s1] "+r" (src1), [dst] "+r" (dst), [n] "+r" (len)
: : "r0", "r1", "r4", "r5", "r8", "r9", "memory", "cc");
}
#elif defined(CPU_COLDFIRE)
static inline
void vector_fmul_add_add(fixed32 *dst, const fixed32 *data,
const fixed32 *window, int n)
{
/* Block sizes are always power of two. Smallest block is always way bigger
* than four too.*/
asm volatile (
"0:"
"movem.l (%[d]), %%d0-%%d3;"
"movem.l (%[w]), %%d4-%%d5/%%a0-%%a1;"
"mac.l %%d0, %%d4, %%acc0;"
"mac.l %%d1, %%d5, %%acc1;"
"mac.l %%d2, %%a0, %%acc2;"
"mac.l %%d3, %%a1, %%acc3;"
"lea.l (16, %[d]), %[d];"
"lea.l (16, %[w]), %[w];"
"movclr.l %%acc0, %%d0;"
"movclr.l %%acc1, %%d1;"
"movclr.l %%acc2, %%d2;"
"movclr.l %%acc3, %%d3;"
"movem.l (%[dst]), %%d4-%%d5/%%a0-%%a1;"
"add.l %%d4, %%d0;"
"add.l %%d5, %%d1;"
"add.l %%a0, %%d2;"
"add.l %%a1, %%d3;"
"movem.l %%d0-%%d3, (%[dst]);"
"lea.l (16, %[dst]), %[dst];"
"subq.l #4, %[n];"
"jne 0b;"
: [d] "+a" (data), [w] "+a" (window), [dst] "+a" (dst), [n] "+d" (n)
: : "d0", "d1", "d2", "d3", "d4", "d5", "a0", "a1", "memory", "cc");
}
static inline
void vector_fmul_reverse(fixed32 *dst, const fixed32 *src0, const fixed32 *src1,
int len)
{
/* Block sizes are always power of two. Smallest block is always way bigger
* than four too.*/
asm volatile (
"lea.l (-16, %[s1], %[n]*4), %[s1];"
"0:"
"movem.l (%[s0]), %%d0-%%d3;"
"movem.l (%[s1]), %%d4-%%d5/%%a0-%%a1;"
"mac.l %%d0, %%a1, %%acc0;"
"mac.l %%d1, %%a0, %%acc1;"
"mac.l %%d2, %%d5, %%acc2;"
"mac.l %%d3, %%d4, %%acc3;"
"lea.l (16, %[s0]), %[s0];"
"lea.l (-16, %[s1]), %[s1];"
"movclr.l %%acc0, %%d0;"
"movclr.l %%acc1, %%d1;"
"movclr.l %%acc2, %%d2;"
"movclr.l %%acc3, %%d3;"
"movem.l %%d0-%%d3, (%[dst]);"
"lea.l (16, %[dst]), %[dst];"
"subq.l #4, %[n];"
"jne 0b;"
: [s0] "+a" (src0), [s1] "+a" (src1), [dst] "+a" (dst), [n] "+d" (len)
: : "d0", "d1", "d2", "d3", "d4", "d5", "a0", "a1", "memory", "cc");
}
#else
static inline void vector_fmul_add_add(fixed32 *dst, const fixed32 *src0, const fixed32 *src1, int len){
int i;
for(i=0; i<len; i++)
dst[i] = fixmul32b(src0[i], src1[i]) + dst[i];
}
static inline void vector_fmul_reverse(fixed32 *dst, const fixed32 *src0, const fixed32 *src1, int len){
int i;
src1 += len-1;
for(i=0; i<len; i++)
dst[i] = fixmul32b(src0[i], src1[-i]);
}
#endif
/**
* Apply MDCT window and add into output.
*
* We ensure that when the windows overlap their squared sum
* is always 1 (MDCT reconstruction rule).
*
* The Vorbis I spec has a great diagram explaining this process.
* See section 1.3.2.3 of http://xiph.org/vorbis/doc/Vorbis_I_spec.html
*/
static void wma_window(WMADecodeContext *s, fixed32 *in, fixed32 *out)
{
//float *in = s->output;
int block_len, bsize, n;
/* left part */
/*previous block was larger, so we'll use the size of the current block to set the window size*/
if (s->block_len_bits <= s->prev_block_len_bits) {
block_len = s->block_len;
bsize = s->frame_len_bits - s->block_len_bits;
vector_fmul_add_add(out, in, s->windows[bsize], block_len);
} else {
/*previous block was smaller or the same size, so use it's size to set the window length*/
block_len = 1 << s->prev_block_len_bits;
/*find the middle of the two overlapped blocks, this will be the first overlapped sample*/
n = (s->block_len - block_len) / 2;
bsize = s->frame_len_bits - s->prev_block_len_bits;
vector_fmul_add_add(out+n, in+n, s->windows[bsize], block_len);
memcpy(out+n+block_len, in+n+block_len, n*sizeof(fixed32));
}
/* Advance to the end of the current block and prepare to window it for the next block.
* Since the window function needs to be reversed, we do it backwards starting with the
* last sample and moving towards the first
*/
out += s->block_len;
in += s->block_len;
/* right part */
if (s->block_len_bits <= s->next_block_len_bits) {
block_len = s->block_len;
bsize = s->frame_len_bits - s->block_len_bits;
vector_fmul_reverse(out, in, s->windows[bsize], block_len);
} else {
block_len = 1 << s->next_block_len_bits;
n = (s->block_len - block_len) / 2;
bsize = s->frame_len_bits - s->next_block_len_bits;
memcpy(out, in, n*sizeof(fixed32));
vector_fmul_reverse(out+n, in+n, s->windows[bsize], block_len);
memset(out+n+block_len, 0, n*sizeof(fixed32));
}
}
/* XXX: use same run/length optimization as mpeg decoders */
static void init_coef_vlc(VLC *vlc,
uint16_t **prun_table, uint16_t **plevel_table,
const CoefVLCTable *vlc_table, int tab)
{
int n = vlc_table->n;
const uint8_t *table_bits = vlc_table->huffbits;
const uint32_t *table_codes = vlc_table->huffcodes;
const uint16_t *levels_table = vlc_table->levels;
uint16_t *run_table, *level_table;
const uint16_t *p;
int i, l, j, level;
init_vlc(vlc, VLCBITS, n, table_bits, 1, 1, table_codes, 4, 4, 0);
run_table = runtabarray[tab];
level_table= levtabarray[tab];
p = levels_table;
i = 2;
level = 1;
while (i < n)
{
l = *p++;
for(j=0;j<l;++j)
{
run_table[i] = j;
level_table[i] = level;
++i;
}
++level;
}
*prun_table = run_table;
*plevel_table = level_table;
}
int wma_decode_init(WMADecodeContext* s, asf_waveformatex_t *wfx)
{
int i, flags1, flags2;
fixed32 *window;
uint8_t *extradata;
fixed64 bps1;
fixed32 high_freq;
fixed64 bps;
int sample_rate1;
update wma example git-svn-id: https://rt-thread.googlecode.com/svn/trunk@44 bbd45198-f89e-11dd-88c7-29a3b14d5316
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int coef_vlc_table;
add libwma codec. git-svn-id: https://rt-thread.googlecode.com/svn/trunk@26 bbd45198-f89e-11dd-88c7-29a3b14d5316
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fixed32 *temp[5];
#ifdef CPU_COLDFIRE
coldfire_set_macsr(EMAC_FRACTIONAL | EMAC_SATURATE);
#endif
/*clear stereo setting to avoid glitches when switching stereo->mono*/
s->channel_coded[0]=0;
s->channel_coded[1]=0;
s->ms_stereo=0;
s->sample_rate = wfx->rate;
s->nb_channels = wfx->channels;
s->bit_rate = wfx->bitrate;
s->block_align = wfx->blockalign;
s->coefs = &coefsarray;
s->frame_out = &frame_out_buf;
if (wfx->codec_id == ASF_CODEC_ID_WMAV1) {
s->version = 1;
} else if (wfx->codec_id == ASF_CODEC_ID_WMAV2 ) {
s->version = 2;
} else {
/*one of those other wma flavors that don't have GPLed decoders */
return -1;
}
/* extract flag infos */
flags1 = 0;
flags2 = 0;
extradata = wfx->data;
if (s->version == 1 && wfx->datalen >= 4) {
flags1 = extradata[0] | (extradata[1] << 8);
flags2 = extradata[2] | (extradata[3] << 8);
}else if (s->version == 2 && wfx->datalen >= 6){
flags1 = extradata[0] | (extradata[1] << 8) |
(extradata[2] << 16) | (extradata[3] << 24);
flags2 = extradata[4] | (extradata[5] << 8);
}
s->use_exp_vlc = flags2 & 0x0001;
s->use_bit_reservoir = flags2 & 0x0002;
s->use_variable_block_len = flags2 & 0x0004;
/* compute MDCT block size */
if (s->sample_rate <= 16000){
s->frame_len_bits = 9;
}else if (s->sample_rate <= 22050 ||
(s->sample_rate <= 32000 && s->version == 1)){
s->frame_len_bits = 10;
}else{
s->frame_len_bits = 11;
}
s->frame_len = 1 << s->frame_len_bits;
if (s-> use_variable_block_len)
{
int nb_max, nb;
nb = ((flags2 >> 3) & 3) + 1;
if ((s->bit_rate / s->nb_channels) >= 32000)
{
nb += 2;
}
nb_max = s->frame_len_bits - BLOCK_MIN_BITS; //max is 11-7
if (nb > nb_max)
nb = nb_max;
s->nb_block_sizes = nb + 1;
}
else
{
s->nb_block_sizes = 1;
}
/* init rate dependant parameters */
s->use_noise_coding = 1;
high_freq = itofix64(s->sample_rate) >> 1;
/* if version 2, then the rates are normalized */
sample_rate1 = s->sample_rate;
if (s->version == 2)
{
if (sample_rate1 >= 44100)
sample_rate1 = 44100;
else if (sample_rate1 >= 22050)
sample_rate1 = 22050;
else if (sample_rate1 >= 16000)
sample_rate1 = 16000;
else if (sample_rate1 >= 11025)
sample_rate1 = 11025;
else if (sample_rate1 >= 8000)
sample_rate1 = 8000;
update wma example git-svn-id: https://rt-thread.googlecode.com/svn/trunk@44 bbd45198-f89e-11dd-88c7-29a3b14d5316
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}
{
fixed64 tmp, tmp2;
fixed64 tim, tmpi;
add more assemble implementation in wma codec. git-svn-id: https://rt-thread.googlecode.com/svn/trunk@27 bbd45198-f89e-11dd-88c7-29a3b14d5316
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add libwma codec. git-svn-id: https://rt-thread.googlecode.com/svn/trunk@26 bbd45198-f89e-11dd-88c7-29a3b14d5316
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tmp = itofix64(s->bit_rate);
tmp2 = itofix64(s->nb_channels * s->sample_rate);
bps = fixdiv64(tmp, tmp2);
tim = bps * s->frame_len;
tmpi = fixdiv64(tim,itofix64(8));
update wma example git-svn-id: https://rt-thread.googlecode.com/svn/trunk@44 bbd45198-f89e-11dd-88c7-29a3b14d5316
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s->byte_offset_bits = av_log2(fixtoi64(tmpi+0x8000)) + 2;
}
add libwma codec. git-svn-id: https://rt-thread.googlecode.com/svn/trunk@26 bbd45198-f89e-11dd-88c7-29a3b14d5316
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/* compute high frequency value and choose if noise coding should
be activated */
bps1 = bps;
if (s->nb_channels == 2)
bps1 = fixmul32(bps,0x1999a);
if (sample_rate1 == 44100)
{
if (bps1 >= 0x9c29)
s->use_noise_coding = 0;
else
high_freq = fixmul32(high_freq,0x6666);
}
else if (sample_rate1 == 22050)
{
if (bps1 >= 0x128f6)
s->use_noise_coding = 0;
else if (bps1 >= 0xb852)
high_freq = fixmul32(high_freq,0xb333);
else
high_freq = fixmul32(high_freq,0x999a);
}
else if (sample_rate1 == 16000)
{
if (bps > 0x8000)
high_freq = fixmul32(high_freq,0x8000);
else
high_freq = fixmul32(high_freq,0x4ccd);
}
else if (sample_rate1 == 11025)
{
high_freq = fixmul32(high_freq,0xb333);
}
else if (sample_rate1 == 8000)
{
if (bps <= 0xa000)
{
high_freq = fixmul32(high_freq,0x8000);
}
else if (bps > 0xc000)
{
s->use_noise_coding = 0;
}
else
{
high_freq = fixmul32(high_freq,0xa666);
}
}
else
{
if (bps >= 0xcccd)
{
high_freq = fixmul32(high_freq,0xc000);
}
else if (bps >= 0x999a)
{
high_freq = fixmul32(high_freq,0x999a);
}
else
{
high_freq = fixmul32(high_freq,0x8000);
}
}
/* compute the scale factor band sizes for each MDCT block size */
{
int a, b, pos, lpos, k, block_len, i, j, n;
update wma example git-svn-id: https://rt-thread.googlecode.com/svn/trunk@44 bbd45198-f89e-11dd-88c7-29a3b14d5316
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const uint8_t *table;
fixed32 tmp1, tmp2;
add libwma codec. git-svn-id: https://rt-thread.googlecode.com/svn/trunk@26 bbd45198-f89e-11dd-88c7-29a3b14d5316
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if (s->version == 1)
{
s->coefs_start = 3;
}
else
{
s->coefs_start = 0;
}
for(k = 0; k < s->nb_block_sizes; ++k)
{
block_len = s->frame_len >> k;
if (s->version == 1)
{
lpos = 0;
for(i=0;i<25;++i)
{
a = wma_critical_freqs[i];
b = s->sample_rate;
pos = ((block_len * 2 * a) + (b >> 1)) / b;
if (pos > block_len)
pos = block_len;
s->exponent_bands[0][i] = pos - lpos;
if (pos >= block_len)
{
++i;
break;
}
lpos = pos;
}
s->exponent_sizes[0] = i;
}
else
{
/* hardcoded tables */
table = NULL;
a = s->frame_len_bits - BLOCK_MIN_BITS - k;
if (a < 3)
{
if (s->sample_rate >= 44100)
table = exponent_band_44100[a];
else if (s->sample_rate >= 32000)
table = exponent_band_32000[a];
else if (s->sample_rate >= 22050)
table = exponent_band_22050[a];
}
if (table)
{
n = *table++;
for(i=0;i<n;++i)
s->exponent_bands[k][i] = table[i];
s->exponent_sizes[k] = n;
}
else
{
j = 0;
lpos = 0;
for(i=0;i<25;++i)
{
a = wma_critical_freqs[i];
b = s->sample_rate;
pos = ((block_len * 2 * a) + (b << 1)) / (4 * b);
pos <<= 2;
if (pos > block_len)
pos = block_len;
if (pos > lpos)
s->exponent_bands[k][j++] = pos - lpos;
if (pos >= block_len)
break;
lpos = pos;
}
s->exponent_sizes[k] = j;
}
}
/* max number of coefs */
s->coefs_end[k] = (s->frame_len - ((s->frame_len * 9) / 100)) >> k;
/* high freq computation */
tmp1 = high_freq*2; /* high_freq is a fixed32!*/
tmp2 = itofix32(s->sample_rate>>1);
s->high_band_start[k] = fixtoi32( fixdiv32(tmp1, tmp2) * (block_len>>1) +0x8000);
/*
s->high_band_start[k] = (int)((block_len * 2 * high_freq) /
s->sample_rate + 0.5);*/
n = s->exponent_sizes[k];
j = 0;
pos = 0;
for(i=0;i<n;++i)
{
int start, end;
start = pos;
pos += s->exponent_bands[k][i];
end = pos;
if (start < s->high_band_start[k])
start = s->high_band_start[k];
if (end > s->coefs_end[k])
end = s->coefs_end[k];
if (end > start)
s->exponent_high_bands[k][j++] = end - start;
}
s->exponent_high_sizes[k] = j;
}
}
/*Not using the ffmpeg IMDCT anymore*/
/* mdct_init_global();
for(i = 0; i < s->nb_block_sizes; ++i)
{
ff_mdct_init(&s->mdct_ctx[i], s->frame_len_bits - i + 1, 1);
}
*/
/*ffmpeg uses malloc to only allocate as many window sizes as needed. However, we're really only interested in the worst case memory usage.
* In the worst case you can have 5 window sizes, 128 doubling up 2048
* Smaller windows are handled differently.
* Since we don't have malloc, just statically allocate this
*/
temp[0] = stat0;
temp[1] = stat1;
temp[2] = stat2;
temp[3] = stat3;
temp[4] = stat4;
/* init MDCT windows : simple sinus window */
for(i = 0; i < s->nb_block_sizes; i++)
{
int n, j;
fixed32 alpha;
n = 1 << (s->frame_len_bits - i);
//window = av_malloc(sizeof(fixed32) * n);
window = temp[i];
//fixed32 n2 = itofix32(n<<1); //2x the window length
//alpha = fixdiv32(M_PI_F, n2); //PI / (2x Window length) == PI<<(s->frame_len_bits - i+1)
//alpha = M_PI_F>>(s->frame_len_bits - i+1);
alpha = (1<<15)>>(s->frame_len_bits - i+1); /* this calculates 0.5/(2*n) */
for(j=0;j<n;++j)
{
fixed32 j2 = itofix32(j) + 0x8000;
window[j] = fsincos(fixmul32(j2,alpha)<<16, 0); //alpha between 0 and pi/2
}
s->windows[i] = window;
}
s->reset_block_lengths = 1;
if (s->use_noise_coding)
{
/* init the noise generator */
if (s->use_exp_vlc)
{
s->noise_mult = 0x51f;
s->noise_table = noisetable_exp;
}
else
{
s->noise_mult = 0xa3d;
/* LSP values are simply 2x the EXP values */
for (i=0;i<NOISE_TAB_SIZE;++i)
noisetable_exp[i] = noisetable_exp[i]<< 1;
s->noise_table = noisetable_exp;
}
s->hgain_vlc.table = vlcbuf4;
s->hgain_vlc.table_allocated = VLCBUF4SIZE;
init_vlc(&s->hgain_vlc, HGAINVLCBITS, sizeof(hgain_huffbits),
hgain_huffbits, 1, 1,
hgain_huffcodes, 2, 2, 0);
}
if (s->use_exp_vlc)
{
s->exp_vlc.table = vlcbuf3;
s->exp_vlc.table_allocated = VLCBUF3SIZE;
init_vlc(&s->exp_vlc, EXPVLCBITS, sizeof(scale_huffbits),
scale_huffbits, 1, 1,
scale_huffcodes, 4, 4, 0);
}
else
{
wma_lsp_to_curve_init(s, s->frame_len);
}
/* choose the VLC tables for the coefficients */
coef_vlc_table = 2;
if (s->sample_rate >= 32000)
{
if (bps1 < 0xb852)
coef_vlc_table = 0;
else if (bps1 < 0x128f6)
coef_vlc_table = 1;
}
runtabarray[0] = runtab0; runtabarray[1] = runtab1;
levtabarray[0] = levtab0; levtabarray[1] = levtab1;
s->coef_vlc[0].table = vlcbuf1;
s->coef_vlc[0].table_allocated = VLCBUF1SIZE;
s->coef_vlc[1].table = vlcbuf2;
s->coef_vlc[1].table_allocated = VLCBUF2SIZE;
init_coef_vlc(&s->coef_vlc[0], &s->run_table[0], &s->level_table[0],
&coef_vlcs[coef_vlc_table * 2], 0);
init_coef_vlc(&s->coef_vlc[1], &s->run_table[1], &s->level_table[1],
&coef_vlcs[coef_vlc_table * 2 + 1], 1);
s->last_superframe_len = 0;
s->last_bitoffset = 0;
return 0;
}
/* compute x^-0.25 with an exponent and mantissa table. We use linear
interpolation to reduce the mantissa table size at a small speed
expense (linear interpolation approximately doubles the number of
bits of precision). */
static inline fixed32 pow_m1_4(WMADecodeContext *s, fixed32 x)
{
union {
float f;
unsigned int v;
} u, t;
unsigned int e, m;
fixed32 a, b;
u.f = fixtof64(x);
e = u.v >> 23;
m = (u.v >> (23 - LSP_POW_BITS)) & ((1 << LSP_POW_BITS) - 1);
/* build interpolation scale: 1 <= t < 2. */
t.v = ((u.v << LSP_POW_BITS) & ((1 << 23) - 1)) | (127 << 23);
a = s->lsp_pow_m_table1[m];
b = s->lsp_pow_m_table2[m];
/* lsp_pow_e_table contains 32.32 format */
/* TODO: Since we're unlikely have value that cover the whole
* IEEE754 range, we probably don't need to have all possible exponents */
return (lsp_pow_e_table[e] * (a + fixmul32(b, ftofix32(t.f))) >>32);
}
static void wma_lsp_to_curve_init(WMADecodeContext *s, int frame_len)
{
fixed32 a, b, temp, temp2;
int i, ix = 0;
// wdel = fixdiv32(M_PI_F, itofix32(frame_len));
temp = fixdiv32(itofix32(1), itofix32(frame_len));
for (i=0; i<frame_len; ++i)
{
/* TODO: can probably reuse the trig_init values here */
fsincos((temp*i)<<15, &temp2);
/* get 3 bits headroom + 1 bit from not doubleing the values */
s->lsp_cos_table[i] = temp2>>3;
}
/* NOTE: these two tables are needed to avoid two operations in
pow_m1_4 */
b = itofix32(1);
/*double check this later*/
for(i=(1 << LSP_POW_BITS) - 1;i>=0;i--)
{
// m = (1 << LSP_POW_BITS) + i;
a = pow_a_table[ix++]<<4;
s->lsp_pow_m_table1[i] = 2 * a - b;
s->lsp_pow_m_table2[i] = b - a;
b = a;
}
}
/* NOTE: We use the same code as Vorbis here */
/* XXX: optimize it further with SSE/3Dnow */
static void wma_lsp_to_curve(WMADecodeContext *s,
fixed32 *out,
fixed32 *val_max_ptr,
int n,
fixed32 *lsp)
{
int i, j;
fixed32 p, q, w, v, val_max, temp, temp2;
val_max = 0;
for(i=0;i<n;++i)
{
/* shift by 2 now to reduce rounding error,
* we can renormalize right before pow_m1_4
*/
p = 0x8000<<5;
q = 0x8000<<5;
w = s->lsp_cos_table[i];
for (j=1;j<NB_LSP_COEFS;j+=2)
{
/* w is 5.27 format, lsp is in 16.16, temp2 becomes 5.27 format */
temp2 = ((w - (lsp[j - 1]<<11)));
temp = q;
/* q is 16.16 format, temp2 is 5.27, q becomes 16.16 */
q = fixmul32b(q, temp2 )<<4;
p = fixmul32b(p, (w - (lsp[j]<<11)))<<4;
}
/* 2 in 5.27 format is 0x10000000 */
p = fixmul32(p, fixmul32b(p, (0x10000000 - w)))<<3;
q = fixmul32(q, fixmul32b(q, (0x10000000 + w)))<<3;
v = (p + q) >>9; /* p/q end up as 16.16 */
v = pow_m1_4(s, v);
if (v > val_max)
val_max = v;
out[i] = v;
}
*val_max_ptr = val_max;
}
/* decode exponents coded with LSP coefficients (same idea as Vorbis) */
static void decode_exp_lsp(WMADecodeContext *s, int ch)
{
fixed32 lsp_coefs[NB_LSP_COEFS];
int val, i;
for (i = 0; i < NB_LSP_COEFS; ++i)
{
if (i == 0 || i >= 8)
val = get_bits(&s->gb, 3);
else
val = get_bits(&s->gb, 4);
lsp_coefs[i] = lsp_codebook[i][val];
}
wma_lsp_to_curve(s,
s->exponents[ch],
&s->max_exponent[ch],
s->block_len,
lsp_coefs);
}
/* decode exponents coded with VLC codes */
static int decode_exp_vlc(WMADecodeContext *s, int ch)
{
int last_exp, n, code;
const uint16_t *ptr, *band_ptr;
fixed32 v, max_scale;
fixed32 *q,*q_end;
/*accommodate the 60 negative indices */
const fixed32 *pow_10_to_yover16_ptr = &pow_10_to_yover16[61];
band_ptr = s->exponent_bands[s->frame_len_bits - s->block_len_bits];
ptr = band_ptr;
q = s->exponents[ch];
q_end = q + s->block_len;
max_scale = 0;
if (s->version == 1) //wmav1 only
{
last_exp = get_bits(&s->gb, 5) + 10;
/* XXX: use a table */
v = pow_10_to_yover16_ptr[last_exp];
max_scale = v;
n = *ptr++;
do
{
*q++ = v;
}
while (--n);
} else {
last_exp = 36;
}
while (q < q_end)
{
code = get_vlc2(&s->gb, s->exp_vlc.table, EXPVLCBITS, EXPMAX);
if (code < 0)
{
return -1;
}
/* NOTE: this offset is the same as MPEG4 AAC ! */
last_exp += code - 60;
/* XXX: use a table */
v = pow_10_to_yover16_ptr[last_exp];
if (v > max_scale)
{
max_scale = v;
}
n = *ptr++;
do
{
*q++ = v;
}
while (--n);
}
s->max_exponent[ch] = max_scale;
return 0;
}
/* return 0 if OK. return 1 if last block of frame. return -1 if
unrecorrable error. */
static int wma_decode_block(WMADecodeContext *s, int32_t *scratch_buffer)
{
int n, v, a, ch, code, bsize;
int coef_nb_bits, total_gain;
int nb_coefs[MAX_CHANNELS];
fixed32 mdct_norm;
/*DEBUGF("***decode_block: %d (%d samples of %d in frame)\n", s->block_num, s->block_len, s->frame_len);*/
/* compute current block length */
if (s->use_variable_block_len)
{
n = av_log2(s->nb_block_sizes - 1) + 1;
if (s->reset_block_lengths)
{
s->reset_block_lengths = 0;
v = get_bits(&s->gb, n);
if (v >= s->nb_block_sizes)
{
return -2;
}
s->prev_block_len_bits = s->frame_len_bits - v;
v = get_bits(&s->gb, n);
if (v >= s->nb_block_sizes)
{
return -3;
}
s->block_len_bits = s->frame_len_bits - v;
}
else
{
/* update block lengths */
s->prev_block_len_bits = s->block_len_bits;
s->block_len_bits = s->next_block_len_bits;
}
v = get_bits(&s->gb, n);
if (v >= s->nb_block_sizes)
{
// rb->splash(HZ*4, "v was %d", v); //5, 7
return -4; //this is it
}
else{
//rb->splash(HZ, "passed v block (%d)!", v);
}
s->next_block_len_bits = s->frame_len_bits - v;
}
else
{
/* fixed block len */
s->next_block_len_bits = s->frame_len_bits;
s->prev_block_len_bits = s->frame_len_bits;
s->block_len_bits = s->frame_len_bits;
}
/* now check if the block length is coherent with the frame length */
s->block_len = 1 << s->block_len_bits;
if ((s->block_pos + s->block_len) > s->frame_len)
{
return -5; //oddly 32k sample from tracker fails here
}
if (s->nb_channels == 2)
{
s->ms_stereo = get_bits1(&s->gb);
}
v = 0;
for (ch = 0; ch < s->nb_channels; ++ch)
{
a = get_bits1(&s->gb);
s->channel_coded[ch] = a;
v |= a;
}
/* if no channel coded, no need to go further */
/* XXX: fix potential framing problems */
if (!v)
{
goto next;
}
bsize = s->frame_len_bits - s->block_len_bits;
/* read total gain and extract corresponding number of bits for
coef escape coding */
total_gain = 1;
for(;;)
{
a = get_bits(&s->gb, 7);
total_gain += a;
if (a != 127)
{
break;
}
}
if (total_gain < 15)
coef_nb_bits = 13;
else if (total_gain < 32)
coef_nb_bits = 12;
else if (total_gain < 40)
coef_nb_bits = 11;
else if (total_gain < 45)
coef_nb_bits = 10;
else
coef_nb_bits = 9;
/* compute number of coefficients */
n = s->coefs_end[bsize] - s->coefs_start;
for(ch = 0; ch < s->nb_channels; ++ch)
{
nb_coefs[ch] = n;
}
/* complex coding */
if (s->use_noise_coding)
{
for(ch = 0; ch < s->nb_channels; ++ch)
{
if (s->channel_coded[ch])
{
int i, n, a;
n = s->exponent_high_sizes[bsize];
for(i=0;i<n;++i)
{
a = get_bits1(&s->gb);
s->high_band_coded[ch][i] = a;
/* if noise coding, the coefficients are not transmitted */
if (a)
nb_coefs[ch] -= s->exponent_high_bands[bsize][i];
}
}
}
for(ch = 0; ch < s->nb_channels; ++ch)
{
if (s->channel_coded[ch])
{
int i, n, val, code;
n = s->exponent_high_sizes[bsize];
val = (int)0x80000000;
for(i=0;i<n;++i)
{
if (s->high_band_coded[ch][i])
{
if (val == (int)0x80000000)
{
val = get_bits(&s->gb, 7) - 19;
}
else
{
//code = get_vlc(&s->gb, &s->hgain_vlc);
code = get_vlc2(&s->gb, s->hgain_vlc.table, HGAINVLCBITS, HGAINMAX);
if (code < 0)
{
return -6;
}
val += code - 18;
}
s->high_band_values[ch][i] = val;
}
}
}
}
}
/* exponents can be reused in short blocks. */
if ((s->block_len_bits == s->frame_len_bits) || get_bits1(&s->gb))
{
for(ch = 0; ch < s->nb_channels; ++ch)
{
if (s->channel_coded[ch])
{
if (s->use_exp_vlc)
{
if (decode_exp_vlc(s, ch) < 0)
{
return -7;
}
}
else
{
decode_exp_lsp(s, ch);
}
s->exponents_bsize[ch] = bsize;
}
}
}
/* parse spectral coefficients : just RLE encoding */
for(ch = 0; ch < s->nb_channels; ++ch)
{
if (s->channel_coded[ch])
{
VLC *coef_vlc;
int level, run, sign, tindex;
int16_t *ptr, *eptr;
const int16_t *level_table, *run_table;
/* special VLC tables are used for ms stereo because
there is potentially less energy there */
tindex = (ch == 1 && s->ms_stereo);
coef_vlc = &s->coef_vlc[tindex];
run_table = s->run_table[tindex];
level_table = s->level_table[tindex];
/* XXX: optimize */
ptr = &s->coefs1[ch][0];
eptr = ptr + nb_coefs[ch];
memset(ptr, 0, s->block_len * sizeof(int16_t));
for(;;)
{
code = get_vlc2(&s->gb, coef_vlc->table, VLCBITS, VLCMAX);
//code = get_vlc(&s->gb, coef_vlc);
if (code < 0)
{
return -8;
}
if (code == 1)
{
/* EOB */
break;
}
else if (code == 0)
{
/* escape */
level = get_bits(&s->gb, coef_nb_bits);
/* NOTE: this is rather suboptimal. reading
block_len_bits would be better */
run = get_bits(&s->gb, s->frame_len_bits);
}
else
{
/* normal code */
run = run_table[code];
level = level_table[code];
}
sign = get_bits1(&s->gb);
if (!sign)
level = -level;
ptr += run;
if (ptr >= eptr)
{
break;
}
*ptr++ = level;
/* NOTE: EOB can be omitted */
if (ptr >= eptr)
break;
}
}
if (s->version == 1 && s->nb_channels >= 2)
{
align_get_bits(&s->gb);
}
}
{
int n4 = s->block_len >> 1;
mdct_norm = 0x10000>>(s->block_len_bits-1);
if (s->version == 1)
{
mdct_norm *= fixtoi32(fixsqrt32(itofix32(n4)));
}
}
/* finally compute the MDCT coefficients */
for(ch = 0; ch < s->nb_channels; ++ch)
{
if (s->channel_coded[ch])
{
int16_t *coefs1;
fixed32 *exponents;
fixed32 *coefs, atemp;
fixed64 mult;
fixed64 mult1;
fixed32 noise, temp1, temp2, mult2;
int i, j, n, n1, last_high_band, esize;
fixed32 exp_power[HIGH_BAND_MAX_SIZE];
//total_gain, coefs1, mdctnorm are lossless
coefs1 = s->coefs1[ch];
exponents = s->exponents[ch];
esize = s->exponents_bsize[ch];
coefs = (*(s->coefs))[ch];
n=0;
/*
* The calculation of coefs has a shift right by 2 built in. This
* prepares samples for the Tremor IMDCT which uses a slightly
* different fixed format then the ffmpeg one. If the old ffmpeg
* imdct is used, each shift storing into coefs should be reduced
* by 1.
* See SVN logs for details.
*/
if (s->use_noise_coding)
{
/*TODO: mult should be converted to 32 bit to speed up noise coding*/
mult = fixdiv64(pow_table[total_gain+20],Fixed32To64(s->max_exponent[ch]));
mult = mult* mdct_norm;
mult1 = mult;
/* very low freqs : noise */
for(i = 0;i < s->coefs_start; ++i)
{
*coefs++ = fixmul32( (fixmul32(s->noise_table[s->noise_index],
exponents[i<<bsize>>esize])>>4),Fixed32From64(mult1)) >>2;
s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1);
}
n1 = s->exponent_high_sizes[bsize];
/* compute power of high bands */
exponents = s->exponents[ch] +(s->high_band_start[bsize]<<bsize);
last_high_band = 0; /* avoid warning */
for (j=0;j<n1;++j)
{
n = s->exponent_high_bands[s->frame_len_bits -
s->block_len_bits][j];
if (s->high_band_coded[ch][j])
{
fixed32 e2, v;
e2 = 0;
for(i = 0;i < n; ++i)
{
/*v is normalized later on so its fixed format is irrelevant*/
v = exponents[i<<bsize>>esize]>>4;
e2 += fixmul32(v, v)>>3;
}
exp_power[j] = e2/n; /*n is an int...*/
last_high_band = j;
}
exponents += n<<bsize;
}
/* main freqs and high freqs */
exponents = s->exponents[ch] + (s->coefs_start<<bsize);
for(j=-1;j<n1;++j)
{
if (j < 0)
{
n = s->high_band_start[bsize] -
s->coefs_start;
}
else
{
n = s->exponent_high_bands[s->frame_len_bits -
s->block_len_bits][j];
}
if (j >= 0 && s->high_band_coded[ch][j])
{
/* use noise with specified power */
fixed32 tmp = fixdiv32(exp_power[j],exp_power[last_high_band]);
/*mult1 is 48.16, pow_table is 48.16*/
mult1 = fixmul32(fixsqrt32(tmp),
pow_table[s->high_band_values[ch][j]+20]) >> 16;
/*this step has a fairly high degree of error for some reason*/
mult1 = fixdiv64(mult1,fixmul32(s->max_exponent[ch],s->noise_mult));
mult1 = mult1*mdct_norm>>PRECISION;
for(i = 0;i < n; ++i)
{
noise = s->noise_table[s->noise_index];
s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1);
*coefs++ = fixmul32((fixmul32(exponents[i<<bsize>>esize],noise)>>4),
Fixed32From64(mult1)) >>2;
}
exponents += n<<bsize;
}
else
{
/* coded values + small noise */
for(i = 0;i < n; ++i)
{
noise = s->noise_table[s->noise_index];
s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1);
/*don't forget to renormalize the noise*/
temp1 = (((int32_t)*coefs1++)<<16) + (noise>>4);
temp2 = fixmul32(exponents[i<<bsize>>esize], mult>>18);
*coefs++ = fixmul32(temp1, temp2);
}
exponents += n<<bsize;
}
}
/* very high freqs : noise */
n = s->block_len - s->coefs_end[bsize];
mult2 = fixmul32(mult>>16,exponents[((-1<<bsize))>>esize]) ;
for (i = 0; i < n; ++i)
{
/*renormalize the noise product and then reduce to 14.18 precison*/
*coefs++ = fixmul32(s->noise_table[s->noise_index],mult2) >>6;
s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1);
}
}
else
{
/*Noise coding not used, simply convert from exp to fixed representation*/
fixed32 mult3 = (fixed32)(fixdiv64(pow_table[total_gain+20],
Fixed32To64(s->max_exponent[ch])));
mult3 = fixmul32(mult3, mdct_norm);
/*zero the first 3 coefficients for WMA V1, does nothing otherwise*/
for(i=0; i<s->coefs_start; i++)
*coefs++=0;
n = nb_coefs[ch];
/* XXX: optimize more, unrolling this loop in asm
might be a good idea */
for(i = 0;i < n; ++i)
{
/*ffmpeg imdct needs 15.17, while tremor 14.18*/
atemp = (coefs1[i] * mult3)>>2;
*coefs++=fixmul32(atemp,exponents[i<<bsize>>esize]);
}
n = s->block_len - s->coefs_end[bsize];
memset(coefs, 0, n*sizeof(fixed32));
}
}
}
if (s->ms_stereo && s->channel_coded[1])
{
fixed32 a, b;
int i;
fixed32 (*coefs)[MAX_CHANNELS][BLOCK_MAX_SIZE] = (s->coefs);
/* nominal case for ms stereo: we do it before mdct */
/* no need to optimize this case because it should almost
never happen */
if (!s->channel_coded[0])
{
memset((*(s->coefs))[0], 0, sizeof(fixed32) * s->block_len);
s->channel_coded[0] = 1;
}
for(i = 0; i < s->block_len; ++i)
{
a = (*coefs)[0][i];
b = (*coefs)[1][i];
(*coefs)[0][i] = a + b;
(*coefs)[1][i] = a - b;
}
}
for(ch = 0; ch < s->nb_channels; ++ch)
{
if (s->channel_coded[ch])
{
int n4, index;
n4 = s->block_len >>1;
/*faster IMDCT from Vorbis*/
mdct_backward( (1 << (s->block_len_bits+1)), (int32_t*)(*(s->coefs))[ch], (int32_t*)scratch_buffer);
/*slower but more easily understood IMDCT from FFMPEG*/
//ff_imdct_calc(&s->mdct_ctx[bsize],
// output,
// (*(s->coefs))[ch]);
/* add in the frame */
index = (s->frame_len / 2) + s->block_pos - n4;
wma_window(s, scratch_buffer, &((*s->frame_out)[ch][index]));
/* specific fast case for ms-stereo : add to second
channel if it is not coded */
if (s->ms_stereo && !s->channel_coded[1])
{
wma_window(s, scratch_buffer, &((*s->frame_out)[1][index]));
}
}
}
next:
/* update block number */
++s->block_num;
s->block_pos += s->block_len;
if (s->block_pos >= s->frame_len)
{
return 1;
}
else
{
return 0;
}
}
/* decode a frame of frame_len samples */
static int wma_decode_frame(WMADecodeContext *s, int32_t *samples)
{
int ret, i, n, ch, incr;
int32_t *ptr;
fixed32 *iptr;
/* read each block */
s->block_num = 0;
s->block_pos = 0;
for(;;)
{
ret = wma_decode_block(s, samples);
if (ret < 0)
{
DEBUGF("wma_decode_block failed with code %d\n", ret);
return -1;
}
if (ret)
{
break;
}
}
/* return frame with full 30-bit precision */
n = s->frame_len;
incr = s->nb_channels;
for(ch = 0; ch < s->nb_channels; ++ch)
{
ptr = samples + ch;
iptr = &((*s->frame_out)[ch][0]);
for (i=0;i<n;++i)
{
*ptr = (*iptr++);
ptr += incr;
}
memmove(&((*s->frame_out)[ch][0]), &((*s->frame_out)[ch][s->frame_len]),
s->frame_len * sizeof(fixed32));
}
return 0;
}
/* Initialise the superframe decoding */
int wma_decode_superframe_init(WMADecodeContext* s,
const uint8_t *buf, /*input*/
int buf_size)
{
if (buf_size==0)
{
s->last_superframe_len = 0;
return 0;
}
s->current_frame = 0;
init_get_bits(&s->gb, buf, buf_size*8);
if (s->use_bit_reservoir)
{
/* read super frame header */
skip_bits(&s->gb, 4); /* super frame index */
s->nb_frames = get_bits(&s->gb, 4);
if (s->last_superframe_len == 0)
s->nb_frames --;
else if (s->nb_frames == 0)
s->nb_frames++;
s->bit_offset = get_bits(&s->gb, s->byte_offset_bits + 3);
} else {
s->nb_frames = 1;
}
return 1;
}
/* Decode a single frame in the current superframe - return -1 if
there was a decoding error, or the number of samples decoded.
*/
int wma_decode_superframe_frame(WMADecodeContext* s,
int32_t* samples, /*output*/
const uint8_t *buf, /*input*/
int buf_size)
{
int pos, len;
uint8_t *q;
int done = 0;
if ((s->use_bit_reservoir) && (s->current_frame == 0))
{
if (s->last_superframe_len > 0)
{
/* add s->bit_offset bits to last frame */
if ((s->last_superframe_len + ((s->bit_offset + 7) >> 3)) >
MAX_CODED_SUPERFRAME_SIZE)
{
DEBUGF("superframe size too large error\n");
goto fail;
}
q = s->last_superframe + s->last_superframe_len;
len = s->bit_offset;
while (len > 7)
{
*q++ = (get_bits)(&s->gb, 8);
len -= 8;
}
if (len > 0)
{
*q++ = (get_bits)(&s->gb, len) << (8 - len);
}
/* XXX: s->bit_offset bits into last frame */
init_get_bits(&s->gb, s->last_superframe, MAX_CODED_SUPERFRAME_SIZE*8);
/* skip unused bits */
if (s->last_bitoffset > 0)
skip_bits(&s->gb, s->last_bitoffset);
/* this frame is stored in the last superframe and in the
current one */
if (wma_decode_frame(s, samples) < 0)
{
goto fail;
}
done = 1;
}
/* read each frame starting from s->bit_offset */
pos = s->bit_offset + 4 + 4 + s->byte_offset_bits + 3;
init_get_bits(&s->gb, buf + (pos >> 3), (MAX_CODED_SUPERFRAME_SIZE - (pos >> 3))*8);
len = pos & 7;
if (len > 0)
skip_bits(&s->gb, len);
s->reset_block_lengths = 1;
}
/* If we haven't decoded a frame yet, do it now */
if (!done)
{
if (wma_decode_frame(s, samples) < 0)
{
goto fail;
}
}
s->current_frame++;
if ((s->use_bit_reservoir) && (s->current_frame == s->nb_frames))
{
/* we copy the end of the frame in the last frame buffer */
pos = get_bits_count(&s->gb) + ((s->bit_offset + 4 + 4 + s->byte_offset_bits + 3) & ~7);
s->last_bitoffset = pos & 7;
pos >>= 3;
len = buf_size - pos;
if (len > MAX_CODED_SUPERFRAME_SIZE || len < 0)
{
DEBUGF("superframe size too large error after decoding\n");
goto fail;
}
s->last_superframe_len = len;
memcpy(s->last_superframe, buf + pos, len);
}
return s->frame_len;
fail:
/* when error, we reset the bit reservoir */
s->last_superframe_len = 0;
return -1;
}