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a31978e97c
ffmpeg/libavcodec/cook.c
Ramiro Polla a31978e97c Simplify, patch by Ramiro Polla. 
ramiro ta lisha tod ufsc tod br

Originally committed as revision 8754 to svn://svn.ffmpeg.org/ffmpeg/trunk
2007-04-18 19:47:22 +00:00

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/*
* COOK compatible decoder
* Copyright (c) 2003 Sascha Sommer
* Copyright (c) 2005 Benjamin Larsson
*
* This file is part of FFmpeg.
*
* FFmpeg 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 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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 FFmpeg; 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. Bastardization of the G.722.1 standard.
* 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 "common.h"
#include "bytestream.h"
#include "random.h"
#include "cookdata.h"
/* the different Cook versions */
#define MONO 0x1000001
#define STEREO 0x1000002
#define JOINT_STEREO 0x1000003
#define MC_COOK 0x2000000 //multichannel Cook, not supported
#define SUBBAND_SIZE 20
//#define COOKDEBUG
typedef struct {
int *now;
int *previous;
} cook_gains;
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;
int cookversion;
/* states */
AVRandomState random_state;
/* transform data */
MDCTContext mdct_ctx;
DECLARE_ALIGNED_16(FFTSample, mdct_tmp[1024]); /* temporary storage for imlt */
float* mlt_window;
/* gain buffers */
cook_gains gains1;
cook_gains gains2;
int gain_1[9];
int gain_2[9];
int gain_3[9];
int gain_4[9];
/* 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;
DECLARE_ALIGNED_16(float,mono_mdct_output[2048]);
float mono_previous_buffer1[1024];
float mono_previous_buffer2[1024];
float decode_buffer_1[1024];
float decode_buffer_2[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;
int mlt_size = q->samples_per_channel;
if ((q->mlt_window = av_malloc(sizeof(float)*mlt_size)) == 0)
return -1;
/* Initialize the MLT window: simple sine window. */
alpha = M_PI / (2.0 * (float)mlt_size);
for(j=0 ; j<mlt_size ; j++)
q->mlt_window[j] = sin((j + 0.5) * alpha) * sqrt(2.0 / q->samples_per_channel);
/* Initialize the MDCT. */
if (ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size)+1, 1)) {
av_free(q->mlt_window);
return -1;
}
av_log(NULL,AV_LOG_DEBUG,"MDCT initialized, order = %d.\n",
av_log2(mlt_size)+1);
return 0;
}
/*************** init functions end ***********/
/**
* Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
* Why? No idea, some checksum/error detection method maybe.
*
* Out buffer size: extra bytes are needed to cope with
* padding/missalignment.
* Subpackets passed to the decoder can contain two, consecutive
* half-subpackets, of identical but arbitrary size.
* 1234 1234 1234 1234 extraA extraB
* Case 1: AAAA BBBB 0 0
* Case 2: AAAA ABBB BB-- 3 3
* Case 3: AAAA AABB BBBB 2 2
* Case 4: AAAA AAAB BBBB BB-- 1 5
*
* Nice way to waste CPU cycles.
*
* @param inbuffer pointer to byte array of indata
* @param out pointer to byte array of outdata
* @param bytes number of bytes
*/
#define DECODE_BYTES_PAD1(bytes) (3 - ((bytes)+3) % 4)
#define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
static inline int decode_bytes(uint8_t* inbuffer, uint8_t* out, int bytes){
int i, off;
uint32_t c;
uint32_t* buf;
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. */
off = (int)((long)inbuffer & 3);
buf = (uint32_t*) (inbuffer - off);
c = be2me_32((0x37c511f2 >> (off*8)) | (0x37c511f2 << (32-(off*8))));
bytes += 3 + off;
for (i = 0; i < bytes/4; i++)
obuf[i] = c ^ buf[i];
return off;
}
/**
* Cook uninit
*/
static int cook_decode_close(AVCodecContext *avctx)
{
int i;
COOKContext *q = avctx->priv_data;
av_log(avctx,AV_LOG_DEBUG, "Deallocating memory.\n");
/* Free allocated memory buffers. */
av_free(q->mlt_window);
av_free(q->decoded_bytes_buffer);
/* Free the transform. */
ff_mdct_end(&q->mdct_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 gain array for the timedomain quantization.
*
* @param q pointer to the COOKContext
* @param gaininfo[9] array of gain indices
*/
static void decode_gain_info(GetBitContext *gb, int *gaininfo)
{
int i, n;
while (get_bits1(gb)) {}
n = get_bits_count(gb) - 1; //amount of elements*2 to update
i = 0;
while (n--) {
int index = get_bits(gb, 3);
int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1;
while (i <= index) gaininfo[i++] = gain;
}
while (i <= 8) gaininfo[i++] = 0;
}
/**
* 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;
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
}
}
/**
* 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 = av_clip((i - quant_index_table[index] + bias) / 2, 0, 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 = av_clip((bias - quant_index_table[i]) / 2, 0, 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 quant_index quantisation index
* @param subband_coef_index array of indexes to quant_centroid_tab
* @param subband_coef_sign signs of coefficients
* @param mlt_p pointer into the mlt buffer
*/
static void scalar_dequant(COOKContext *q, int index, int quant_index,
int* subband_coef_index, int* subband_coef_sign,
float* mlt_p){
int i;
float f1;
for(i=0 ; i<SUBBAND_SIZE ; i++) {
if (subband_coef_index[i]) {
f1 = quant_centroid_tab[index][subband_coef_index[i]];
if (subband_coef_sign[i]) f1 = -f1;
} else {
/* noise coding if subband_coef_index[i] == 0 */
f1 = dither_tab[index];
if (av_random(&q->random_state) < 0x80000000) f1 = -f1;
}
mlt_p[i] = f1 * q->rootpow2tab[quant_index+63];
}
}
/**
* Unpack the subband_coef_index and subband_coef_sign 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_sign signs of coefficients
*/
static int unpack_SQVH(COOKContext *q, int category, int* subband_coef_index,
int* subband_coef_sign) {
int i,j;
int vlc, vd ,tmp, result;
vd = vd_tab[category];
result = 0;
for(i=0 ; i<vpr_tab[category] ; i++){
vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);
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_sign[i*vd+j] = get_bits1(&q->gb);
} else {
result=1;
subband_coef_sign[i*vd+j]=0;
}
} else {
subband_coef_sign[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_index_table pointer to the array
* @param mlt_buffer pointer to mlt coefficients
*/
static void decode_vectors(COOKContext* q, int* category,
int *quant_index_table, float* mlt_buffer){
/* A zero in this table means that the subband coefficient is
random noise coded. */
int subband_coef_index[SUBBAND_SIZE];
/* A zero in this table means that the subband coefficient is a
positive multiplicator. */
int subband_coef_sign[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_sign)){
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_sign, 0, sizeof(subband_coef_sign));
}
scalar_dequant(q, index, quant_index_table[band],
subband_coef_index, subband_coef_sign,
&mlt_buffer[band * 20]);
}
if(q->total_subbands*SUBBAND_SIZE >= q->samples_per_channel){
return;
} /* FIXME: should this be removed, or moved into loop above? */
}
/**
* function for decoding mono data
*
* @param q pointer to the COOKContext
* @param mlt_buffer pointer to mlt coefficients
*/
static void mono_decode(COOKContext *q, float* mlt_buffer) {
int category_index[128];
int quant_index_table[102];
int category[128];
memset(&category, 0, 128*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);
categorize(q, quant_index_table, category, category_index);
expand_category(q, category, category_index);
decode_vectors(q, category, quant_index_table, mlt_buffer);
}
/**
* 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;
}
}
/**
* The modulated lapped transform, this takes transform coefficients
* and transforms them into timedomain samples.
* Apply transform window, overlap buffers, apply gain profile
* and buffer management.
*
* @param q pointer to the COOKContext
* @param inbuffer pointer to the mltcoefficients
* @param gains_ptr current and previous gains
* @param previous_buffer pointer to the previous buffer to be used for overlapping
*/
static void imlt_gain(COOKContext *q, float *inbuffer,
cook_gains *gains_ptr, float* previous_buffer)
{
const float fc = q->pow2tab[gains_ptr->previous[0] + 63];
float *buffer0 = q->mono_mdct_output;
float *buffer1 = q->mono_mdct_output + q->samples_per_channel;
int i;
/* Inverse modified discrete cosine transform */
q->mdct_ctx.fft.imdct_calc(&q->mdct_ctx, q->mono_mdct_output,
inbuffer, q->mdct_tmp);
/* The weird thing here, is that the two halves of the time domain
* buffer are swapped. Also, the newest data, that we save away for
* next frame, has the wrong sign. Hence the subtraction below.
* Almost sounds like a complex conjugate/reverse data/FFT effect.
*/
/* Apply window and overlap */
for(i = 0; i < q->samples_per_channel; i++){
buffer1[i] = buffer1[i] * fc * q->mlt_window[i] -
previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
}
/* Apply gain profile */
for (i = 0; i < 8; i++) {
if (gains_ptr->now[i] || gains_ptr->now[i + 1])
interpolate(q, &buffer1[q->gain_size_factor * i],
gains_ptr->now[i], gains_ptr->now[i + 1]);
}
/* Save away the current to be previous block. */
memcpy(previous_buffer, buffer0, 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;
}
}
/**
* First part of subpacket decoding:
* decode raw stream bytes and read gain info.
*
* @param q pointer to the COOKContext
* @param inbuffer pointer to raw stream data
* @param gain_ptr array of current/prev gain pointers
*/
static inline void
decode_bytes_and_gain(COOKContext *q, uint8_t *inbuffer,
cook_gains *gains_ptr)
{
int offset;
offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,
q->bits_per_subpacket/8);
init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,
q->bits_per_subpacket);
decode_gain_info(&q->gb, gains_ptr->now);
/* Swap current and previous gains */
FFSWAP(int *, gains_ptr->now, gains_ptr->previous);
}
/**
* Final part of subpacket decoding:
* Apply modulated lapped transform, gain compensation,
* clip and convert to integer.
*
* @param q pointer to the COOKContext
* @param decode_buffer pointer to the mlt coefficients
* @param gain_ptr array of current/prev gain pointers
* @param previous_buffer pointer to the previous buffer to be used for overlapping
* @param out pointer to the output buffer
* @param chan 0: left or single channel, 1: right channel
*/
static inline void
mlt_compensate_output(COOKContext *q, float *decode_buffer,
cook_gains *gains, float *previous_buffer,
int16_t *out, int chan)
{
float *output = q->mono_mdct_output + q->samples_per_channel;
int j;
imlt_gain(q, decode_buffer, gains, previous_buffer);
/* Clip and convert floats to 16 bits.
*/
for (j = 0; j < q->samples_per_channel; j++) {
out[chan + q->nb_channels * j] =
av_clip(lrintf(output[j]), -32768, 32767);
}
}
/**
* 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) {
/* 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_and_gain(q, inbuffer, &q->gains1);
if (q->joint_stereo) {
joint_decode(q, q->decode_buffer_1, q->decode_buffer_2);
} else {
mono_decode(q, q->decode_buffer_1);
if (q->nb_channels == 2) {
decode_bytes_and_gain(q, inbuffer + sub_packet_size/2, &q->gains2);
mono_decode(q, q->decode_buffer_2);
}
}
mlt_compensate_output(q, q->decode_buffer_1, &q->gains1,
q->mono_previous_buffer1, outbuffer, 0);
if (q->nb_channels == 2) {
if (q->joint_stereo) {
mlt_compensate_output(q, q->decode_buffer_2, &q->gains1,
q->mono_previous_buffer2, outbuffer, 1);
} else {
mlt_compensate_output(q, q->decode_buffer_2, &q->gains2,
q->mono_previous_buffer2, outbuffer, 1);
}
}
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);
/* Discard the first two frames: no valid audio. */
if (avctx->frame_number < 2) *data_size = 0;
return avctx->block_align;
}
#ifdef COOKDEBUG
static void dump_cook_context(COOKContext *q)
{
//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",q->cookversion);
if (q->cookversion > STEREO) {
PRINT("js_subband_start",q->js_subband_start);
PRINT("js_vlc_bits",q->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("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)
{
COOKContext *q = avctx->priv_data;
uint8_t *edata_ptr = avctx->extradata;
/* Take care of the codec specific extradata. */
if (avctx->extradata_size <= 0) {
av_log(avctx,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(avctx,AV_LOG_DEBUG,"codecdata_length=%d\n",avctx->extradata_size);
if (avctx->extradata_size >= 8){
q->cookversion = bytestream_get_be32(&edata_ptr);
q->samples_per_frame = bytestream_get_be16(&edata_ptr);
q->subbands = bytestream_get_be16(&edata_ptr);
}
if (avctx->extradata_size >= 16){
bytestream_get_be32(&edata_ptr); //Unknown unused
q->js_subband_start = bytestream_get_be16(&edata_ptr);
q->js_vlc_bits = bytestream_get_be16(&edata_ptr);
}
}
/* 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 RNG. */
av_init_random(1, &q->random_state);
/* Initialize extradata related variables. */
q->samples_per_channel = q->samples_per_frame / q->nb_channels;
q->bits_per_subpacket = avctx->block_align * 8;
/* Initialize default data states. */
q->log2_numvector_size = 5;
q->total_subbands = q->subbands;
/* Initialize version-dependent variables */
av_log(NULL,AV_LOG_DEBUG,"q->cookversion=%x\n",q->cookversion);
q->joint_stereo = 0;
switch (q->cookversion) {
case MONO:
if (q->nb_channels != 1) {
av_log(avctx,AV_LOG_ERROR,"Container channels != 1, report sample!\n");
return -1;
}
av_log(avctx,AV_LOG_DEBUG,"MONO\n");
break;
case STEREO:
if (q->nb_channels != 1) {
q->bits_per_subpacket = q->bits_per_subpacket/2;
}
av_log(avctx,AV_LOG_DEBUG,"STEREO\n");
break;
case JOINT_STEREO:
if (q->nb_channels != 2) {
av_log(avctx,AV_LOG_ERROR,"Container channels != 2, report sample!\n");
return -1;
}
av_log(avctx,AV_LOG_DEBUG,"JOINT_STEREO\n");
if (avctx->extradata_size >= 16){
q->total_subbands = q->subbands + q->js_subband_start;
q->joint_stereo = 1;
}
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(avctx,AV_LOG_ERROR,"MC_COOK not supported!\n");
return -1;
break;
default:
av_log(avctx,AV_LOG_ERROR,"Unknown Cook version, report sample!\n");
return -1;
break;
}
/* Initialize variable relations */
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:
DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
FF_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
if (q->nb_channels==2 && q->joint_stereo==0) {
q->decoded_bytes_buffer =
av_mallocz(avctx->block_align/2
+ DECODE_BYTES_PAD2(avctx->block_align/2)
+ FF_INPUT_BUFFER_PADDING_SIZE);
} else {
q->decoded_bytes_buffer =
av_mallocz(avctx->block_align
+ DECODE_BYTES_PAD1(avctx->block_align)
+ FF_INPUT_BUFFER_PADDING_SIZE);
}
if (q->decoded_bytes_buffer == NULL)
return -1;
q->gains1.now = q->gain_1;
q->gains1.previous = q->gain_2;
q->gains2.now = q->gain_3;
q->gains2.previous = q->gain_4;
/* 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(avctx,AV_LOG_ERROR,"total_subbands > 53, report sample!\n");
return -1;
}
if (q->subbands > 50) {
av_log(avctx,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(avctx,AV_LOG_ERROR,"unknown amount of samples_per_channel = %d, report sample!\n",q->samples_per_channel);
return -1;
}
if ((q->js_vlc_bits > 6) || (q->js_vlc_bits < 0)) {
av_log(avctx,AV_LOG_ERROR,"q->js_vlc_bits = %d, only >= 0 and <= 6 allowed!\n",q->js_vlc_bits);
return -1;
}
#ifdef COOKDEBUG
dump_cook_context(q);
#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,
};
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