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foxbox/lib/rbcodec/codecs/libwma/wmadeci.c
Michael Giacomelli 87d3dde15a Fix corrupt when repeatidly playing very low bitrate WMA files. 
The LSP feature in WMA requires that the noise table values be
doubled verses when it is not used.  Unfortunately, the previous
code would double the same values every time a LSP file was
decoded without first resetting them to their original values.
Change the code to check if the values are already doubled, and
then double/halve them as needed.  This is still a bit ugly,
in the future consider using the built in rockbox dither instead
of a lookup table.

Fixes playback when skipping back and forth between low and high
bitrate WMA.

Change-Id: I4c393092e4a789bc8f98d74274fe207400b9550e
Reviewed-on: http://gerrit.rockbox.org/226
Reviewed-by: Michael Giacomelli <giac2000@hotmail.com>
Tested-by: Michael Giacomelli <giac2000@hotmail.com>
2012-05-01 05:42:38 +02:00

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/*
* 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 <codecs/lib/codeclib.h>
#include <codecs/libasf/asf.h>
#include "wmadec.h"
#include "wmafixed.h"
#include "wmadata.h"
static void wma_lsp_to_curve_init(WMADecodeContext *s, int frame_len);
/*declarations of statically allocated variables used to remove malloc calls*/
static fixed32 coefsarray[MAX_CHANNELS][BLOCK_MAX_SIZE] IBSS_ATTR MEM_ALIGN_ATTR;
/*decode and window into IRAM on targets with at least 80KB of codec IRAM*/
static fixed32 frame_out_buf[MAX_CHANNELS][BLOCK_MAX_SIZE * 2] IBSS_ATTR_WMA_LARGE_IRAM MEM_ALIGN_ATTR;
/*MDCT reconstruction windows*/
static fixed32 stat0[2048] IBSS_ATTR_WMA_XL_IRAM MEM_ALIGN_ATTR;
static fixed32 stat1[1024] IBSS_ATTR_WMA_XL_IRAM MEM_ALIGN_ATTR;
static fixed32 stat2[ 512] IBSS_ATTR_WMA_XL_IRAM MEM_ALIGN_ATTR;
static fixed32 stat3[ 256] IBSS_ATTR_WMA_XL_IRAM MEM_ALIGN_ATTR;
static fixed32 stat4[ 128] IBSS_ATTR_WMA_XL_IRAM MEM_ALIGN_ATTR;
/*VLC lookup tables*/
static uint16_t *runtabarray[2];
static uint16_t *levtabarray[2];
static uint16_t runtab_big[1336] MEM_ALIGN_ATTR;
static uint16_t runtab_small[1072] MEM_ALIGN_ATTR;
static uint16_t levtab_big[1336] MEM_ALIGN_ATTR;
static uint16_t levtab_small[1072] MEM_ALIGN_ATTR;
#define VLCBUF1SIZE 4598
#define VLCBUF2SIZE 3574
#define VLCBUF3SIZE 360
#define VLCBUF4SIZE 540
/*putting these in IRAM actually makes PP slower*/
static VLC_TYPE vlcbuf1[VLCBUF1SIZE][2] IBSS_ATTR_WMA_XL_IRAM MEM_ALIGN_ATTR;
static VLC_TYPE vlcbuf2[VLCBUF2SIZE][2] MEM_ALIGN_ATTR;
/* This buffer gets reused for lsp tables */
static VLC_TYPE vlcbuf3[VLCBUF3SIZE][2] MEM_ALIGN_ATTR;
static VLC_TYPE vlcbuf4[VLCBUF4SIZE][2] MEM_ALIGN_ATTR;
/**
* 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, INIT_VLC_USE_NEW_STATIC);
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, flags2;
fixed32 *window;
uint8_t *extradata;
fixed64 bps1;
fixed32 high_freq;
fixed64 bps;
int sample_rate1;
int coef_vlc_table;
// int filehandle;
#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 */
flags2 = 0;
extradata = wfx->data;
if (s->version == 1 && wfx->datalen >= 4) {
flags2 = extradata[2] | (extradata[3] << 8);
}else if (s->version == 2 && wfx->datalen >= 6){
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;
}
fixed64 tmp = itofix64(s->bit_rate);
fixed64 tmp2 = itofix64(s->nb_channels * s->sample_rate);
bps = fixdiv64(tmp, tmp2);
fixed64 tim = bps * s->frame_len;
fixed64 tmpi = fixdiv64(tim,itofix64(8));
s->byte_offset_bits = av_log2(fixtoi64(tmpi+0x8000)) + 2;
/* 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;
const uint8_t *table;
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 */
fixed32 tmp1 = high_freq*2; /* high_freq is a fixed32!*/
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;
}
}
/* 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
*/
fixed32 *temp[5];
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 = temp[i];
/* this calculates 0.5/(2*n) */
alpha = (1<<15)>>(s->frame_len_bits - i+1);
for(j=0;j<n;++j)
{
fixed32 j2 = itofix32(j) + 0x8000;
/*alpha between 0 and pi/2*/
window[j] = fsincos(fixmul32(j2,alpha)<<16, 0);
}
s->windows[i] = window;
}
s->reset_block_lengths = 1;
if (s->use_noise_coding) /* init the noise generator */
{
/* LSP values are simply 2x the EXP values */
if (s->use_exp_vlc)
{
s->noise_mult = 0x51f;
/*unlikely, but we may have previoiusly used this table for LSP,
so halve the values if needed*/
if(noisetable_exp[0] == 0x10) {
for (i=0;i<NOISE_TAB_SIZE;++i)
noisetable_exp[i] >>= 1;
}
s->noise_table = noisetable_exp;
}
else
{
s->noise_mult = 0xa3d;
/*check that we haven't already doubled this table*/
if(noisetable_exp[0] == 0x5) {
for (i=0;i<NOISE_TAB_SIZE;++i)
noisetable_exp[i] <<= 1;
}
s->noise_table = noisetable_exp;
}
#if 0
/*TODO: Rockbox has a dither function. Consider using it for noise coding*/
/* We use a lookup table computered in advance, so no need to do this*/
{
unsigned int seed;
fixed32 norm;
seed = 1;
norm = 0; // PJJ: near as makes any diff to 0!
for (i=0;i<NOISE_TAB_SIZE;++i)
{
seed = seed * 314159 + 1;
s->noise_table[i] = itofix32((int)seed) * norm;
}
}
#endif
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, INIT_VLC_USE_NEW_STATIC);
}
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, INIT_VLC_USE_NEW_STATIC);
}
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;
}
/* since the coef2 table is the biggest and that has index 2 in coef_vlcs
it's safe to always assign like this */
runtabarray[0] = runtab_big; runtabarray[1] = runtab_small;
levtabarray[0] = levtab_big; levtabarray[1] = levtab_small;
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 = ((fixed32*)s->lsp_pow_m_table1)[m];
b = ((fixed32*)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 wdel, a, b, temp2;
int i;
wdel = fixdiv32(itofix32(1), itofix32(frame_len));
for (i=0; i<frame_len; ++i)
{
/* TODO: can probably reuse the trig_init values here */
fsincos((wdel*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);
int ix = 0;
s->lsp_pow_m_table1 = &vlcbuf3[0];
s->lsp_pow_m_table2 = &vlcbuf3[1<<LSP_POW_BITS];
/*double check this later*/
for(i=(1 << LSP_POW_BITS) - 1;i>=0;i--)
{
a = pow_a_table[ix++]<<4;
((fixed32*)s->lsp_pow_m_table1)[i] = 2 * a - b;
((fixed32*)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, 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)));
/* 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)
* only used for low bitrate (< 16kbps) files
*/
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 - used for bitrate >= 32kbps*/
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;
v = pow_10_to_yover16_ptr[last_exp];
max_scale = v;
n = *ptr++;
switch (n & 3) do {
case 0: *q++ = v;
case 3: *q++ = v;
case 2: *q++ = v;
case 1: *q++ = v;
} while ((n -= 4) > 0);
} 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;
v = pow_10_to_yover16_ptr[last_exp];
if (v > max_scale)
{
max_scale = v;
}
n = *ptr++;
switch (n & 3) do {
case 0: *q++ = v;
case 3: *q++ = v;
case 2: *q++ = v;
case 1: *q++ = v;
} while ((n -= 4) > 0);
}
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)
{
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);
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)
{
/*This case is only used for low bitrates (typically less then 32kbps)*/
/*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)
{
/* BLOCK_MAX_SIZE is 2048 (samples) and MAX_CHANNELS is 2. */
static uint32_t scratch_buf[BLOCK_MAX_SIZE * MAX_CHANNELS] IBSS_ATTR MEM_ALIGN_ATTR;
if (s->channel_coded[ch])
{
int n4, index;
n4 = s->block_len >>1;
ff_imdct_calc((s->frame_len_bits - bsize + 1),
scratch_buf,
(*(s->coefs))[ch]);
/* add in the frame */
index = (s->frame_len / 2) + s->block_pos - n4;
wma_window(s, scratch_buf, &((*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_buf, &((*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)
{
int ret;
/* read each block */
s->block_num = 0;
s->block_pos = 0;
for(;;)
{
ret = wma_decode_block(s);
if (ret < 0)
{
DEBUGF("wma_decode_block failed with code %d\n", ret);
return -1;
}
if (ret)
{
break;
}
}
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,
const uint8_t *buf, /*input*/
int buf_size)
{
int pos, len, ch;
uint8_t *q;
int done = 0;
for(ch = 0; ch < s->nb_channels; ch++)
memmove(&((*s->frame_out)[ch][0]),
&((*s->frame_out)[ch][s->frame_len]),
s->frame_len * sizeof(fixed32));
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) < 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) < 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;
}
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