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FFmpeg/libavcodec/cook.c

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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 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 "libavutil/lfg.h"
#include "libavutil/random_seed.h"
#include "avcodec.h"
#include "get_bits.h"
#include "dsputil.h"
#include "bytestream.h"
#include "fft.h"
#include "libavutil/audioconvert.h"
#include "sinewin.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 MAX_SUBPACKETS 5
//#define COOKDEBUG
typedef struct {
int *now;
int *previous;
} cook_gains;
typedef struct {
int ch_idx;
int size;
int num_channels;
int cookversion;
int samples_per_frame;
int subbands;
int js_subband_start;
int js_vlc_bits;
int samples_per_channel;
int log2_numvector_size;
unsigned int channel_mask;
VLC ccpl; ///< channel coupling
int joint_stereo;
int bits_per_subpacket;
int bits_per_subpdiv;
int total_subbands;
int numvector_size; ///< 1 << log2_numvector_size;
float mono_previous_buffer1[1024];
float mono_previous_buffer2[1024];
/** gain buffers */
cook_gains gains1;
cook_gains gains2;
int gain_1[9];
int gain_2[9];
int gain_3[9];
int gain_4[9];
} COOKSubpacket;
typedef struct cook {
/*
* The following 5 functions provide the lowlevel arithmetic on
* the internal audio buffers.
*/
void (* scalar_dequant)(struct cook *q, int index, int quant_index,
int* subband_coef_index, int* subband_coef_sign,
float* mlt_p);
void (* decouple) (struct cook *q,
COOKSubpacket *p,
int subband,
float f1, float f2,
float *decode_buffer,
float *mlt_buffer1, float *mlt_buffer2);
void (* imlt_window) (struct cook *q, float *buffer1,
cook_gains *gains_ptr, float *previous_buffer);
void (* interpolate) (struct cook *q, float* buffer,
int gain_index, int gain_index_next);
void (* saturate_output) (struct cook *q, int chan, int16_t *out);
AVCodecContext* avctx;
GetBitContext gb;
/* stream data */
int nb_channels;
int bit_rate;
int sample_rate;
int num_vectors;
int samples_per_channel;
/* states */
AVLFG random_state;
/* transform data */
FFTContext mdct_ctx;
float* mlt_window;
/* VLC data */
VLC envelope_quant_index[13];
VLC sqvh[7]; //scalar quantization
/* generatable tables and related variables */
int gain_size_factor;
float gain_table[23];
/* data buffers */
uint8_t* decoded_bytes_buffer;
DECLARE_ALIGNED(32, float, mono_mdct_output)[2048];
float decode_buffer_1[1024];
float decode_buffer_2[1024];
float decode_buffer_0[1060]; /* static allocation for joint decode */
const float *cplscales[5];
int num_subpackets;
COOKSubpacket subpacket[MAX_SUBPACKETS];
} COOKContext;
static float pow2tab[127];
static float rootpow2tab[127];
/* 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 av_cold void init_pow2table(void){
int i;
for (i=-63 ; i<64 ; i++){
pow2tab[63+i]= pow(2, i);
rootpow2tab[63+i]=sqrt(pow(2, i));
}
}
/* table generator */
static av_cold 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(pow2tab[i+52] ,
(1.0/(double)q->gain_size_factor));
}
}
static av_cold 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(q->avctx,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);
}
for(i=0;i<q->num_subpackets;i++){
if (q->subpacket[i].joint_stereo==1){
result |= init_vlc (&q->subpacket[i].ccpl, 6, (1<<q->subpacket[i].js_vlc_bits)-1,
ccpl_huffbits[q->subpacket[i].js_vlc_bits-2], 1, 1,
ccpl_huffcodes[q->subpacket[i].js_vlc_bits-2], 2, 2, 0);
av_log(q->avctx,AV_LOG_DEBUG,"subpacket %i Joint-stereo VLC used.\n",i);
}
}
av_log(q->avctx,AV_LOG_DEBUG,"VLC tables initialized.\n");
return result;
}
static av_cold int init_cook_mlt(COOKContext *q) {
int j;
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. */
ff_sine_window_init(q->mlt_window, mlt_size);
for(j=0 ; j<mlt_size ; j++)
q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
/* Initialize the MDCT. */
if (ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size)+1, 1, 1.0)) {
av_free(q->mlt_window);
return -1;
}
av_log(q->avctx,AV_LOG_DEBUG,"MDCT initialized, order = %d.\n",
av_log2(mlt_size)+1);
return 0;
}
static const float *maybe_reformat_buffer32 (COOKContext *q, const float *ptr, int n)
{
if (1)
return ptr;
}
static av_cold void init_cplscales_table (COOKContext *q) {
int i;
for (i=0;i<5;i++)
q->cplscales[i] = maybe_reformat_buffer32 (q, cplscales[i], (1<<(i+2))-1);
}
/*************** init functions end ***********/
#define DECODE_BYTES_PAD1(bytes) (3 - ((bytes)+3) % 4)
#define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
/**
* 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/misalignment.
* 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
*/
static inline int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){
int i, off;
uint32_t c;
const 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^av_be2ne64(int64_t)in[i]);
* Buffer alignment needs to be checked. */
off = (intptr_t)inbuffer & 3;
buf = (const uint32_t*) (inbuffer - off);
c = av_be2ne32((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 av_cold 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]);
}
for (i=0 ; i<q->num_subpackets ; i++) {
free_vlc(&q->subpacket[i].ccpl);
}
av_log(avctx,AV_LOG_DEBUG,"Memory deallocated.\n");
return 0;
}
/**
* Fill the gain array for the timedomain quantization.
*
* @param gb pointer to the GetBitContext
* @param gaininfo[9] array of gain indexes
*/
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, COOKSubpacket *p, 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 < p->total_subbands ; i++){
vlc_index=i;
if (i >= p->js_subband_start * 2) {
vlc_index-=p->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, COOKSubpacket *p, int* quant_index_table,
int* category, int* category_index){
int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;
int exp_index2[102];
int exp_index1[102];
int tmp_categorize_array[128*2];
int tmp_categorize_array1_idx=p->numvector_size;
int tmp_categorize_array2_idx=p->numvector_size;
bits_left = p->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(q->avctx, 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_array,0,128*2*sizeof(int));
bias=-32;
/* Estimate bias. */
for (i=32 ; i>0 ; i=i/2){
num_bits = 0;
index = 0;
for (j=p->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<p->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;
}
tmpbias1 = tmpbias2 = num_bits;
for (j = 1 ; j < p->numvector_size ; j++) {
if (tmpbias1 + tmpbias2 > 2*bits_left) { /* ---> */
int max = -999999;
index=-1;
for (i=0 ; i<p->total_subbands ; i++){
if (exp_index1[i] < 7) {
v = (-2*exp_index1[i]) - quant_index_table[i] + bias;
if ( v >= max) {
max = v;
index = i;
}
}
}
if(index==-1)break;
tmp_categorize_array[tmp_categorize_array1_idx++] = index;
tmpbias1 -= expbits_tab[exp_index1[index]] -
expbits_tab[exp_index1[index]+1];
++exp_index1[index];
} else { /* <--- */
int min = 999999;
index=-1;
for (i=0 ; i<p->total_subbands ; i++){
if(exp_index2[i] > 0){
v = (-2*exp_index2[i])-quant_index_table[i]+bias;
if ( v < min) {
min = v;
index = i;
}
}
}
if(index == -1)break;
tmp_categorize_array[--tmp_categorize_array2_idx] = index;
tmpbias2 -= expbits_tab[exp_index2[index]] -
expbits_tab[exp_index2[index]-1];
--exp_index2[index];
}
}
for(i=0 ; i<p->total_subbands ; i++)
category[i] = exp_index2[i];
for(i=0 ; i<p->numvector_size-1 ; i++)
category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];
}
/**
* 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 inline void 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_float(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_lfg_get(&q->random_state) < 0x80000000) f1 = -f1;
}
mlt_p[i] = f1 * 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, COOKSubpacket *p, 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 (p->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) < p->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, COOKSubpacket* p, 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<p->total_subbands ; band++){
index = category[band];
if(category[band] < 7){
if(unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)){
index=7;
for(j=0 ; j<p->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));
}
q->scalar_dequant(q, index, quant_index_table[band],
subband_coef_index, subband_coef_sign,
&mlt_buffer[band * SUBBAND_SIZE]);
}
if(p->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, COOKSubpacket *p, 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, p, quant_index_table);
q->num_vectors = get_bits(&q->gb,p->log2_numvector_size);
categorize(q, p, quant_index_table, category, category_index);
expand_category(q, category, category_index);
decode_vectors(q, p, 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_float(COOKContext *q, float* buffer,
int gain_index, int gain_index_next){
int i;
float fc1, fc2;
fc1 = 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;
}
}
/**
* Apply transform window, overlap buffers.
*
* @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_window_float (COOKContext *q, float *inbuffer,
cook_gains *gains_ptr, float *previous_buffer)
{
const float fc = pow2tab[gains_ptr->previous[0] + 63];
int i;
/* 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++){
inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] -
previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
}
}
/**
* 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)
{
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.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
q->imlt_window (q, buffer1, gains_ptr, previous_buffer);
/* Apply gain profile */
for (i = 0; i < 8; i++) {
if (gains_ptr->now[i] || gains_ptr->now[i + 1])
q->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, COOKSubpacket *p, int* decouple_tab){
int length, i;
if(get_bits1(&q->gb)) {
if(cplband[p->js_subband_start] > cplband[p->subbands-1]) return;
length = cplband[p->subbands-1] - cplband[p->js_subband_start] + 1;
for (i=0 ; i<length ; i++) {
decouple_tab[cplband[p->js_subband_start] + i] = get_vlc2(&q->gb, p->ccpl.table, p->ccpl.bits, 2);
}
return;
}
if(cplband[p->js_subband_start] > cplband[p->subbands-1]) return;
length = cplband[p->subbands-1] - cplband[p->js_subband_start] + 1;
for (i=0 ; i<length ; i++) {
decouple_tab[cplband[p->js_subband_start] + i] = get_bits(&q->gb, p->js_vlc_bits);
}
return;
}
/*
* function decouples a pair of signals from a single signal via multiplication.
*
* @param q pointer to the COOKContext
* @param subband index of the current subband
* @param f1 multiplier for channel 1 extraction
* @param f2 multiplier for channel 2 extraction
* @param decode_buffer input buffer
* @param mlt_buffer1 pointer to left channel mlt coefficients
* @param mlt_buffer2 pointer to right channel mlt coefficients
*/
static void decouple_float (COOKContext *q,
COOKSubpacket *p,
int subband,
float f1, float f2,
float *decode_buffer,
float *mlt_buffer1, float *mlt_buffer2)
{
int j, tmp_idx;
for (j=0 ; j<SUBBAND_SIZE ; j++) {
tmp_idx = ((p->js_subband_start + subband)*SUBBAND_SIZE)+j;
mlt_buffer1[SUBBAND_SIZE*subband + j] = f1 * decode_buffer[tmp_idx];
mlt_buffer2[SUBBAND_SIZE*subband + j] = f2 * decode_buffer[tmp_idx];
}
}
/**
* 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, COOKSubpacket *p, float* mlt_buffer1,
float* mlt_buffer2) {
int i,j;
int decouple_tab[SUBBAND_SIZE];
float *decode_buffer = q->decode_buffer_0;
int idx, cpl_tmp;
float f1,f2;
const 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, p, decouple_tab);
mono_decode(q, p, decode_buffer);
/* The two channels are stored interleaved in decode_buffer. */
for (i=0 ; i<p->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 << p->js_vlc_bits) - 1;
for (i=p->js_subband_start ; i<p->subbands ; i++) {
cpl_tmp = cplband[i];
idx -=decouple_tab[cpl_tmp];
cplscale = q->cplscales[p->js_vlc_bits-2]; //choose decoupler table
f1 = cplscale[decouple_tab[cpl_tmp]];
f2 = cplscale[idx-1];
q->decouple (q, p, i, f1, f2, decode_buffer, mlt_buffer1, mlt_buffer2);
idx = (1 << p->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 gains_ptr array of current/prev gain pointers
*/
static inline void
decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p, const uint8_t *inbuffer,
cook_gains *gains_ptr)
{
int offset;
offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,
p->bits_per_subpacket/8);
init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,
p->bits_per_subpacket);
decode_gain_info(&q->gb, gains_ptr->now);
/* Swap current and previous gains */
FFSWAP(int *, gains_ptr->now, gains_ptr->previous);
}
/**
* Saturate the output signal to signed 16bit integers.
*
* @param q pointer to the COOKContext
* @param chan channel to saturate
* @param out pointer to the output vector
*/
static void
saturate_output_float (COOKContext *q, int chan, int16_t *out)
{
int j;
float *output = q->mono_mdct_output + q->samples_per_channel;
/* Clip and convert floats to 16 bits.
*/
for (j = 0; j < q->samples_per_channel; j++) {
out[chan + q->nb_channels * j] =
av_clip_int16(lrintf(output[j]));
}
}
/**
* 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 gains_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_ptr, float *previous_buffer,
int16_t *out, int chan)
{
imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
q->saturate_output (q, chan, out);
}
/**
* 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 outbuffer pointer to the outbuffer
*/
static void decode_subpacket(COOKContext *q, COOKSubpacket* p, const uint8_t *inbuffer, int16_t *outbuffer) {
int sub_packet_size = p->size;
/* packet dump */
// for (i=0 ; i<sub_packet_size ; i++) {
// av_log(q->avctx, AV_LOG_ERROR, "%02x", inbuffer[i]);
// }
// av_log(q->avctx, AV_LOG_ERROR, "\n");
memset(q->decode_buffer_1,0,sizeof(q->decode_buffer_1));
decode_bytes_and_gain(q, p, inbuffer, &p->gains1);
if (p->joint_stereo) {
joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2);
} else {
mono_decode(q, p, q->decode_buffer_1);
if (p->num_channels == 2) {
decode_bytes_and_gain(q, p, inbuffer + sub_packet_size/2, &p->gains2);
mono_decode(q, p, q->decode_buffer_2);
}
}
mlt_compensate_output(q, q->decode_buffer_1, &p->gains1,
p->mono_previous_buffer1, outbuffer, p->ch_idx);
if (p->num_channels == 2) {
if (p->joint_stereo) {
mlt_compensate_output(q, q->decode_buffer_2, &p->gains1,
p->mono_previous_buffer2, outbuffer, p->ch_idx + 1);
} else {
mlt_compensate_output(q, q->decode_buffer_2, &p->gains2,
p->mono_previous_buffer2, outbuffer, p->ch_idx + 1);
}
}
}
/**
* Cook frame decoding
*
* @param avctx pointer to the AVCodecContext
*/
static int cook_decode_frame(AVCodecContext *avctx,
void *data, int *data_size,
AVPacket *avpkt) {
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
COOKContext *q = avctx->priv_data;
int i;
int offset = 0;
int chidx = 0;
if (buf_size < avctx->block_align)
return buf_size;
/* estimate subpacket sizes */
q->subpacket[0].size = avctx->block_align;
for(i=1;i<q->num_subpackets;i++){
q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];
q->subpacket[0].size -= q->subpacket[i].size + 1;
if (q->subpacket[0].size < 0) {
av_log(avctx,AV_LOG_DEBUG,"frame subpacket size total > avctx->block_align!\n");
return -1;
}
}
/* decode supbackets */
*data_size = 0;
for(i=0;i<q->num_subpackets;i++){
q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size*8)>>q->subpacket[i].bits_per_subpdiv;
q->subpacket[i].ch_idx = chidx;
av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] size %i js %i %i block_align %i\n",i,q->subpacket[i].size,q->subpacket[i].joint_stereo,offset,avctx->block_align);
decode_subpacket(q, &q->subpacket[i], buf + offset, (int16_t*)data);
offset += q->subpacket[i].size;
chidx += q->subpacket[i].num_channels;
av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] %i %i\n",i,q->subpacket[i].size * 8,get_bits_count(&q->gb));
}
*data_size = sizeof(int16_t) * q->nb_channels * q->samples_per_channel;
/* 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(q->avctx,AV_LOG_ERROR," %s = %d\n", a, b);
av_log(q->avctx,AV_LOG_ERROR,"COOKextradata\n");
av_log(q->avctx,AV_LOG_ERROR,"cookversion=%x\n",q->subpacket[0].cookversion);
if (q->subpacket[0].cookversion > STEREO) {
PRINT("js_subband_start",q->subpacket[0].js_subband_start);
PRINT("js_vlc_bits",q->subpacket[0].js_vlc_bits);
}
av_log(q->avctx,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->subpacket[0].samples_per_channel);
PRINT("samples_per_frame",q->subpacket[0].samples_per_frame);
PRINT("subbands",q->subpacket[0].subbands);
PRINT("random_state",q->random_state);
PRINT("js_subband_start",q->subpacket[0].js_subband_start);
PRINT("log2_numvector_size",q->subpacket[0].log2_numvector_size);
PRINT("numvector_size",q->subpacket[0].numvector_size);
PRINT("total_subbands",q->subpacket[0].total_subbands);
}
#endif
static av_cold int cook_count_channels(unsigned int mask){
int i;
int channels = 0;
for(i = 0;i<32;i++){
if(mask & (1<<i))
++channels;
}
return channels;
}
/**
* Cook initialization
*
* @param avctx pointer to the AVCodecContext
*/
static av_cold int cook_decode_init(AVCodecContext *avctx)
{
COOKContext *q = avctx->priv_data;
const uint8_t *edata_ptr = avctx->extradata;
const uint8_t *edata_ptr_end = edata_ptr + avctx->extradata_size;
int extradata_size = avctx->extradata_size;
int s = 0;
unsigned int channel_mask = 0;
q->avctx = avctx;
/* Take care of the codec specific extradata. */
if (extradata_size <= 0) {
av_log(avctx,AV_LOG_ERROR,"Necessary extradata missing!\n");
return -1;
}
av_log(avctx,AV_LOG_DEBUG,"codecdata_length=%d\n",avctx->extradata_size);
/* 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_lfg_init(&q->random_state, 0);
while(edata_ptr < edata_ptr_end){
/* 8 for mono, 16 for stereo, ? for multichannel
Swap to right endianness so we don't need to care later on. */
if (extradata_size >= 8){
q->subpacket[s].cookversion = bytestream_get_be32(&edata_ptr);
q->subpacket[s].samples_per_frame = bytestream_get_be16(&edata_ptr);
q->subpacket[s].subbands = bytestream_get_be16(&edata_ptr);
extradata_size -= 8;
}
if (avctx->extradata_size >= 8){
bytestream_get_be32(&edata_ptr); //Unknown unused
q->subpacket[s].js_subband_start = bytestream_get_be16(&edata_ptr);
q->subpacket[s].js_vlc_bits = bytestream_get_be16(&edata_ptr);
extradata_size -= 8;
}
/* Initialize extradata related variables. */
q->subpacket[s].samples_per_channel = q->subpacket[s].samples_per_frame / q->nb_channels;
q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;
/* Initialize default data states. */
q->subpacket[s].log2_numvector_size = 5;
q->subpacket[s].total_subbands = q->subpacket[s].subbands;
q->subpacket[s].num_channels = 1;
/* Initialize version-dependent variables */
av_log(avctx,AV_LOG_DEBUG,"subpacket[%i].cookversion=%x\n",s,q->subpacket[s].cookversion);
q->subpacket[s].joint_stereo = 0;
switch (q->subpacket[s].cookversion) {
case MONO:
if (q->nb_channels != 1) {
av_log_ask_for_sample(avctx, "Container channels != 1.\n");
return -1;
}
av_log(avctx,AV_LOG_DEBUG,"MONO\n");
break;
case STEREO:
if (q->nb_channels != 1) {
q->subpacket[s].bits_per_subpdiv = 1;
q->subpacket[s].num_channels = 2;
}
av_log(avctx,AV_LOG_DEBUG,"STEREO\n");
break;
case JOINT_STEREO:
if (q->nb_channels != 2) {
av_log_ask_for_sample(avctx, "Container channels != 2.\n");
return -1;
}
av_log(avctx,AV_LOG_DEBUG,"JOINT_STEREO\n");
if (avctx->extradata_size >= 16){
q->subpacket[s].total_subbands = q->subpacket[s].subbands + q->subpacket[s].js_subband_start;
q->subpacket[s].joint_stereo = 1;
q->subpacket[s].num_channels = 2;
}
if (q->subpacket[s].samples_per_channel > 256) {
q->subpacket[s].log2_numvector_size = 6;
}
if (q->subpacket[s].samples_per_channel > 512) {
q->subpacket[s].log2_numvector_size = 7;
}
break;
case MC_COOK:
av_log(avctx,AV_LOG_DEBUG,"MULTI_CHANNEL\n");
if(extradata_size >= 4)
channel_mask |= q->subpacket[s].channel_mask = bytestream_get_be32(&edata_ptr);
if(cook_count_channels(q->subpacket[s].channel_mask) > 1){
q->subpacket[s].total_subbands = q->subpacket[s].subbands + q->subpacket[s].js_subband_start;
q->subpacket[s].joint_stereo = 1;
q->subpacket[s].num_channels = 2;
q->subpacket[s].samples_per_channel = q->subpacket[s].samples_per_frame >> 1;
if (q->subpacket[s].samples_per_channel > 256) {
q->subpacket[s].log2_numvector_size = 6;
}
if (q->subpacket[s].samples_per_channel > 512) {
q->subpacket[s].log2_numvector_size = 7;
}
}else
q->subpacket[s].samples_per_channel = q->subpacket[s].samples_per_frame;
break;
default:
av_log_ask_for_sample(avctx, "Unknown Cook version.\n");
return -1;
break;
}
if(s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {
av_log(avctx,AV_LOG_ERROR,"different number of samples per channel!\n");
return -1;
} else
q->samples_per_channel = q->subpacket[0].samples_per_channel;
/* Initialize variable relations */
q->subpacket[s].numvector_size = (1 << q->subpacket[s].log2_numvector_size);
/* Try to catch some obviously faulty streams, othervise it might be exploitable */
if (q->subpacket[s].total_subbands > 53) {
av_log_ask_for_sample(avctx, "total_subbands > 53\n");
return -1;
}
if ((q->subpacket[s].js_vlc_bits > 6) || (q->subpacket[s].js_vlc_bits < 0)) {
av_log(avctx,AV_LOG_ERROR,"js_vlc_bits = %d, only >= 0 and <= 6 allowed!\n",q->subpacket[s].js_vlc_bits);
return -1;
}
if (q->subpacket[s].subbands > 50) {
av_log_ask_for_sample(avctx, "subbands > 50\n");
return -1;
}
q->subpacket[s].gains1.now = q->subpacket[s].gain_1;
q->subpacket[s].gains1.previous = q->subpacket[s].gain_2;
q->subpacket[s].gains2.now = q->subpacket[s].gain_3;
q->subpacket[s].gains2.previous = q->subpacket[s].gain_4;
q->num_subpackets++;
s++;
if (s > MAX_SUBPACKETS) {
av_log_ask_for_sample(avctx, "Too many subpackets > 5\n");
return -1;
}
}
/* Generate tables */
init_pow2table();
init_gain_table(q);
init_cplscales_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. */
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;
/* Initialize transform. */
if ( init_cook_mlt(q) != 0 )
return -1;
/* Initialize COOK signal arithmetic handling */
if (1) {
q->scalar_dequant = scalar_dequant_float;
q->decouple = decouple_float;
q->imlt_window = imlt_window_float;
q->interpolate = interpolate_float;
q->saturate_output = saturate_output_float;
}
/* Try to catch some obviously faulty streams, othervise it might be exploitable */
if ((q->samples_per_channel == 256) || (q->samples_per_channel == 512) || (q->samples_per_channel == 1024)) {
} else {
av_log_ask_for_sample(avctx,
"unknown amount of samples_per_channel = %d\n",
q->samples_per_channel);
return -1;
}
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
if (channel_mask)
avctx->channel_layout = channel_mask;
else
avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
#ifdef COOKDEBUG
dump_cook_context(q);
#endif
return 0;
}
AVCodec ff_cook_decoder =
{
.name = "cook",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_COOK,
.priv_data_size = sizeof(COOKContext),
.init = cook_decode_init,
.close = cook_decode_close,
.decode = cook_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("COOK"),
};