RepoMirrors/ffmpeg
1
0
Fork 0
mirror of https://git.ffmpeg.org/ffmpeg.git synced 2024-12-24 00:02:52 +00:00
Code Issues Projects Releases Wiki Activity
2a2311bee3
ffmpeg/libavcodec/cook.c
Aurelien Jacobs 5902bdd8d4 cook shouldn't try to enforce the bitstream reader type 
Originally committed as revision 6080 to svn://svn.ffmpeg.org/ffmpeg/trunk
2006-08-24 21:59:20 +00:00

1311 lines
43 KiB
C
Raw Blame History

/*
* COOK compatible decoder
* Copyright (c) 2003 Sascha Sommer
* Copyright (c) 2005 Benjamin Larsson
*
* 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
/**
* @file cook.c
* Cook compatible decoder.
* This decoder handles RealNetworks, RealAudio G2 data.
* Cook is identified by the codec name cook in RM files.
*
* To use this decoder, a calling application must supply the extradata
* bytes provided from the RM container; 8+ bytes for mono streams and
* 16+ for stereo streams (maybe more).
*
* Codec technicalities (all this assume a buffer length of 1024):
* Cook works with several different techniques to achieve its compression.
* In the timedomain the buffer is divided into 8 pieces and quantized. If
* two neighboring pieces have different quantization index a smooth
* quantization curve is used to get a smooth overlap between the different
* pieces.
* To get to the transformdomain Cook uses a modulated lapped transform.
* The transform domain has 50 subbands with 20 elements each. This
* means only a maximum of 50*20=1000 coefficients are used out of the 1024
* available.
*/
#include <math.h>
#include <stddef.h>
#include <stdio.h>
#include "avcodec.h"
#include "bitstream.h"
#include "dsputil.h"
#include "cookdata.h"
/* the different Cook versions */
#define MONO_COOK1 0x1000001
#define MONO_COOK2 0x1000002
#define JOINT_STEREO 0x1000003
#define MC_COOK 0x2000000 //multichannel Cook, not supported
#define SUBBAND_SIZE 20
//#define COOKDEBUG
typedef struct {
int size;
int qidx_table1[8];
int qidx_table2[8];
} COOKgain;
typedef struct __attribute__((__packed__)){
/* codec data start */
uint32_t cookversion; //in network order, bigendian
uint16_t samples_per_frame; //amount of samples per frame per channel, bigendian
uint16_t subbands; //amount of bands used in the frequency domain, bigendian
/* Mono extradata ends here. */
uint32_t unused;
uint16_t js_subband_start; //bigendian
uint16_t js_vlc_bits; //bigendian
/* Stereo extradata ends here. */
} COOKextradata;
typedef struct {
GetBitContext gb;
/* stream data */
int nb_channels;
int joint_stereo;
int bit_rate;
int sample_rate;
int samples_per_channel;
int samples_per_frame;
int subbands;
int log2_numvector_size;
int numvector_size; //1 << log2_numvector_size;
int js_subband_start;
int total_subbands;
int num_vectors;
int bits_per_subpacket;
/* states */
int random_state;
/* transform data */
FFTContext fft_ctx;
FFTSample mlt_tmp[1024] __attribute__((aligned(16))); /* temporary storage for imlt */
float* mlt_window;
float* mlt_precos;
float* mlt_presin;
float* mlt_postcos;
int fft_size;
int fft_order;
int mlt_size; //modulated lapped transform size
/* gain buffers */
COOKgain* gain_now_ptr;
COOKgain* gain_previous_ptr;
COOKgain gain_current;
COOKgain gain_now;
COOKgain gain_previous;
COOKgain gain_channel1[2];
COOKgain gain_channel2[2];
/* VLC data */
int js_vlc_bits;
VLC envelope_quant_index[13];
VLC sqvh[7]; //scalar quantization
VLC ccpl; //channel coupling
/* generatable tables and related variables */
int gain_size_factor;
float gain_table[23];
float pow2tab[127];
float rootpow2tab[127];
/* data buffers */
uint8_t* decoded_bytes_buffer;
float mono_mdct_output[2048] __attribute__((aligned(16)));
float* previous_buffer_ptr[2];
float mono_previous_buffer1[1024];
float mono_previous_buffer2[1024];
float* decode_buf_ptr[4];
float* decode_buf_ptr2[2];
float decode_buffer_1[1024];
float decode_buffer_2[1024];
float decode_buffer_3[1024];
float decode_buffer_4[1024];
} COOKContext;
/* debug functions */
#ifdef COOKDEBUG
static void dump_float_table(float* table, int size, int delimiter) {
int i=0;
av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i);
for (i=0 ; i<size ; i++) {
av_log(NULL, AV_LOG_ERROR, "%5.1f, ", table[i]);
if ((i+1)%delimiter == 0) av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i+1);
}
}
static void dump_int_table(int* table, int size, int delimiter) {
int i=0;
av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i);
for (i=0 ; i<size ; i++) {
av_log(NULL, AV_LOG_ERROR, "%d, ", table[i]);
if ((i+1)%delimiter == 0) av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i+1);
}
}
static void dump_short_table(short* table, int size, int delimiter) {
int i=0;
av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i);
for (i=0 ; i<size ; i++) {
av_log(NULL, AV_LOG_ERROR, "%d, ", table[i]);
if ((i+1)%delimiter == 0) av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i+1);
}
}
#endif
/*************** init functions ***************/
/* table generator */
static void init_pow2table(COOKContext *q){
int i;
q->pow2tab[63] = 1.0;
for (i=1 ; i<64 ; i++){
q->pow2tab[63+i]=(float)((uint64_t)1<<i);
q->pow2tab[63-i]=1.0/(float)((uint64_t)1<<i);
}
}
/* table generator */
static void init_rootpow2table(COOKContext *q){
int i;
q->rootpow2tab[63] = 1.0;
for (i=1 ; i<64 ; i++){
q->rootpow2tab[63+i]=sqrt((float)((uint64_t)1<<i));
q->rootpow2tab[63-i]=sqrt(1.0/(float)((uint64_t)1<<i));
}
}
/* table generator */
static void init_gain_table(COOKContext *q) {
int i;
q->gain_size_factor = q->samples_per_channel/8;
for (i=0 ; i<23 ; i++) {
q->gain_table[i] = pow((double)q->pow2tab[i+52] ,
(1.0/(double)q->gain_size_factor));
}
}
static int init_cook_vlc_tables(COOKContext *q) {
int i, result;
result = 0;
for (i=0 ; i<13 ; i++) {
result &= init_vlc (&q->envelope_quant_index[i], 9, 24,
envelope_quant_index_huffbits[i], 1, 1,
envelope_quant_index_huffcodes[i], 2, 2, 0);
}
av_log(NULL,AV_LOG_DEBUG,"sqvh VLC init\n");
for (i=0 ; i<7 ; i++) {
result &= init_vlc (&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i],
cvh_huffbits[i], 1, 1,
cvh_huffcodes[i], 2, 2, 0);
}
if (q->nb_channels==2 && q->joint_stereo==1){
result &= init_vlc (&q->ccpl, 6, (1<<q->js_vlc_bits)-1,
ccpl_huffbits[q->js_vlc_bits-2], 1, 1,
ccpl_huffcodes[q->js_vlc_bits-2], 2, 2, 0);
av_log(NULL,AV_LOG_DEBUG,"Joint-stereo VLC used.\n");
}
av_log(NULL,AV_LOG_DEBUG,"VLC tables initialized.\n");
return result;
}
static int init_cook_mlt(COOKContext *q) {
int j;
float alpha;
/* Allocate the buffers, could be replaced with a static [512]
array if needed. */
q->mlt_size = q->samples_per_channel;
q->mlt_window = av_malloc(sizeof(float)*q->mlt_size);
q->mlt_precos = av_malloc(sizeof(float)*q->mlt_size/2);
q->mlt_presin = av_malloc(sizeof(float)*q->mlt_size/2);
q->mlt_postcos = av_malloc(sizeof(float)*q->mlt_size/2);
/* Initialize the MLT window: simple sine window. */
alpha = M_PI / (2.0 * (float)q->mlt_size);
for(j=0 ; j<q->mlt_size ; j++) {
q->mlt_window[j] = sin((j + 512.0/(float)q->mlt_size) * alpha);
}
/* pre/post twiddle factors */
for (j=0 ; j<q->mlt_size/2 ; j++){
q->mlt_precos[j] = cos( ((j+0.25)*M_PI)/q->mlt_size);
q->mlt_presin[j] = sin( ((j+0.25)*M_PI)/q->mlt_size);
q->mlt_postcos[j] = (float)sqrt(2.0/(float)q->mlt_size)*cos( ((float)j*M_PI) /q->mlt_size); //sqrt(2/MLT_size) = scalefactor
}
/* Initialize the FFT. */
ff_fft_init(&q->fft_ctx, av_log2(q->mlt_size)-1, 0);
av_log(NULL,AV_LOG_DEBUG,"FFT initialized, order = %d.\n",
av_log2(q->samples_per_channel)-1);
return (int)(q->mlt_window && q->mlt_precos && q->mlt_presin && q->mlt_postcos);
}
/*************** init functions end ***********/
/**
* Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
* Why? No idea, some checksum/error detection method maybe.
* Nice way to waste CPU cycles.
*
* @param in pointer to 32bit array of indata
* @param bits amount of bits
* @param out pointer to 32bit array of outdata
*/
static inline void decode_bytes(uint8_t* inbuffer, uint8_t* out, int bytes){
int i;
uint32_t* buf = (uint32_t*) inbuffer;
uint32_t* obuf = (uint32_t*) out;
/* FIXME: 64 bit platforms would be able to do 64 bits at a time.
* I'm too lazy though, should be something like
* for(i=0 ; i<bitamount/64 ; i++)
* (int64_t)out[i] = 0x37c511f237c511f2^be2me_64(int64_t)in[i]);
* Buffer alignment needs to be checked. */
for(i=0 ; i<bytes/4 ; i++){
#ifdef WORDS_BIGENDIAN
obuf[i] = 0x37c511f2^buf[i];
#else
obuf[i] = 0xf211c537^buf[i];
#endif
}
}
/**
* Cook uninit
*/
static int cook_decode_close(AVCodecContext *avctx)
{
int i;
COOKContext *q = avctx->priv_data;
av_log(NULL,AV_LOG_DEBUG, "Deallocating memory.\n");
/* Free allocated memory buffers. */
av_free(q->mlt_window);
av_free(q->mlt_precos);
av_free(q->mlt_presin);
av_free(q->mlt_postcos);
av_free(q->decoded_bytes_buffer);
/* Free the transform. */
ff_fft_end(&q->fft_ctx);
/* Free the VLC tables. */
for (i=0 ; i<13 ; i++) {
free_vlc(&q->envelope_quant_index[i]);
}
for (i=0 ; i<7 ; i++) {
free_vlc(&q->sqvh[i]);
}
if(q->nb_channels==2 && q->joint_stereo==1 ){
free_vlc(&q->ccpl);
}
av_log(NULL,AV_LOG_DEBUG,"Memory deallocated.\n");
return 0;
}
/**
* Fill the COOKgain structure for the timedomain quantization.
*
* @param q pointer to the COOKContext
* @param gaininfo pointer to the COOKgain
*/
static void decode_gain_info(GetBitContext *gb, COOKgain* gaininfo) {
int i;
while (get_bits1(gb)) {}
gaininfo->size = get_bits_count(gb) - 1; //amount of elements*2 to update
if (get_bits_count(gb) - 1 <= 0) return;
for (i=0 ; i<gaininfo->size ; i++){
gaininfo->qidx_table1[i] = get_bits(gb,3);
if (get_bits1(gb)) {
gaininfo->qidx_table2[i] = get_bits(gb,4) - 7; //convert to signed
} else {
gaininfo->qidx_table2[i] = -1;
}
}
}
/**
* Create the quant index table needed for the envelope.
*
* @param q pointer to the COOKContext
* @param quant_index_table pointer to the array
*/
static void decode_envelope(COOKContext *q, int* quant_index_table) {
int i,j, vlc_index;
int bitbias;
bitbias = get_bits_count(&q->gb);
quant_index_table[0]= get_bits(&q->gb,6) - 6; //This is used later in categorize
for (i=1 ; i < q->total_subbands ; i++){
vlc_index=i;
if (i >= q->js_subband_start * 2) {
vlc_index-=q->js_subband_start;
} else {
vlc_index/=2;
if(vlc_index < 1) vlc_index = 1;
}
if (vlc_index>13) vlc_index = 13; //the VLC tables >13 are identical to No. 13
j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index-1].table,
q->envelope_quant_index[vlc_index-1].bits,2);
quant_index_table[i] = quant_index_table[i-1] + j - 12; //differential encoding
}
}
/**
* Create the quant value table.
*
* @param q pointer to the COOKContext
* @param quant_value_table pointer to the array
*/
static void inline dequant_envelope(COOKContext *q, int* quant_index_table,
float* quant_value_table){
int i;
for(i=0 ; i < q->total_subbands ; i++){
quant_value_table[i] = q->rootpow2tab[quant_index_table[i]+63];
}
}
/**
* Calculate the category and category_index vector.
*
* @param q pointer to the COOKContext
* @param quant_index_table pointer to the array
* @param category pointer to the category array
* @param category_index pointer to the category_index array
*/
static void categorize(COOKContext *q, int* quant_index_table,
int* category, int* category_index){
int exp_idx, bias, tmpbias, bits_left, num_bits, index, v, i, j;
int exp_index2[102];
int exp_index1[102];
int tmp_categorize_array1[128];
int tmp_categorize_array1_idx=0;
int tmp_categorize_array2[128];
int tmp_categorize_array2_idx=0;
int category_index_size=0;
bits_left = q->bits_per_subpacket - get_bits_count(&q->gb);
if(bits_left > q->samples_per_channel) {
bits_left = q->samples_per_channel +
((bits_left - q->samples_per_channel)*5)/8;
//av_log(NULL, AV_LOG_ERROR, "bits_left = %d\n",bits_left);
}
memset(&exp_index1,0,102*sizeof(int));
memset(&exp_index2,0,102*sizeof(int));
memset(&tmp_categorize_array1,0,128*sizeof(int));
memset(&tmp_categorize_array2,0,128*sizeof(int));
bias=-32;
/* Estimate bias. */
for (i=32 ; i>0 ; i=i/2){
num_bits = 0;
index = 0;
for (j=q->total_subbands ; j>0 ; j--){
exp_idx = (i - quant_index_table[index] + bias) / 2;
if (exp_idx<0){
exp_idx=0;
} else if(exp_idx >7) {
exp_idx=7;
}
index++;
num_bits+=expbits_tab[exp_idx];
}
if(num_bits >= bits_left - 32){
bias+=i;
}
}
/* Calculate total number of bits. */
num_bits=0;
for (i=0 ; i<q->total_subbands ; i++) {
exp_idx = (bias - quant_index_table[i]) / 2;
if (exp_idx<0) {
exp_idx=0;
} else if(exp_idx >7) {
exp_idx=7;
}
num_bits += expbits_tab[exp_idx];
exp_index1[i] = exp_idx;
exp_index2[i] = exp_idx;
}
tmpbias = bias = num_bits;
for (j = 1 ; j < q->numvector_size ; j++) {
if (tmpbias + bias > 2*bits_left) { /* ---> */
int max = -999999;
index=-1;
for (i=0 ; i<q->total_subbands ; i++){
if (exp_index1[i] < 7) {
v = (-2*exp_index1[i]) - quant_index_table[i] - 32;
if ( v >= max) {
max = v;
index = i;
}
}
}
if(index==-1)break;
tmp_categorize_array1[tmp_categorize_array1_idx++] = index;
tmpbias -= expbits_tab[exp_index1[index]] -
expbits_tab[exp_index1[index]+1];
++exp_index1[index];
} else { /* <--- */
int min = 999999;
index=-1;
for (i=0 ; i<q->total_subbands ; i++){
if(exp_index2[i] > 0){
v = (-2*exp_index2[i])-quant_index_table[i];
if ( v < min) {
min = v;
index = i;
}
}
}
if(index == -1)break;
tmp_categorize_array2[tmp_categorize_array2_idx++] = index;
tmpbias -= expbits_tab[exp_index2[index]] -
expbits_tab[exp_index2[index]-1];
--exp_index2[index];
}
}
for(i=0 ; i<q->total_subbands ; i++)
category[i] = exp_index2[i];
/* Concatenate the two arrays. */
for(i=tmp_categorize_array2_idx-1 ; i >= 0; i--)
category_index[category_index_size++] = tmp_categorize_array2[i];
for(i=0;i<tmp_categorize_array1_idx;i++)
category_index[category_index_size++ ] = tmp_categorize_array1[i];
/* FIXME: mc_sich_ra8_20.rm triggers this, not sure with what we
should fill the remaining bytes. */
for(i=category_index_size;i<q->numvector_size;i++)
category_index[i]=0;
}
/**
* Expand the category vector.
*
* @param q pointer to the COOKContext
* @param category pointer to the category array
* @param category_index pointer to the category_index array
*/
static void inline expand_category(COOKContext *q, int* category,
int* category_index){
int i;
for(i=0 ; i<q->num_vectors ; i++){
++category[category_index[i]];
}
}
/**
* The real requantization of the mltcoefs
*
* @param q pointer to the COOKContext
* @param index index
* @param band current subband
* @param quant_value_table pointer to the array
* @param subband_coef_index array of indexes to quant_centroid_tab
* @param subband_coef_noise use random noise instead of predetermined value
* @param mlt_buffer pointer to the mlt buffer
*/
static void scalar_dequant(COOKContext *q, int index, int band,
float* quant_value_table, int* subband_coef_index,
int* subband_coef_noise, float* mlt_buffer){
int i;
float f1;
for(i=0 ; i<SUBBAND_SIZE ; i++) {
if (subband_coef_index[i]) {
if (subband_coef_noise[i]) {
f1 = -quant_centroid_tab[index][subband_coef_index[i]];
} else {
f1 = quant_centroid_tab[index][subband_coef_index[i]];
}
} else {
/* noise coding if subband_coef_noise[i] == 0 */
q->random_state = q->random_state * 214013 + 2531011; //typical RNG numbers
f1 = randsign[(q->random_state/0x1000000)&1] * dither_tab[index]; //>>31
}
mlt_buffer[band*20+ i] = f1 * quant_value_table[band];
}
}
/**
* Unpack the subband_coef_index and subband_coef_noise vectors.
*
* @param q pointer to the COOKContext
* @param category pointer to the category array
* @param subband_coef_index array of indexes to quant_centroid_tab
* @param subband_coef_noise use random noise instead of predetermined value
*/
static int unpack_SQVH(COOKContext *q, int category, int* subband_coef_index,
int* subband_coef_noise) {
int i,j;
int vlc, vd ,tmp, result;
int ub;
int cb;
vd = vd_tab[category];
result = 0;
for(i=0 ; i<vpr_tab[category] ; i++){
ub = get_bits_count(&q->gb);
vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);
cb = get_bits_count(&q->gb);
if (q->bits_per_subpacket < get_bits_count(&q->gb)){
vlc = 0;
result = 1;
}
for(j=vd-1 ; j>=0 ; j--){
tmp = (vlc * invradix_tab[category])/0x100000;
subband_coef_index[vd*i+j] = vlc - tmp * (kmax_tab[category]+1);
vlc = tmp;
}
for(j=0 ; j<vd ; j++){
if (subband_coef_index[i*vd + j]) {
if(get_bits_count(&q->gb) < q->bits_per_subpacket){
subband_coef_noise[i*vd+j] = get_bits1(&q->gb);
} else {
result=1;
subband_coef_noise[i*vd+j]=0;
}
} else {
subband_coef_noise[i*vd+j]=0;
}
}
}
return result;
}
/**
* Fill the mlt_buffer with mlt coefficients.
*
* @param q pointer to the COOKContext
* @param category pointer to the category array
* @param quant_value_table pointer to the array
* @param mlt_buffer pointer to mlt coefficients
*/
static void decode_vectors(COOKContext* q, int* category,
float* quant_value_table, float* mlt_buffer){
/* A zero in this table means that the subband coefficient is
random noise coded. */
int subband_coef_noise[SUBBAND_SIZE];
/* A zero in this table means that the subband coefficient is a
positive multiplicator. */
int subband_coef_index[SUBBAND_SIZE];
int band, j;
int index=0;
for(band=0 ; band<q->total_subbands ; band++){
index = category[band];
if(category[band] < 7){
if(unpack_SQVH(q, category[band], subband_coef_index, subband_coef_noise)){
index=7;
for(j=0 ; j<q->total_subbands ; j++) category[band+j]=7;
}
}
if(index==7) {
memset(subband_coef_index, 0, sizeof(subband_coef_index));
memset(subband_coef_noise, 0, sizeof(subband_coef_noise));
}
scalar_dequant(q, index, band, quant_value_table, subband_coef_index,
subband_coef_noise, mlt_buffer);
}
if(q->total_subbands*SUBBAND_SIZE >= q->samples_per_channel){
return;
}
}
/**
* function for decoding mono data
*
* @param q pointer to the COOKContext
* @param mlt_buffer1 pointer to left channel mlt coefficients
* @param mlt_buffer2 pointer to right channel mlt coefficients
*/
static void mono_decode(COOKContext *q, float* mlt_buffer) {
int category_index[128];
float quant_value_table[102];
int quant_index_table[102];
int category[128];
memset(&category, 0, 128*sizeof(int));
memset(&quant_value_table, 0, 102*sizeof(int));
memset(&category_index, 0, 128*sizeof(int));
decode_envelope(q, quant_index_table);
q->num_vectors = get_bits(&q->gb,q->log2_numvector_size);
dequant_envelope(q, quant_index_table, quant_value_table);
categorize(q, quant_index_table, category, category_index);
expand_category(q, category, category_index);
decode_vectors(q, category, quant_value_table, mlt_buffer);
}
/**
* The modulated lapped transform, this takes transform coefficients
* and transforms them into timedomain samples. This is done through
* an FFT-based algorithm with pre- and postrotation steps.
* A window and reorder step is also included.
*
* @param q pointer to the COOKContext
* @param inbuffer pointer to the mltcoefficients
* @param outbuffer pointer to the timedomain buffer
* @param mlt_tmp pointer to temporary storage space
*/
static void cook_imlt(COOKContext *q, float* inbuffer, float* outbuffer,
float* mlt_tmp){
int i;
/* prerotation */
for(i=0 ; i<q->mlt_size ; i+=2){
outbuffer[i] = (q->mlt_presin[i/2] * inbuffer[q->mlt_size-1-i]) +
(q->mlt_precos[i/2] * inbuffer[i]);
outbuffer[i+1] = (q->mlt_precos[i/2] * inbuffer[q->mlt_size-1-i]) -
(q->mlt_presin[i/2] * inbuffer[i]);
}
/* FFT */
ff_fft_permute(&q->fft_ctx, (FFTComplex *) outbuffer);
ff_fft_calc (&q->fft_ctx, (FFTComplex *) outbuffer);
/* postrotation */
for(i=0 ; i<q->mlt_size ; i+=2){
mlt_tmp[i] = (q->mlt_postcos[(q->mlt_size-1-i)/2] * outbuffer[i+1]) +
(q->mlt_postcos[i/2] * outbuffer[i]);
mlt_tmp[q->mlt_size-1-i] = (q->mlt_postcos[(q->mlt_size-1-i)/2] * outbuffer[i]) -
(q->mlt_postcos[i/2] * outbuffer[i+1]);
}
/* window and reorder */
for(i=0 ; i<q->mlt_size/2 ; i++){
outbuffer[i] = mlt_tmp[q->mlt_size/2-1-i] * q->mlt_window[i];
outbuffer[q->mlt_size-1-i]= mlt_tmp[q->mlt_size/2-1-i] *
q->mlt_window[q->mlt_size-1-i];
outbuffer[q->mlt_size+i]= mlt_tmp[q->mlt_size/2+i] *
q->mlt_window[q->mlt_size-1-i];
outbuffer[2*q->mlt_size-1-i]= -(mlt_tmp[q->mlt_size/2+i] *
q->mlt_window[i]);
}
}
/**
* the actual requantization of the timedomain samples
*
* @param q pointer to the COOKContext
* @param buffer pointer to the timedomain buffer
* @param gain_index index for the block multiplier
* @param gain_index_next index for the next block multiplier
*/
static void interpolate(COOKContext *q, float* buffer,
int gain_index, int gain_index_next){
int i;
float fc1, fc2;
fc1 = q->pow2tab[gain_index+63];
if(gain_index == gain_index_next){ //static gain
for(i=0 ; i<q->gain_size_factor ; i++){
buffer[i]*=fc1;
}
return;
} else { //smooth gain
fc2 = q->gain_table[11 + (gain_index_next-gain_index)];
for(i=0 ; i<q->gain_size_factor ; i++){
buffer[i]*=fc1;
fc1*=fc2;
}
return;
}
}
/**
* timedomain requantization of the timedomain samples
*
* @param q pointer to the COOKContext
* @param buffer pointer to the timedomain buffer
* @param gain_now current gain structure
* @param gain_previous previous gain structure
*/
static void gain_window(COOKContext *q, float* buffer, COOKgain* gain_now,
COOKgain* gain_previous){
int i, index;
int gain_index[9];
int tmp_gain_index;
gain_index[8]=0;
index = gain_previous->size;
for (i=7 ; i>=0 ; i--) {
if(index && gain_previous->qidx_table1[index-1]==i) {
gain_index[i] = gain_previous->qidx_table2[index-1];
index--;
} else {
gain_index[i]=gain_index[i+1];
}
}
/* This is applied to the to be previous data buffer. */
for(i=0;i<8;i++){
interpolate(q, &buffer[q->samples_per_channel+q->gain_size_factor*i],
gain_index[i], gain_index[i+1]);
}
tmp_gain_index = gain_index[0];
index = gain_now->size;
for (i=7 ; i>=0 ; i--) {
if(index && gain_now->qidx_table1[index-1]==i) {
gain_index[i]= gain_now->qidx_table2[index-1];
index--;
} else {
gain_index[i]=gain_index[i+1];
}
}
/* This is applied to the to be current block. */
for(i=0;i<8;i++){
interpolate(q, &buffer[i*q->gain_size_factor],
tmp_gain_index+gain_index[i],
tmp_gain_index+gain_index[i+1]);
}
}
/**
* mlt overlapping and buffer management
*
* @param q pointer to the COOKContext
* @param buffer pointer to the timedomain buffer
* @param gain_now current gain structure
* @param gain_previous previous gain structure
* @param previous_buffer pointer to the previous buffer to be used for overlapping
*
*/
static void gain_compensate(COOKContext *q, float* buffer, COOKgain* gain_now,
COOKgain* gain_previous, float* previous_buffer) {
int i;
if((gain_now->size || gain_previous->size)) {
gain_window(q, buffer, gain_now, gain_previous);
}
/* Overlap with the previous block. */
for(i=0 ; i<q->samples_per_channel ; i++) buffer[i]+=previous_buffer[i];
/* Save away the current to be previous block. */
memcpy(previous_buffer, buffer+q->samples_per_channel,
sizeof(float)*q->samples_per_channel);
}
/**
* function for getting the jointstereo coupling information
*
* @param q pointer to the COOKContext
* @param decouple_tab decoupling array
*
*/
static void decouple_info(COOKContext *q, int* decouple_tab){
int length, i;
if(get_bits1(&q->gb)) {
if(cplband[q->js_subband_start] > cplband[q->subbands-1]) return;
length = cplband[q->subbands-1] - cplband[q->js_subband_start] + 1;
for (i=0 ; i<length ; i++) {
decouple_tab[cplband[q->js_subband_start] + i] = get_vlc2(&q->gb, q->ccpl.table, q->ccpl.bits, 2);
}
return;
}
if(cplband[q->js_subband_start] > cplband[q->subbands-1]) return;
length = cplband[q->subbands-1] - cplband[q->js_subband_start] + 1;
for (i=0 ; i<length ; i++) {
decouple_tab[cplband[q->js_subband_start] + i] = get_bits(&q->gb, q->js_vlc_bits);
}
return;
}
/**
* function for decoding joint stereo data
*
* @param q pointer to the COOKContext
* @param mlt_buffer1 pointer to left channel mlt coefficients
* @param mlt_buffer2 pointer to right channel mlt coefficients
*/
static void joint_decode(COOKContext *q, float* mlt_buffer1,
float* mlt_buffer2) {
int i,j;
int decouple_tab[SUBBAND_SIZE];
float decode_buffer[1060];
int idx, cpl_tmp,tmp_idx;
float f1,f2;
float* cplscale;
memset(decouple_tab, 0, sizeof(decouple_tab));
memset(decode_buffer, 0, sizeof(decode_buffer));
/* Make sure the buffers are zeroed out. */
memset(mlt_buffer1,0, 1024*sizeof(float));
memset(mlt_buffer2,0, 1024*sizeof(float));
decouple_info(q, decouple_tab);
mono_decode(q, decode_buffer);
/* The two channels are stored interleaved in decode_buffer. */
for (i=0 ; i<q->js_subband_start ; i++) {
for (j=0 ; j<SUBBAND_SIZE ; j++) {
mlt_buffer1[i*20+j] = decode_buffer[i*40+j];
mlt_buffer2[i*20+j] = decode_buffer[i*40+20+j];
}
}
/* When we reach js_subband_start (the higher frequencies)
the coefficients are stored in a coupling scheme. */
idx = (1 << q->js_vlc_bits) - 1;
for (i=q->js_subband_start ; i<q->subbands ; i++) {
cpl_tmp = cplband[i];
idx -=decouple_tab[cpl_tmp];
cplscale = (float*)cplscales[q->js_vlc_bits-2]; //choose decoupler table
f1 = cplscale[decouple_tab[cpl_tmp]];
f2 = cplscale[idx-1];
for (j=0 ; j<SUBBAND_SIZE ; j++) {
tmp_idx = ((q->js_subband_start + i)*20)+j;
mlt_buffer1[20*i + j] = f1 * decode_buffer[tmp_idx];
mlt_buffer2[20*i + j] = f2 * decode_buffer[tmp_idx];
}
idx = (1 << q->js_vlc_bits) - 1;
}
}
/**
* Cook subpacket decoding. This function returns one decoded subpacket,
* usually 1024 samples per channel.
*
* @param q pointer to the COOKContext
* @param inbuffer pointer to the inbuffer
* @param sub_packet_size subpacket size
* @param outbuffer pointer to the outbuffer
*/
static int decode_subpacket(COOKContext *q, uint8_t *inbuffer,
int sub_packet_size, int16_t *outbuffer) {
int i,j;
int value;
float* tmp_ptr;
/* packet dump */
// for (i=0 ; i<sub_packet_size ; i++) {
// av_log(NULL, AV_LOG_ERROR, "%02x", inbuffer[i]);
// }
// av_log(NULL, AV_LOG_ERROR, "\n");
decode_bytes(inbuffer, q->decoded_bytes_buffer, sub_packet_size);
init_get_bits(&q->gb, q->decoded_bytes_buffer, sub_packet_size*8);
decode_gain_info(&q->gb, &q->gain_current);
if(q->nb_channels==2 && q->joint_stereo==1){
joint_decode(q, q->decode_buf_ptr[0], q->decode_buf_ptr[2]);
/* Swap buffer pointers. */
tmp_ptr = q->decode_buf_ptr[1];
q->decode_buf_ptr[1] = q->decode_buf_ptr[0];
q->decode_buf_ptr[0] = tmp_ptr;
tmp_ptr = q->decode_buf_ptr[3];
q->decode_buf_ptr[3] = q->decode_buf_ptr[2];
q->decode_buf_ptr[2] = tmp_ptr;
/* FIXME: Rethink the gainbuffer handling, maybe a rename?
now/previous swap */
q->gain_now_ptr = &q->gain_now;
q->gain_previous_ptr = &q->gain_previous;
for (i=0 ; i<q->nb_channels ; i++){
cook_imlt(q, q->decode_buf_ptr[i*2], q->mono_mdct_output, q->mlt_tmp);
gain_compensate(q, q->mono_mdct_output, q->gain_now_ptr,
q->gain_previous_ptr, q->previous_buffer_ptr[0]);
/* Swap out the previous buffer. */
tmp_ptr = q->previous_buffer_ptr[0];
q->previous_buffer_ptr[0] = q->previous_buffer_ptr[1];
q->previous_buffer_ptr[1] = tmp_ptr;
/* Clip and convert the floats to 16 bits. */
for (j=0 ; j<q->samples_per_frame ; j++){
value = lrintf(q->mono_mdct_output[j]);
if(value < -32768) value = -32768;
else if(value > 32767) value = 32767;
outbuffer[2*j+i] = value;
}
}
memcpy(&q->gain_now, &q->gain_previous, sizeof(COOKgain));
memcpy(&q->gain_previous, &q->gain_current, sizeof(COOKgain));
} else if (q->nb_channels==2 && q->joint_stereo==0) {
/* channel 0 */
mono_decode(q, q->decode_buf_ptr2[0]);
tmp_ptr = q->decode_buf_ptr2[0];
q->decode_buf_ptr2[0] = q->decode_buf_ptr2[1];
q->decode_buf_ptr2[1] = tmp_ptr;
memcpy(&q->gain_channel1[0], &q->gain_current ,sizeof(COOKgain));
q->gain_now_ptr = &q->gain_channel1[0];
q->gain_previous_ptr = &q->gain_channel1[1];
cook_imlt(q, q->decode_buf_ptr2[0], q->mono_mdct_output,q->mlt_tmp);
gain_compensate(q, q->mono_mdct_output, q->gain_now_ptr,
q->gain_previous_ptr, q->mono_previous_buffer1);
memcpy(&q->gain_channel1[1], &q->gain_channel1[0],sizeof(COOKgain));
for (j=0 ; j<q->samples_per_frame ; j++){
value = lrintf(q->mono_mdct_output[j]);
if(value < -32768) value = -32768;
else if(value > 32767) value = 32767;
outbuffer[2*j+1] = value;
}
/* channel 1 */
//av_log(NULL,AV_LOG_ERROR,"bits = %d\n",get_bits_count(&q->gb));
init_get_bits(&q->gb, q->decoded_bytes_buffer, sub_packet_size*8+q->bits_per_subpacket);
q->gain_now_ptr = &q->gain_channel2[0];
q->gain_previous_ptr = &q->gain_channel2[1];
decode_gain_info(&q->gb, &q->gain_channel2[0]);
mono_decode(q, q->decode_buf_ptr[0]);
tmp_ptr = q->decode_buf_ptr[0];
q->decode_buf_ptr[0] = q->decode_buf_ptr[1];
q->decode_buf_ptr[1] = tmp_ptr;
cook_imlt(q, q->decode_buf_ptr[0], q->mono_mdct_output,q->mlt_tmp);
gain_compensate(q, q->mono_mdct_output, q->gain_now_ptr,
q->gain_previous_ptr, q->mono_previous_buffer2);
/* Swap out the previous buffer. */
tmp_ptr = q->previous_buffer_ptr[0];
q->previous_buffer_ptr[0] = q->previous_buffer_ptr[1];
q->previous_buffer_ptr[1] = tmp_ptr;
memcpy(&q->gain_channel2[1], &q->gain_channel2[0] ,sizeof(COOKgain));
for (j=0 ; j<q->samples_per_frame ; j++){
value = lrintf(q->mono_mdct_output[j]);
if(value < -32768) value = -32768;
else if(value > 32767) value = 32767;
outbuffer[2*j] = value;
}
} else {
mono_decode(q, q->decode_buf_ptr[0]);
/* Swap buffer pointers. */
tmp_ptr = q->decode_buf_ptr[1];
q->decode_buf_ptr[1] = q->decode_buf_ptr[0];
q->decode_buf_ptr[0] = tmp_ptr;
/* FIXME: Rethink the gainbuffer handling, maybe a rename?
now/previous swap */
q->gain_now_ptr = &q->gain_now;
q->gain_previous_ptr = &q->gain_previous;
cook_imlt(q, q->decode_buf_ptr[0], q->mono_mdct_output,q->mlt_tmp);
gain_compensate(q, q->mono_mdct_output, q->gain_now_ptr,
q->gain_previous_ptr, q->mono_previous_buffer1);
/* Clip and convert the floats to 16 bits */
for (j=0 ; j<q->samples_per_frame ; j++){
value = lrintf(q->mono_mdct_output[j]);
if(value < -32768) value = -32768;
else if(value > 32767) value = 32767;
outbuffer[j] = value;
}
memcpy(&q->gain_now, &q->gain_previous, sizeof(COOKgain));
memcpy(&q->gain_previous, &q->gain_current, sizeof(COOKgain));
}
return q->samples_per_frame * sizeof(int16_t);
}
/**
* Cook frame decoding
*
* @param avctx pointer to the AVCodecContext
*/
static int cook_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
uint8_t *buf, int buf_size) {
COOKContext *q = avctx->priv_data;
if (buf_size < avctx->block_align)
return buf_size;
*data_size = decode_subpacket(q, buf, avctx->block_align, data);
return avctx->block_align;
}
#ifdef COOKDEBUG
static void dump_cook_context(COOKContext *q, COOKextradata *e)
{
//int i=0;
#define PRINT(a,b) av_log(NULL,AV_LOG_ERROR," %s = %d\n", a, b);
av_log(NULL,AV_LOG_ERROR,"COOKextradata\n");
av_log(NULL,AV_LOG_ERROR,"cookversion=%x\n",e->cookversion);
if (e->cookversion > MONO_COOK2) {
PRINT("js_subband_start",e->js_subband_start);
PRINT("js_vlc_bits",e->js_vlc_bits);
}
av_log(NULL,AV_LOG_ERROR,"COOKContext\n");
PRINT("nb_channels",q->nb_channels);
PRINT("bit_rate",q->bit_rate);
PRINT("sample_rate",q->sample_rate);
PRINT("samples_per_channel",q->samples_per_channel);
PRINT("samples_per_frame",q->samples_per_frame);
PRINT("subbands",q->subbands);
PRINT("random_state",q->random_state);
PRINT("mlt_size",q->mlt_size);
PRINT("js_subband_start",q->js_subband_start);
PRINT("log2_numvector_size",q->log2_numvector_size);
PRINT("numvector_size",q->numvector_size);
PRINT("total_subbands",q->total_subbands);
}
#endif
/**
* Cook initialization
*
* @param avctx pointer to the AVCodecContext
*/
static int cook_decode_init(AVCodecContext *avctx)
{
COOKextradata *e = avctx->extradata;
COOKContext *q = avctx->priv_data;
/* Take care of the codec specific extradata. */
if (avctx->extradata_size <= 0) {
av_log(NULL,AV_LOG_ERROR,"Necessary extradata missing!\n");
return -1;
} else {
/* 8 for mono, 16 for stereo, ? for multichannel
Swap to right endianness so we don't need to care later on. */
av_log(NULL,AV_LOG_DEBUG,"codecdata_length=%d\n",avctx->extradata_size);
if (avctx->extradata_size >= 8){
e->cookversion = be2me_32(e->cookversion);
e->samples_per_frame = be2me_16(e->samples_per_frame);
e->subbands = be2me_16(e->subbands);
}
if (avctx->extradata_size >= 16){
e->js_subband_start = be2me_16(e->js_subband_start);
e->js_vlc_bits = be2me_16(e->js_vlc_bits);
}
}
/* Take data from the AVCodecContext (RM container). */
q->sample_rate = avctx->sample_rate;
q->nb_channels = avctx->channels;
q->bit_rate = avctx->bit_rate;
/* Initialize state. */
q->random_state = 1;
/* Initialize extradata related variables. */
q->samples_per_channel = e->samples_per_frame / q->nb_channels;
q->samples_per_frame = e->samples_per_frame;
q->subbands = e->subbands;
q->bits_per_subpacket = avctx->block_align * 8;
/* Initialize default data states. */
q->js_subband_start = 0;
q->log2_numvector_size = 5;
q->total_subbands = q->subbands;
/* Initialize version-dependent variables */
av_log(NULL,AV_LOG_DEBUG,"e->cookversion=%x\n",e->cookversion);
switch (e->cookversion) {
case MONO_COOK1:
if (q->nb_channels != 1) {
av_log(NULL,AV_LOG_ERROR,"Container channels != 1, report sample!\n");
return -1;
}
av_log(NULL,AV_LOG_DEBUG,"MONO_COOK1\n");
break;
case MONO_COOK2:
if (q->nb_channels != 1) {
q->joint_stereo = 0;
q->bits_per_subpacket = q->bits_per_subpacket/2;
}
av_log(NULL,AV_LOG_DEBUG,"MONO_COOK2\n");
break;
case JOINT_STEREO:
if (q->nb_channels != 2) {
av_log(NULL,AV_LOG_ERROR,"Container channels != 2, report sample!\n");
return -1;
}
av_log(NULL,AV_LOG_DEBUG,"JOINT_STEREO\n");
if (avctx->extradata_size >= 16){
q->total_subbands = q->subbands + e->js_subband_start;
q->js_subband_start = e->js_subband_start;
q->joint_stereo = 1;
q->js_vlc_bits = e->js_vlc_bits;
}
if (q->samples_per_channel > 256) {
q->log2_numvector_size = 6;
}
if (q->samples_per_channel > 512) {
q->log2_numvector_size = 7;
}
break;
case MC_COOK:
av_log(NULL,AV_LOG_ERROR,"MC_COOK not supported!\n");
return -1;
break;
default:
av_log(NULL,AV_LOG_ERROR,"Unknown Cook version, report sample!\n");
return -1;
break;
}
/* Initialize variable relations */
q->mlt_size = q->samples_per_channel;
q->numvector_size = (1 << q->log2_numvector_size);
/* Generate tables */
init_rootpow2table(q);
init_pow2table(q);
init_gain_table(q);
if (init_cook_vlc_tables(q) != 0)
return -1;
if(avctx->block_align >= UINT_MAX/2)
return -1;
/* Pad the databuffer with FF_INPUT_BUFFER_PADDING_SIZE,
this is for the bitstreamreader. */
if ((q->decoded_bytes_buffer = av_mallocz((avctx->block_align+(4-avctx->block_align%4) + FF_INPUT_BUFFER_PADDING_SIZE)*sizeof(uint8_t))) == NULL)
return -1;
q->decode_buf_ptr[0] = q->decode_buffer_1;
q->decode_buf_ptr[1] = q->decode_buffer_2;
q->decode_buf_ptr[2] = q->decode_buffer_3;
q->decode_buf_ptr[3] = q->decode_buffer_4;
q->decode_buf_ptr2[0] = q->decode_buffer_3;
q->decode_buf_ptr2[1] = q->decode_buffer_4;
q->previous_buffer_ptr[0] = q->mono_previous_buffer1;
q->previous_buffer_ptr[1] = q->mono_previous_buffer2;
/* Initialize transform. */
if ( init_cook_mlt(q) == 0 )
return -1;
/* Try to catch some obviously faulty streams, othervise it might be exploitable */
if (q->total_subbands > 53) {
av_log(NULL,AV_LOG_ERROR,"total_subbands > 53, report sample!\n");
return -1;
}
if (q->subbands > 50) {
av_log(NULL,AV_LOG_ERROR,"subbands > 50, report sample!\n");
return -1;
}
if ((q->samples_per_channel == 256) || (q->samples_per_channel == 512) || (q->samples_per_channel == 1024)) {
} else {
av_log(NULL,AV_LOG_ERROR,"unknown amount of samples_per_channel = %d, report sample!\n",q->samples_per_channel);
return -1;
}
#ifdef COOKDEBUG
dump_cook_context(q,e);
#endif
return 0;
}
AVCodec cook_decoder =
{
.name = "cook",
.type = CODEC_TYPE_AUDIO,
.id = CODEC_ID_COOK,
.priv_data_size = sizeof(COOKContext),
.init = cook_decode_init,
.close = cook_decode_close,
.decode = cook_decode_frame,
};
Reference in New Issue View Git Blame Copy Permalink