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FFmpeg/libavcodec/cook.c
Benjamin Larsson e0f7e32970 Cook compatibe decoder, patch by Benjamin Larsson 
Add cook demucing, change rm demuxer so that it reorders audio packets
before sending them to the decoder, and send minimum decodeable sized
packets; pass only real codec extradata fo the decoder
Fix 28_8 decoder for the new demuxer strategy

Originally committed as revision 4726 to svn://svn.ffmpeg.org/ffmpeg/trunk
2005-12-09 16:08:18 +00:00

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/*
* COOK compatible decoder
* Copyright (c) 2003 Sascha Sommer
* Copyright (c) 2005 Benjamin Larsson
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
*/
/**
* @file cook.c
* Cook compatible decoder.
* This decoder handles RealNetworks, RealAudio G2 data.
* Cook is identified by the codec name cook in RM files.
*
* To use this decoder, a calling application must supply the extradata
* bytes provided from the RM container; 8+ bytes for mono streams and
* 16+ for stereo streams (maybe more).
*
* Codec technicalities (all this assume a buffer length of 1024):
* Cook works with several different techniques to achieve its compression.
* In the timedomain the buffer is divided into 8 pieces and quantized. If
* two neighboring pieces have different quantization index a smooth
* quantization curve is used to get a smooth overlap between the different
* pieces.
* To get to the transformdomain Cook uses a modulated lapped transform.
* The transform domain has 50 subbands with 20 elements each. This
* means only a maximum of 50*20=1000 coefficients are used out of the 1024
* available.
*/
#include <math.h>
#include <stddef.h>
#include <stdio.h>
#define ALT_BITSTREAM_READER
#include "avcodec.h"
#include "bitstream.h"
#include "dsputil.h"
#include "cookdata.h"
/* the different Cook versions */
#define MONO_COOK1 0x1000001
#define MONO_COOK2 0x1000002
#define JOINT_STEREO 0x1000003
#define MC_COOK 0x2000000 //multichannel Cook, not supported
#define SUBBAND_SIZE 20
//#define COOKDEBUG
typedef struct {
int size;
int qidx_table1[8];
int qidx_table2[8];
} COOKgain;
typedef struct __attribute__((__packed__)){
/* codec data start */
uint32_t cookversion; //in network order, bigendian
uint16_t samples_per_frame; //amount of samples per frame per channel, bigendian
uint16_t subbands; //amount of bands used in the frequency domain, bigendian
/* Mono extradata ends here. */
uint32_t unused;
uint16_t js_subband_start; //bigendian
uint16_t js_vlc_bits; //bigendian
/* Stereo extradata ends here. */
} COOKextradata;
typedef struct {
GetBitContext gb;
/* stream data */
int nb_channels;
int joint_stereo;
int bit_rate;
int sample_rate;
int samples_per_channel;
int samples_per_frame;
int subbands;
int numvector_bits;
int numvector_size; //1 << numvector_bits;
int js_subband_start;
int total_subbands;
int num_vectors;
int bits_per_subpacket;
/* states */
int random_state;
/* transform data */
FFTContext fft_ctx;
FFTSample mlt_tmp[1024] __attribute__((aligned(16))); /* temporary storage for imlt */
float* mlt_window;
float* mlt_precos;
float* mlt_presin;
float* mlt_postcos;
int fft_size;
int fft_order;
int mlt_size; //modulated lapped transform size
/* gain buffers */
COOKgain* gain_now_ptr;
COOKgain* gain_previous_ptr;
COOKgain gain_copy;
COOKgain gain_current;
COOKgain gain_now;
COOKgain gain_previous;
/* 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* frame_reorder_buffer;
int* frame_reorder_index;
int frame_reorder_counter;
int frame_reorder_complete;
int frame_reorder_index_size;
uint8_t* decoded_bytes_buffer;
float mono_mdct_output[2048] __attribute__((aligned(16)));
float* previous_buffer_ptr[2];
float mono_previous_buffer1[1024];
float mono_previous_buffer2[1024];
float* decode_buf_ptr[4];
float decode_buffer_1[1024];
float decode_buffer_2[1024];
float decode_buffer_3[1024];
float decode_buffer_4[1024];
} COOKContext;
/* debug functions */
#ifdef COOKDEBUG
static void dump_float_table(float* table, int size, int delimiter) {
int i=0;
av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i);
for (i=0 ; i<size ; i++) {
av_log(NULL, AV_LOG_ERROR, "%5.1f, ", table[i]);
if ((i+1)%delimiter == 0) av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i+1);
}
}
static void dump_int_table(int* table, int size, int delimiter) {
int i=0;
av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i);
for (i=0 ; i<size ; i++) {
av_log(NULL, AV_LOG_ERROR, "%d, ", table[i]);
if ((i+1)%delimiter == 0) av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i+1);
}
}
static void dump_short_table(short* table, int size, int delimiter) {
int i=0;
av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i);
for (i=0 ; i<size ; i++) {
av_log(NULL, AV_LOG_ERROR, "%d, ", table[i]);
if ((i+1)%delimiter == 0) av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i+1);
}
}
#endif
/*************** init functions ***************/
/* table generator */
static void init_pow2table(COOKContext *q){
int i;
q->pow2tab[63] = 1.0;
for (i=1 ; i<64 ; i++){
q->pow2tab[63+i]=(float)pow(2.0,(double)i);
q->pow2tab[63-i]=1.0/(float)pow(2.0,(double)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)powf(2.0,(float)i));
q->rootpow2tab[63-i]=sqrt(1.0/(float)powf(2.0,(float)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));
}
memset(&q->gain_copy, 0, sizeof(COOKgain));
memset(&q->gain_current, 0, sizeof(COOKgain));
memset(&q->gain_now, 0, sizeof(COOKgain));
memset(&q->gain_previous, 0, sizeof(COOKgain));
}
static int init_cook_vlc_tables(COOKContext *q) {
int i, result;
result = 0;
for (i=0 ; i<13 ; i++) {
result &= init_vlc (&q->envelope_quant_index[i], 9, 24,
envelope_quant_index_huffbits[i], 1, 1,
envelope_quant_index_huffcodes[i], 2, 2, 0);
}
av_log(NULL,AV_LOG_DEBUG,"sqvh VLC init\n");
for (i=0 ; i<7 ; i++) {
result &= init_vlc (&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i],
cvh_huffbits[i], 1, 1,
cvh_huffcodes[i], 2, 2, 0);
}
if (q->nb_channels==2 && q->joint_stereo==1){
result &= init_vlc (&q->ccpl, 6, (1<<q->js_vlc_bits)-1,
ccpl_huffbits[q->js_vlc_bits-2], 1, 1,
ccpl_huffcodes[q->js_vlc_bits-2], 2, 2, 0);
av_log(NULL,AV_LOG_DEBUG,"Joint-stereo VLC used.\n");
}
av_log(NULL,AV_LOG_DEBUG,"VLC tables initialized.\n");
return result;
}
static int init_cook_mlt(COOKContext *q) {
int j;
float alpha;
/* Allocate the buffers, could be replaced with a static [512]
array if needed. */
q->mlt_size = q->samples_per_channel;
q->mlt_window = av_malloc(sizeof(float)*q->mlt_size);
q->mlt_precos = av_malloc(sizeof(float)*q->mlt_size/2);
q->mlt_presin = av_malloc(sizeof(float)*q->mlt_size/2);
q->mlt_postcos = av_malloc(sizeof(float)*q->mlt_size/2);
/* Initialize the MLT window: simple sine window. */
alpha = M_PI / (2.0 * (float)q->mlt_size);
for(j=0 ; j<q->mlt_size ; j++) {
q->mlt_window[j] = sin((j + 512.0/(float)q->mlt_size) * alpha);
}
/* pre/post twiddle factors */
for (j=0 ; j<q->mlt_size/2 ; j++){
q->mlt_precos[j] = cos( ((j+0.25)*M_PI)/q->mlt_size);
q->mlt_presin[j] = sin( ((j+0.25)*M_PI)/q->mlt_size);
q->mlt_postcos[j] = (float)sqrt(2.0/(float)q->mlt_size)*cos( ((float)j*M_PI) /q->mlt_size); //sqrt(2/MLT_size) = scalefactor
}
/* Initialize the FFT. */
ff_fft_init(&q->fft_ctx, av_log2(q->mlt_size)-1, 0);
av_log(NULL,AV_LOG_DEBUG,"FFT initialized, order = %d.\n",
av_log2(q->samples_per_channel)-1);
return (int)(q->mlt_window && q->mlt_precos && q->mlt_presin && q->mlt_postcos);
}
/*************** init functions end ***********/
/**
* Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
* Why? No idea, some checksum/error detection method maybe.
* Nice way to waste CPU cycles.
*
* @param in pointer to 32bit array of indata
* @param bits amount of bits
* @param out pointer to 32bit array of outdata
*/
static inline void decode_bytes(uint8_t* inbuffer, uint8_t* out, int bytes){
int i;
uint32_t* buf = (uint32_t*) inbuffer;
uint32_t* obuf = (uint32_t*) out;
/* FIXME: 64 bit platforms would be able to do 64 bits at a time.
* I'm too lazy though, should be something like
* for(i=0 ; i<bitamount/64 ; i++)
* (int64_t)out[i] = 0x37c511f237c511f2^be2me_64(int64_t)in[i]);
* Buffer alignment needs to be checked. */
for(i=0 ; i<bytes/4 ; i++){
#ifdef WORDS_BIGENDIAN
obuf[i] = 0x37c511f2^buf[i];
#else
obuf[i] = 0xf211c537^buf[i];
#endif
}
}
/**
* Cook uninit
*/
static int cook_decode_close(AVCodecContext *avctx)
{
int i;
COOKContext *q = avctx->priv_data;
av_log(NULL,AV_LOG_DEBUG, "Deallocating memory.\n");
/* Free allocated memory buffers. */
av_free(q->mlt_window);
av_free(q->mlt_precos);
av_free(q->mlt_presin);
av_free(q->mlt_postcos);
av_free(q->frame_reorder_index);
av_free(q->frame_reorder_buffer);
av_free(q->decoded_bytes_buffer);
/* Free the transform. */
ff_fft_end(&q->fft_ctx);
/* Free the VLC tables. */
for (i=0 ; i<13 ; i++) {
free_vlc(&q->envelope_quant_index[i]);
}
for (i=0 ; i<7 ; i++) {
free_vlc(&q->sqvh[i]);
}
if(q->nb_channels==2 && q->joint_stereo==1 ){
free_vlc(&q->ccpl);
}
av_log(NULL,AV_LOG_DEBUG,"Memory deallocated.\n");
return 0;
}
/**
* Fill the COOKgain structure for the timedomain quantization.
*
* @param q pointer to the COOKContext
* @param gaininfo pointer to the COOKgain
*/
static void decode_gain_info(GetBitContext *gb, COOKgain* gaininfo) {
int i;
while (get_bits1(gb)) {}
gaininfo->size = get_bits_count(gb) - 1; //amount of elements*2 to update
if (get_bits_count(gb) - 1 <= 0) return;
for (i=0 ; i<gaininfo->size ; i++){
gaininfo->qidx_table1[i] = get_bits(gb,3);
if (get_bits1(gb)) {
gaininfo->qidx_table2[i] = get_bits(gb,4) - 7; //convert to signed
} else {
gaininfo->qidx_table2[i] = -1;
}
}
}
/**
* Create the quant index table needed for the envelope.
*
* @param q pointer to the COOKContext
* @param quant_index_table pointer to the array
*/
static void decode_envelope(COOKContext *q, int* quant_index_table) {
int i,j, vlc_index;
int bitbias;
bitbias = get_bits_count(&q->gb);
quant_index_table[0]= get_bits(&q->gb,6) - 6; //This is used later in categorize
for (i=1 ; i < q->total_subbands ; i++){
vlc_index=i;
if (i >= q->js_subband_start * 2) {
vlc_index-=q->js_subband_start;
} else {
vlc_index/=2;
if(vlc_index < 1) vlc_index = 1;
}
if (vlc_index>13) vlc_index = 13; //the VLC tables >13 are identical to No. 13
j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index-1].table,
q->envelope_quant_index[vlc_index-1].bits,2);
quant_index_table[i] = quant_index_table[i-1] + j - 12; //differential encoding
}
}
/**
* Create the quant value table.
*
* @param q pointer to the COOKContext
* @param quant_value_table pointer to the array
*/
static void inline dequant_envelope(COOKContext *q, int* quant_index_table,
float* quant_value_table){
int i;
for(i=0 ; i < q->total_subbands ; i++){
quant_value_table[i] = q->rootpow2tab[quant_index_table[i]+63];
}
}
/**
* Calculate the category and category_index vector.
*
* @param q pointer to the COOKContext
* @param quant_index_table pointer to the array
* @param category pointer to the category array
* @param category_index pointer to the category_index array
*/
static void categorize(COOKContext *q, int* quant_index_table,
int* category, int* category_index){
int exp_idx, bias, tmpbias, bits_left, num_bits, index, v, i, j;
int exp_index2[102];
int exp_index1[102];
int tmp_categorize_array1[128];
int tmp_categorize_array1_idx=0;
int tmp_categorize_array2[128];
int tmp_categorize_array2_idx=0;
int category_index_size=0;
bits_left = q->bits_per_subpacket - get_bits_count(&q->gb);
if(bits_left > q->samples_per_channel) {
bits_left = q->samples_per_channel +
((bits_left - q->samples_per_channel)*5)/8;
//av_log(NULL, AV_LOG_ERROR, "bits_left = %d\n",bits_left);
}
memset(&exp_index1,0,102*sizeof(int));
memset(&exp_index2,0,102*sizeof(int));
memset(&tmp_categorize_array1,0,128*sizeof(int));
memset(&tmp_categorize_array2,0,128*sizeof(int));
bias=-32;
/* Estimate bias. */
for (i=32 ; i>0 ; i=i/2){
num_bits = 0;
index = 0;
for (j=q->total_subbands ; j>0 ; j--){
exp_idx = (i - quant_index_table[index] + bias) / 2;
if (exp_idx<0){
exp_idx=0;
} else if(exp_idx >7) {
exp_idx=7;
}
index++;
num_bits+=expbits_tab[exp_idx];
}
if(num_bits >= bits_left - 32){
bias+=i;
}
}
/* Calculate total number of bits. */
num_bits=0;
for (i=0 ; i<q->total_subbands ; i++) {
exp_idx = (bias - quant_index_table[i]) / 2;
if (exp_idx<0) {
exp_idx=0;
} else if(exp_idx >7) {
exp_idx=7;
}
num_bits += expbits_tab[exp_idx];
exp_index1[i] = exp_idx;
exp_index2[i] = exp_idx;
}
tmpbias = bias = num_bits;
for (j = 1 ; j < q->numvector_size ; j++) {
if (tmpbias + bias > 2*bits_left) { /* ---> */
int max = -999999;
index=-1;
for (i=0 ; i<q->total_subbands ; i++){
if (exp_index1[i] < 7) {
v = (-2*exp_index1[i]) - quant_index_table[i] - 32;
if ( v >= max) {
max = v;
index = i;
}
}
}
if(index==-1)break;
tmp_categorize_array1[tmp_categorize_array1_idx++] = index;
tmpbias -= expbits_tab[exp_index1[index]] -
expbits_tab[exp_index1[index]+1];
++exp_index1[index];
} else { /* <--- */
int min = 999999;
index=-1;
for (i=0 ; i<q->total_subbands ; i++){
if(exp_index2[i] > 0){
v = (-2*exp_index2[i])-quant_index_table[i];
if ( v < min) {
min = v;
index = i;
}
}
}
if(index == -1)break;
tmp_categorize_array2[tmp_categorize_array2_idx++] = index;
tmpbias -= expbits_tab[exp_index2[index]] -
expbits_tab[exp_index2[index]-1];
--exp_index2[index];
}
}
for(i=0 ; i<q->total_subbands ; i++)
category[i] = exp_index2[i];
/* Concatenate the two arrays. */
for(i=tmp_categorize_array2_idx-1 ; i >= 0; i--)
category_index[category_index_size++] = tmp_categorize_array2[i];
for(i=0;i<tmp_categorize_array1_idx;i++)
category_index[category_index_size++ ] = tmp_categorize_array1[i];
/* FIXME: mc_sich_ra8_20.rm triggers this, not sure with what we
should fill the remaining bytes. */
for(i=category_index_size;i<q->numvector_size;i++)
category_index[i]=0;
}
/**
* Expand the category vector.
*
* @param q pointer to the COOKContext
* @param category pointer to the category array
* @param category_index pointer to the category_index array
*/
static void inline expand_category(COOKContext *q, int* category,
int* category_index){
int i;
for(i=0 ; i<q->num_vectors ; i++){
++category[category_index[i]];
}
}
/**
* The real requantization of the mltcoefs
*
* @param q pointer to the COOKContext
* @param index index
* @param band current subband
* @param quant_value_table pointer to the array
* @param subband_coef_index array of indexes to quant_centroid_tab
* @param subband_coef_noise use random noise instead of predetermined value
* @param mlt_buffer pointer to the mlt buffer
*/
static void scalar_dequant(COOKContext *q, int index, int band,
float* quant_value_table, int* subband_coef_index,
int* subband_coef_noise, float* mlt_buffer){
int i;
float f1;
for(i=0 ; i<SUBBAND_SIZE ; i++) {
if (subband_coef_index[i]) {
if (subband_coef_noise[i]) {
f1 = -quant_centroid_tab[index][subband_coef_index[i]];
} else {
f1 = quant_centroid_tab[index][subband_coef_index[i]];
}
} else {
/* noise coding if subband_coef_noise[i] == 0 */
q->random_state = q->random_state * 214013 + 2531011; //typical RNG numbers
f1 = randsign[(q->random_state/0x1000000)&1] * dither_tab[index]; //>>31
}
mlt_buffer[band*20+ i] = f1 * quant_value_table[band];
}
}
/**
* Unpack the subband_coef_index and subband_coef_noise vectors.
*
* @param q pointer to the COOKContext
* @param category pointer to the category array
* @param subband_coef_index array of indexes to quant_centroid_tab
* @param subband_coef_noise use random noise instead of predetermined value
*/
static int unpack_SQVH(COOKContext *q, int category, int* subband_coef_index,
int* subband_coef_noise) {
int i,j;
int vlc, vd ,tmp, result;
int ub;
int cb;
vd = vd_tab[category];
result = 0;
for(i=0 ; i<vpr_tab[category] ; i++){
ub = get_bits_count(&q->gb);
vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);
cb = get_bits_count(&q->gb);
if (q->bits_per_subpacket < get_bits_count(&q->gb)){
vlc = 0;
result = 1;
}
for(j=vd-1 ; j>=0 ; j--){
tmp = (vlc * invradix_tab[category])/0x100000;
subband_coef_index[vd*i+j] = vlc - tmp * (kmax_tab[category]+1);
vlc = tmp;
}
for(j=0 ; j<vd ; j++){
if (subband_coef_index[i*vd + j]) {
if(get_bits_count(&q->gb) < q->bits_per_subpacket){
subband_coef_noise[i*vd+j] = get_bits1(&q->gb);
} else {
result=1;
subband_coef_noise[i*vd+j]=0;
}
} else {
subband_coef_noise[i*vd+j]=0;
}
}
}
return result;
}
/**
* Fill the mlt_buffer with mlt coefficients.
*
* @param q pointer to the COOKContext
* @param category pointer to the category array
* @param quant_value_table pointer to the array
* @param mlt_buffer pointer to mlt coefficients
*/
static void decode_vectors(COOKContext* q, int* category,
float* quant_value_table, float* mlt_buffer){
/* A zero in this table means that the subband coefficient is
random noise coded. */
int subband_coef_noise[SUBBAND_SIZE];
/* A zero in this table means that the subband coefficient is a
positive multiplicator. */
int subband_coef_index[SUBBAND_SIZE];
int band, j;
int index=0;
for(band=0 ; band<q->total_subbands ; band++){
index = category[band];
if(category[band] < 7){
if(unpack_SQVH(q, category[band], subband_coef_index, subband_coef_noise)){
index=7;
for(j=0 ; j<q->total_subbands ; j++) category[band+j]=7;
}
}
if(index==7) {
memset(subband_coef_index, 0, sizeof(subband_coef_index));
memset(subband_coef_noise, 0, sizeof(subband_coef_noise));
}
scalar_dequant(q, index, band, quant_value_table, subband_coef_index,
subband_coef_noise, mlt_buffer);
}
if(q->total_subbands*SUBBAND_SIZE >= q->samples_per_channel){
return;
}
}
/**
* function for decoding mono data
*
* @param q pointer to the COOKContext
* @param mlt_buffer1 pointer to left channel mlt coefficients
* @param mlt_buffer2 pointer to right channel mlt coefficients
*/
static void mono_decode(COOKContext *q, float* mlt_buffer) {
int category_index[128];
float quant_value_table[102];
int quant_index_table[102];
int category[128];
memset(&category, 0, 128*sizeof(int));
memset(&quant_value_table, 0, 102*sizeof(int));
memset(&category_index, 0, 128*sizeof(int));
decode_envelope(q, quant_index_table);
q->num_vectors = get_bits(&q->gb,q->numvector_bits);
dequant_envelope(q, quant_index_table, quant_value_table);
categorize(q, quant_index_table, category, category_index);
expand_category(q, category, category_index);
decode_vectors(q, category, quant_value_table, mlt_buffer);
}
/**
* The modulated lapped transform, this takes transform coefficients
* and transforms them into timedomain samples. This is done through
* an FFT-based algorithm with pre- and postrotation steps.
* A window and reorder step is also included.
*
* @param q pointer to the COOKContext
* @param inbuffer pointer to the mltcoefficients
* @param outbuffer pointer to the timedomain buffer
* @param mlt_tmp pointer to temporary storage space
*/
static void cook_imlt(COOKContext *q, float* inbuffer, float* outbuffer,
float* mlt_tmp){
int i;
/* prerotation */
for(i=0 ; i<q->mlt_size ; i+=2){
outbuffer[i] = (q->mlt_presin[i/2] * inbuffer[q->mlt_size-1-i]) +
(q->mlt_precos[i/2] * inbuffer[i]);
outbuffer[i+1] = (q->mlt_precos[i/2] * inbuffer[q->mlt_size-1-i]) -
(q->mlt_presin[i/2] * inbuffer[i]);
}
/* FFT */
ff_fft_permute(&q->fft_ctx, (FFTComplex *) outbuffer);
ff_fft_calc (&q->fft_ctx, (FFTComplex *) outbuffer);
/* postrotation */
for(i=0 ; i<q->mlt_size ; i+=2){
mlt_tmp[i] = (q->mlt_postcos[(q->mlt_size-1-i)/2] * outbuffer[i+1]) +
(q->mlt_postcos[i/2] * outbuffer[i]);
mlt_tmp[q->mlt_size-1-i] = (q->mlt_postcos[(q->mlt_size-1-i)/2] * outbuffer[i]) -
(q->mlt_postcos[i/2] * outbuffer[i+1]);
}
/* window and reorder */
for(i=0 ; i<q->mlt_size/2 ; i++){
outbuffer[i] = mlt_tmp[q->mlt_size/2-1-i] * q->mlt_window[i];
outbuffer[q->mlt_size-1-i]= mlt_tmp[q->mlt_size/2-1-i] *
q->mlt_window[q->mlt_size-1-i];
outbuffer[q->mlt_size+i]= mlt_tmp[q->mlt_size/2+i] *
q->mlt_window[q->mlt_size-1-i];
outbuffer[2*q->mlt_size-1-i]= -(mlt_tmp[q->mlt_size/2+i] *
q->mlt_window[i]);
}
}
/**
* the actual requantization of the timedomain samples
*
* @param q pointer to the COOKContext
* @param buffer pointer to the timedomain buffer
* @param gain_index index for the block multiplier
* @param gain_index_next index for the next block multiplier
*/
static void interpolate(COOKContext *q, float* buffer,
int gain_index, int gain_index_next){
int i;
float fc1, fc2;
fc1 = q->pow2tab[gain_index+63];
if(gain_index == gain_index_next){ //static gain
for(i=0 ; i<q->gain_size_factor ; i++){
buffer[i]*=fc1;
}
return;
} else { //smooth gain
fc2 = q->gain_table[11 + (gain_index_next-gain_index)];
for(i=0 ; i<q->gain_size_factor ; i++){
buffer[i]*=fc1;
fc1*=fc2;
}
return;
}
}
/**
* timedomain requantization of the timedomain samples
*
* @param q pointer to the COOKContext
* @param buffer pointer to the timedomain buffer
* @param gain_now current gain structure
* @param gain_previous previous gain structure
*/
static void gain_window(COOKContext *q, float* buffer, COOKgain* gain_now,
COOKgain* gain_previous){
int i, index;
int gain_index[9];
int tmp_gain_index;
gain_index[8]=0;
index = gain_previous->size;
for (i=7 ; i>=0 ; i--) {
if(index && gain_previous->qidx_table1[index-1]==i) {
gain_index[i] = gain_previous->qidx_table2[index-1];
index--;
} else {
gain_index[i]=gain_index[i+1];
}
}
/* This is applied to the to be previous data buffer. */
for(i=0;i<8;i++){
interpolate(q, &buffer[q->samples_per_channel+q->gain_size_factor*i],
gain_index[i], gain_index[i+1]);
}
tmp_gain_index = gain_index[0];
index = gain_now->size;
for (i=7 ; i>=0 ; i--) {
if(index && gain_now->qidx_table1[index-1]==i) {
gain_index[i]= gain_now->qidx_table2[index-1];
index--;
} else {
gain_index[i]=gain_index[i+1];
}
}
/* This is applied to the to be current block. */
for(i=0;i<8;i++){
interpolate(q, &buffer[i*q->gain_size_factor],
tmp_gain_index+gain_index[i],
tmp_gain_index+gain_index[i+1]);
}
}
/**
* mlt overlapping and buffer management
*
* @param q pointer to the COOKContext
* @param buffer pointer to the timedomain buffer
* @param gain_now current gain structure
* @param gain_previous previous gain structure
* @param previous_buffer pointer to the previous buffer to be used for overlapping
*
*/
static void gain_compensate(COOKContext *q, float* buffer, COOKgain* gain_now,
COOKgain* gain_previous, float* previous_buffer) {
int i;
if((gain_now->size || gain_previous->size)) {
gain_window(q, buffer, gain_now, gain_previous);
}
/* Overlap with the previous block. */
for(i=0 ; i<q->samples_per_channel ; i++) buffer[i]+=previous_buffer[i];
/* Save away the current to be previous block. */
memcpy(previous_buffer, buffer+q->samples_per_channel,
sizeof(float)*q->samples_per_channel);
}
/**
* function for getting the jointstereo coupling information
*
* @param q pointer to the COOKContext
* @param decouple_tab decoupling array
*
*/
static void decouple_info(COOKContext *q, int* decouple_tab){
int length, i;
if(get_bits1(&q->gb)) {
if(cplband[q->js_subband_start] > cplband[q->subbands-1]) return;
length = cplband[q->subbands-1] - cplband[q->js_subband_start] + 1;
for (i=0 ; i<length ; i++) {
decouple_tab[cplband[q->js_subband_start] + i] = get_vlc2(&q->gb, q->ccpl.table, q->ccpl.bits, 2);
}
return;
}
if(cplband[q->js_subband_start] > cplband[q->subbands-1]) return;
length = cplband[q->subbands-1] - cplband[q->js_subband_start] + 1;
for (i=0 ; i<length ; i++) {
decouple_tab[cplband[q->js_subband_start] + i] = get_bits(&q->gb, q->js_vlc_bits);
}
return;
}
/**
* function for decoding joint stereo data
*
* @param q pointer to the COOKContext
* @param mlt_buffer1 pointer to left channel mlt coefficients
* @param mlt_buffer2 pointer to right channel mlt coefficients
*/
static void joint_decode(COOKContext *q, float* mlt_buffer1,
float* mlt_buffer2) {
int i,j;
int decouple_tab[SUBBAND_SIZE];
float decode_buffer[2048]; //Only 1060 might be needed.
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;
if (q->js_subband_start < q->subbands) {
for (i=0 ; i<q->subbands ; i++) {
cpl_tmp = cplband[i + q->js_subband_start];
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 = ((2*q->js_subband_start + i)*20)+j;
mlt_buffer1[20*(i+q->js_subband_start) + j] = f1 * decode_buffer[tmp_idx];
mlt_buffer2[20*(i+q->js_subband_start) + j] = f2 * decode_buffer[tmp_idx];
}
idx = (1 << q->js_vlc_bits) - 1;
}
}
}
/**
* Cook subpacket decoding. This function returns one decoded subpacket,
* usually 1024 samples per channel.
*
* @param q pointer to the COOKContext
* @param inbuffer pointer to the inbuffer
* @param sub_packet_size subpacket size
* @param outbuffer pointer to the outbuffer
* @param pos the subpacket number in the frame
*/
static int decode_subpacket(COOKContext *q, uint8_t *inbuffer,
int sub_packet_size, int16_t *outbuffer) {
int i,j;
int value;
float* tmp_ptr;
/* packet dump */
// for (i=0 ; i<sub_packet_size ; i++) {
// av_log(NULL, AV_LOG_ERROR, "%02x", inbuffer[i]);
// }
// av_log(NULL, AV_LOG_ERROR, "\n");
decode_bytes(inbuffer, q->decoded_bytes_buffer, sub_packet_size);
init_get_bits(&q->gb, q->decoded_bytes_buffer, sub_packet_size*8);
decode_gain_info(&q->gb, &q->gain_current);
memcpy(&q->gain_copy, &q->gain_current ,sizeof(COOKgain)); //This copy does not seem to be used. FIXME
//fprintf(stdout,"cu bits ds = %d\n",get_bits_count(&q->gb));
if(q->nb_channels==2 && q->joint_stereo==1){
joint_decode(q, q->decode_buf_ptr[0], q->decode_buf_ptr[2]);
/* Swap buffer pointers. */
tmp_ptr = q->decode_buf_ptr[1];
q->decode_buf_ptr[1] = q->decode_buf_ptr[0];
q->decode_buf_ptr[0] = tmp_ptr;
tmp_ptr = q->decode_buf_ptr[3];
q->decode_buf_ptr[3] = q->decode_buf_ptr[2];
q->decode_buf_ptr[2] = tmp_ptr;
/* FIXME: Rethink the gainbuffer handling, maybe a rename?
now/previous swap */
q->gain_now_ptr = &q->gain_now;
q->gain_previous_ptr = &q->gain_previous;
for (i=0 ; i<q->nb_channels ; i++){
cook_imlt(q, q->decode_buf_ptr[i*2], q->mono_mdct_output, q->mlt_tmp);
gain_compensate(q, q->mono_mdct_output, q->gain_now_ptr,
q->gain_previous_ptr, q->previous_buffer_ptr[0]);
/* Swap out the previous buffer. */
tmp_ptr = q->previous_buffer_ptr[0];
q->previous_buffer_ptr[0] = q->previous_buffer_ptr[1];
q->previous_buffer_ptr[1] = tmp_ptr;
/* Clip and convert the floats to 16 bits. */
for (j=0 ; j<q->samples_per_frame ; j++){
value = lrintf(q->mono_mdct_output[j]);
if(value < -32768) value = -32768;
else if(value > 32767) value = 32767;
outbuffer[2*j+i] = value;
}
}
memcpy(&q->gain_now, &q->gain_previous, sizeof(COOKgain));
memcpy(&q->gain_previous, &q->gain_current, sizeof(COOKgain));
} else if (q->nb_channels==2 && q->joint_stereo==0) {
for (i=0 ; i<q->nb_channels ; i++){
mono_decode(q, q->decode_buf_ptr[0]);
av_log(NULL,AV_LOG_ERROR,"Non-joint-stereo files are not supported at the moment, do not report as a bug!\n");
tmp_ptr = q->decode_buf_ptr[0];
q->decode_buf_ptr[0] = q->decode_buf_ptr[1];
q->decode_buf_ptr[1] = q->decode_buf_ptr[2];
q->decode_buf_ptr[2] = q->decode_buf_ptr[3];
q->decode_buf_ptr[3] = tmp_ptr;
q->gain_now_ptr = &q->gain_now;
q->gain_previous_ptr = &q->gain_previous;
cook_imlt(q, q->decode_buf_ptr[0], q->mono_mdct_output,q->mlt_tmp);
gain_compensate(q, q->mono_mdct_output, q->gain_now_ptr,
q->gain_previous_ptr, q->previous_buffer_ptr[0]);
/* Swap out the previous buffer. */
tmp_ptr = q->previous_buffer_ptr[0];
q->previous_buffer_ptr[0] = q->previous_buffer_ptr[1];
q->previous_buffer_ptr[1] = tmp_ptr;
for (j=0 ; j<q->samples_per_frame ; j++){
value = lrintf(q->mono_mdct_output[j]);
if(value < -32768) value = -32768;
else if(value > 32767) value = 32767;
outbuffer[2*j+i] = value;
}
memcpy(&q->gain_now, &q->gain_previous, sizeof(COOKgain));
memcpy(&q->gain_previous, &q->gain_current, sizeof(COOKgain));
}
} else {
mono_decode(q, q->decode_buf_ptr[0]);
/* Swap buffer pointers. */
tmp_ptr = q->decode_buf_ptr[1];
q->decode_buf_ptr[1] = q->decode_buf_ptr[0];
q->decode_buf_ptr[0] = tmp_ptr;
/* FIXME: Rethink the gainbuffer handling, maybe a rename?
now/previous swap */
q->gain_now_ptr = &q->gain_now;
q->gain_previous_ptr = &q->gain_previous;
cook_imlt(q, q->decode_buf_ptr[0], q->mono_mdct_output,q->mlt_tmp);
gain_compensate(q, q->mono_mdct_output, q->gain_now_ptr,
q->gain_previous_ptr, q->mono_previous_buffer1);
/* Clip and convert the floats to 16 bits */
for (j=0 ; j<q->samples_per_frame ; j++){
value = lrintf(q->mono_mdct_output[j]);
if(value < -32768) value = -32768;
else if(value > 32767) value = 32767;
outbuffer[j] = value;
}
memcpy(&q->gain_now, &q->gain_previous, sizeof(COOKgain));
memcpy(&q->gain_previous, &q->gain_current, sizeof(COOKgain));
}
/* FIXME: Shouldn't the total number of bytes be returned? */
return /*q->nb_channels*/ 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) {
/* This stuff is quite messy, the Cook packets are sent unordered
* and need to be ordered before they are sent to the rest of the
* decoder. The order can be found in the q->frame_reorder_index.
* Currently decoding of the last packets is not handled at
* all. FIXME */
COOKContext *q = avctx->priv_data;
if (buf_size < avctx->block_align)
return buf_size;
*data_size = decode_subpacket(q, buf, avctx->block_align, data);
return avctx->block_align;
}
#ifdef COOKDEBUG
static void dump_cook_context(COOKContext *q, COOKextradata *e)
{
//int i=0;
#define PRINT(a,b) av_log(NULL,AV_LOG_ERROR," %s = %d\n", a, b);
av_log(NULL,AV_LOG_ERROR,"COOKextradata\n");
av_log(NULL,AV_LOG_ERROR,"cookversion=%x\n",e->cookversion);
if (e->cookversion > MONO_COOK2) {
PRINT("js_subband_start",e->js_subband_start);
PRINT("js_vlc_bits",e->js_vlc_bits);
}
av_log(NULL,AV_LOG_ERROR,"COOKContext\n");
PRINT("nb_channels",q->nb_channels);
PRINT("bit_rate",q->bit_rate);
PRINT("sample_rate",q->sample_rate);
PRINT("samples_per_channel",q->samples_per_channel);
PRINT("samples_per_frame",q->samples_per_frame);
PRINT("subbands",q->subbands);
PRINT("random_state",q->random_state);
PRINT("mlt_size",q->mlt_size);
PRINT("js_subband_start",q->js_subband_start);
PRINT("numvector_bits",q->numvector_bits);
PRINT("numvector_size",q->numvector_size);
PRINT("total_subbands",q->total_subbands);
PRINT("frame_reorder_counter",q->frame_reorder_counter);
PRINT("frame_reorder_index_size",q->frame_reorder_index_size);
}
#endif
/**
* Cook initialization
*
* @param avctx pointer to the AVCodecContext
*/
static int cook_decode_init(AVCodecContext *avctx)
{
COOKextradata *e = avctx->extradata;
COOKContext *q = avctx->priv_data;
/* Take care of the codec specific extradata. */
if (avctx->extradata_size <= 0) {
av_log(NULL,AV_LOG_ERROR,"Necessary extradata missing!\n");
return -1;
} else {
/* 8 for mono, 16 for stereo, ? for multichannel
Swap to right endianness so we don't need to care later on. */
av_log(NULL,AV_LOG_DEBUG,"codecdata_length=%d\n",avctx->extradata_size);
if (avctx->extradata_size >= 8){
e->cookversion = be2me_32(e->cookversion);
e->samples_per_frame = be2me_16(e->samples_per_frame);
e->subbands = be2me_16(e->subbands);
}
if (avctx->extradata_size >= 16){
e->js_subband_start = be2me_16(e->js_subband_start);
e->js_vlc_bits = be2me_16(e->js_vlc_bits);
}
}
/* Take data from the AVCodecContext (RM container). */
q->sample_rate = avctx->sample_rate;
q->nb_channels = avctx->channels;
q->bit_rate = avctx->bit_rate;
/* Initialize state. */
q->random_state = 1;
/* Initialize extradata related variables. */
q->samples_per_channel = e->samples_per_frame / q->nb_channels;
q->samples_per_frame = e->samples_per_frame;
q->subbands = e->subbands;
q->bits_per_subpacket = avctx->block_align * 8;
/* Initialize default data states. */
q->js_subband_start = 0;
q->numvector_bits = 5;
q->total_subbands = q->subbands;
/* Initialize version-dependent variables */
av_log(NULL,AV_LOG_DEBUG,"e->cookversion=%x\n",e->cookversion);
switch (e->cookversion) {
case MONO_COOK1:
if (q->nb_channels != 1) {
av_log(NULL,AV_LOG_ERROR,"Container channels != 1, report sample!\n");
return -1;
}
av_log(NULL,AV_LOG_DEBUG,"MONO_COOK1\n");
break;
case MONO_COOK2:
if (q->nb_channels != 1) {
q->joint_stereo = 0;
av_log(NULL,AV_LOG_ERROR,"Non-joint-stereo files are not supported at the moment!\n");
return -1;
}
av_log(NULL,AV_LOG_DEBUG,"MONO_COOK2\n");
break;
case JOINT_STEREO:
if (q->nb_channels != 2) {
av_log(NULL,AV_LOG_ERROR,"Container channels != 2, report sample!\n");
return -1;
}
av_log(NULL,AV_LOG_DEBUG,"JOINT_STEREO\n");
if (avctx->extradata_size >= 16){
q->total_subbands = q->subbands + e->js_subband_start;
q->js_subband_start = e->js_subband_start;
q->joint_stereo = 1;
q->js_vlc_bits = e->js_vlc_bits;
}
if (q->samples_per_channel > 256) {
q->numvector_bits++; // q->numvector_bits = 6
}
if (q->samples_per_channel > 512) {
q->numvector_bits++; // q->numvector_bits = 7
}
break;
case MC_COOK:
av_log(NULL,AV_LOG_ERROR,"MC_COOK not supported!\n");
return -1;
break;
default:
av_log(NULL,AV_LOG_ERROR,"Unknown Cook version, report sample!\n");
return -1;
break;
}
/* Initialize variable relations */
q->mlt_size = q->samples_per_channel;
q->numvector_size = (1 << q->numvector_bits);
/* Generate tables */
init_rootpow2table(q);
init_pow2table(q);
init_gain_table(q);
if (init_cook_vlc_tables(q) != 0)
return -1;
/* Pad the databuffer with FF_INPUT_BUFFER_PADDING_SIZE,
this is for the bitstreamreader. */
if ((q->decoded_bytes_buffer = av_mallocz((avctx->block_align+(4-avctx->block_align%4) + FF_INPUT_BUFFER_PADDING_SIZE)*sizeof(uint8_t))) == NULL)
return -1;
q->decode_buf_ptr[0] = q->decode_buffer_1;
q->decode_buf_ptr[1] = q->decode_buffer_2;
q->decode_buf_ptr[2] = q->decode_buffer_3;
q->decode_buf_ptr[3] = q->decode_buffer_4;
q->previous_buffer_ptr[0] = q->mono_previous_buffer1;
q->previous_buffer_ptr[1] = q->mono_previous_buffer2;
memset(q->decode_buffer_1,0,1024*sizeof(float));
memset(q->decode_buffer_2,0,1024*sizeof(float));
memset(q->decode_buffer_3,0,1024*sizeof(float));
memset(q->decode_buffer_4,0,1024*sizeof(float));
/* Initialize transform. */
if ( init_cook_mlt(q) == 0 )
return -1;
//dump_cook_context(q,e);
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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