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72415b2adb
ffmpeg/libavcodec/cook.c
Stefano Sabatini 72415b2adb Define AVMediaType enum, and use it instead of enum CodecType, which 
is deprecated and will be dropped at the next major bump.

Originally committed as revision 22735 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-03-30 23:30:55 +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 libavcodec/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 "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 "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(16, 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 ***********/
/**
* 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
*/
#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(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^be2me_64(int64_t)in[i]);
* Buffer alignment needs to be checked. */
off = (intptr_t)inbuffer & 3;
buf = (const 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 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 q pointer to the COOKContext
* @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 *buffer1,
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++){
buffer1[i] = buffer1[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 */
ff_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 gain_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 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)
{
imlt_gain(q, decode_buffer, gains, 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 sub_packet_size subpacket size
* @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(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->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(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->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(avctx,AV_LOG_ERROR,"Unknown Cook version, report sample!\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(avctx,AV_LOG_ERROR,"total_subbands > 53, report sample!\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(avctx,AV_LOG_ERROR,"subbands > 50, report sample!\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(avctx,AV_LOG_ERROR,"Too many subpackets > 5, report file!\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(avctx,AV_LOG_ERROR,"unknown amount of samples_per_channel = %d, report sample!\n",q->samples_per_channel);
return -1;
}
avctx->sample_fmt = SAMPLE_FMT_S16;
if (channel_mask)
avctx->channel_layout = channel_mask;
else
avctx->channel_layout = (avctx->channels==2) ? CH_LAYOUT_STEREO : CH_LAYOUT_MONO;
#ifdef COOKDEBUG
dump_cook_context(q);
#endif
return 0;
}
AVCodec 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"),
};
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