7d485f165f
Scaling (i)MDCT output has no runtime overhead and can be used to improve performance of audio codecs. All the changes are only needed in 'ff_mdct_init' function and slow down initialization a bit. Originally committed as revision 18855 to svn://svn.ffmpeg.org/ffmpeg/trunk
1298 lines
43 KiB
C
1298 lines
43 KiB
C
/*
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* COOK compatible decoder
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* Copyright (c) 2003 Sascha Sommer
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* Copyright (c) 2005 Benjamin Larsson
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file libavcodec/cook.c
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* Cook compatible decoder. Bastardization of the G.722.1 standard.
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* This decoder handles RealNetworks, RealAudio G2 data.
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* Cook is identified by the codec name cook in RM files.
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*
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* To use this decoder, a calling application must supply the extradata
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* bytes provided from the RM container; 8+ bytes for mono streams and
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* 16+ for stereo streams (maybe more).
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*
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* Codec technicalities (all this assume a buffer length of 1024):
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* Cook works with several different techniques to achieve its compression.
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* In the timedomain the buffer is divided into 8 pieces and quantized. If
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* two neighboring pieces have different quantization index a smooth
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* quantization curve is used to get a smooth overlap between the different
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* pieces.
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* To get to the transformdomain Cook uses a modulated lapped transform.
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* The transform domain has 50 subbands with 20 elements each. This
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* means only a maximum of 50*20=1000 coefficients are used out of the 1024
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* available.
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*/
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#include <math.h>
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#include <stddef.h>
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#include <stdio.h>
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#include "libavutil/lfg.h"
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#include "libavutil/random_seed.h"
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#include "avcodec.h"
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#include "get_bits.h"
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#include "dsputil.h"
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#include "bytestream.h"
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#include "cookdata.h"
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/* the different Cook versions */
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#define MONO 0x1000001
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#define STEREO 0x1000002
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#define JOINT_STEREO 0x1000003
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#define MC_COOK 0x2000000 //multichannel Cook, not supported
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#define SUBBAND_SIZE 20
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#define MAX_SUBPACKETS 5
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//#define COOKDEBUG
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typedef struct {
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int *now;
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int *previous;
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} cook_gains;
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typedef struct {
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int ch_idx;
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int size;
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int num_channels;
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int cookversion;
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int samples_per_frame;
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int subbands;
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int js_subband_start;
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int js_vlc_bits;
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int samples_per_channel;
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int log2_numvector_size;
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unsigned int channel_mask;
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VLC ccpl; ///< channel coupling
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int joint_stereo;
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int bits_per_subpacket;
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int bits_per_subpdiv;
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int total_subbands;
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int numvector_size; ///< 1 << log2_numvector_size;
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float mono_previous_buffer1[1024];
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float mono_previous_buffer2[1024];
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/** gain buffers */
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cook_gains gains1;
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cook_gains gains2;
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int gain_1[9];
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int gain_2[9];
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int gain_3[9];
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int gain_4[9];
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} COOKSubpacket;
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typedef struct cook {
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/*
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* The following 5 functions provide the lowlevel arithmetic on
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* the internal audio buffers.
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*/
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void (* scalar_dequant)(struct cook *q, int index, int quant_index,
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int* subband_coef_index, int* subband_coef_sign,
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float* mlt_p);
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void (* decouple) (struct cook *q,
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COOKSubpacket *p,
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int subband,
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float f1, float f2,
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float *decode_buffer,
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float *mlt_buffer1, float *mlt_buffer2);
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void (* imlt_window) (struct cook *q, float *buffer1,
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cook_gains *gains_ptr, float *previous_buffer);
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void (* interpolate) (struct cook *q, float* buffer,
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int gain_index, int gain_index_next);
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void (* saturate_output) (struct cook *q, int chan, int16_t *out);
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AVCodecContext* avctx;
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GetBitContext gb;
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/* stream data */
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int nb_channels;
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int bit_rate;
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int sample_rate;
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int num_vectors;
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int samples_per_channel;
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/* states */
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AVLFG random_state;
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/* transform data */
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MDCTContext mdct_ctx;
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float* mlt_window;
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/* VLC data */
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VLC envelope_quant_index[13];
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VLC sqvh[7]; //scalar quantization
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/* generatable tables and related variables */
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int gain_size_factor;
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float gain_table[23];
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/* data buffers */
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uint8_t* decoded_bytes_buffer;
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DECLARE_ALIGNED_16(float,mono_mdct_output[2048]);
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float decode_buffer_1[1024];
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float decode_buffer_2[1024];
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float decode_buffer_0[1060]; /* static allocation for joint decode */
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const float *cplscales[5];
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int num_subpackets;
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COOKSubpacket subpacket[MAX_SUBPACKETS];
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} COOKContext;
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static float pow2tab[127];
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static float rootpow2tab[127];
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/* debug functions */
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#ifdef COOKDEBUG
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static void dump_float_table(float* table, int size, int delimiter) {
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int i=0;
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av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i);
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for (i=0 ; i<size ; i++) {
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av_log(NULL, AV_LOG_ERROR, "%5.1f, ", table[i]);
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if ((i+1)%delimiter == 0) av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i+1);
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}
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}
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static void dump_int_table(int* table, int size, int delimiter) {
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int i=0;
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av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i);
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for (i=0 ; i<size ; i++) {
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av_log(NULL, AV_LOG_ERROR, "%d, ", table[i]);
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if ((i+1)%delimiter == 0) av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i+1);
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}
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}
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static void dump_short_table(short* table, int size, int delimiter) {
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int i=0;
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av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i);
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for (i=0 ; i<size ; i++) {
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av_log(NULL, AV_LOG_ERROR, "%d, ", table[i]);
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if ((i+1)%delimiter == 0) av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i+1);
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}
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}
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#endif
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/*************** init functions ***************/
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/* table generator */
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static av_cold void init_pow2table(void){
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int i;
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for (i=-63 ; i<64 ; i++){
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pow2tab[63+i]= pow(2, i);
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rootpow2tab[63+i]=sqrt(pow(2, i));
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}
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}
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/* table generator */
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static av_cold void init_gain_table(COOKContext *q) {
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int i;
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q->gain_size_factor = q->samples_per_channel/8;
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for (i=0 ; i<23 ; i++) {
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q->gain_table[i] = pow(pow2tab[i+52] ,
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(1.0/(double)q->gain_size_factor));
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}
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}
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static av_cold int init_cook_vlc_tables(COOKContext *q) {
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int i, result;
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result = 0;
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for (i=0 ; i<13 ; i++) {
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result |= init_vlc (&q->envelope_quant_index[i], 9, 24,
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envelope_quant_index_huffbits[i], 1, 1,
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envelope_quant_index_huffcodes[i], 2, 2, 0);
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}
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av_log(q->avctx,AV_LOG_DEBUG,"sqvh VLC init\n");
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for (i=0 ; i<7 ; i++) {
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result |= init_vlc (&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i],
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cvh_huffbits[i], 1, 1,
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cvh_huffcodes[i], 2, 2, 0);
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}
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for(i=0;i<q->num_subpackets;i++){
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if (q->subpacket[i].joint_stereo==1){
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result |= init_vlc (&q->subpacket[i].ccpl, 6, (1<<q->subpacket[i].js_vlc_bits)-1,
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ccpl_huffbits[q->subpacket[i].js_vlc_bits-2], 1, 1,
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ccpl_huffcodes[q->subpacket[i].js_vlc_bits-2], 2, 2, 0);
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av_log(q->avctx,AV_LOG_DEBUG,"subpacket %i Joint-stereo VLC used.\n",i);
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}
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}
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av_log(q->avctx,AV_LOG_DEBUG,"VLC tables initialized.\n");
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return result;
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}
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static av_cold int init_cook_mlt(COOKContext *q) {
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int j;
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int mlt_size = q->samples_per_channel;
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if ((q->mlt_window = av_malloc(sizeof(float)*mlt_size)) == 0)
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return -1;
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/* Initialize the MLT window: simple sine window. */
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ff_sine_window_init(q->mlt_window, mlt_size);
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for(j=0 ; j<mlt_size ; j++)
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q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
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/* Initialize the MDCT. */
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if (ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size)+1, 1, 1.0)) {
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av_free(q->mlt_window);
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return -1;
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}
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av_log(q->avctx,AV_LOG_DEBUG,"MDCT initialized, order = %d.\n",
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av_log2(mlt_size)+1);
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return 0;
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}
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static const float *maybe_reformat_buffer32 (COOKContext *q, const float *ptr, int n)
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{
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if (1)
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return ptr;
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}
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static av_cold void init_cplscales_table (COOKContext *q) {
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int i;
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for (i=0;i<5;i++)
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q->cplscales[i] = maybe_reformat_buffer32 (q, cplscales[i], (1<<(i+2))-1);
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}
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/*************** init functions end ***********/
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/**
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* Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
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* Why? No idea, some checksum/error detection method maybe.
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*
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* Out buffer size: extra bytes are needed to cope with
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* padding/misalignment.
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* Subpackets passed to the decoder can contain two, consecutive
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* half-subpackets, of identical but arbitrary size.
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* 1234 1234 1234 1234 extraA extraB
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* Case 1: AAAA BBBB 0 0
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* Case 2: AAAA ABBB BB-- 3 3
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* Case 3: AAAA AABB BBBB 2 2
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* Case 4: AAAA AAAB BBBB BB-- 1 5
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*
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* Nice way to waste CPU cycles.
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*
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* @param inbuffer pointer to byte array of indata
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* @param out pointer to byte array of outdata
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* @param bytes number of bytes
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*/
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#define DECODE_BYTES_PAD1(bytes) (3 - ((bytes)+3) % 4)
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#define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
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static inline int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){
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int i, off;
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uint32_t c;
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const uint32_t* buf;
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uint32_t* obuf = (uint32_t*) out;
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/* FIXME: 64 bit platforms would be able to do 64 bits at a time.
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* I'm too lazy though, should be something like
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* for(i=0 ; i<bitamount/64 ; i++)
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* (int64_t)out[i] = 0x37c511f237c511f2^be2me_64(int64_t)in[i]);
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* Buffer alignment needs to be checked. */
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off = (intptr_t)inbuffer & 3;
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buf = (const uint32_t*) (inbuffer - off);
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c = be2me_32((0x37c511f2 >> (off*8)) | (0x37c511f2 << (32-(off*8))));
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bytes += 3 + off;
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for (i = 0; i < bytes/4; i++)
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obuf[i] = c ^ buf[i];
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return off;
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}
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/**
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* Cook uninit
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*/
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static av_cold int cook_decode_close(AVCodecContext *avctx)
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{
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int i;
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COOKContext *q = avctx->priv_data;
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av_log(avctx,AV_LOG_DEBUG, "Deallocating memory.\n");
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/* Free allocated memory buffers. */
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av_free(q->mlt_window);
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av_free(q->decoded_bytes_buffer);
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/* Free the transform. */
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ff_mdct_end(&q->mdct_ctx);
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/* Free the VLC tables. */
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for (i=0 ; i<13 ; i++) {
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free_vlc(&q->envelope_quant_index[i]);
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}
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for (i=0 ; i<7 ; i++) {
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free_vlc(&q->sqvh[i]);
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}
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for (i=0 ; i<q->num_subpackets ; i++) {
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free_vlc(&q->subpacket[i].ccpl);
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}
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av_log(avctx,AV_LOG_DEBUG,"Memory deallocated.\n");
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return 0;
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}
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/**
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* Fill the gain array for the timedomain quantization.
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*
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* @param q pointer to the COOKContext
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* @param gaininfo[9] array of gain indexes
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*/
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static void decode_gain_info(GetBitContext *gb, int *gaininfo)
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{
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int i, n;
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while (get_bits1(gb)) {}
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n = get_bits_count(gb) - 1; //amount of elements*2 to update
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i = 0;
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while (n--) {
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int index = get_bits(gb, 3);
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int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1;
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while (i <= index) gaininfo[i++] = gain;
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}
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while (i <= 8) gaininfo[i++] = 0;
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}
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/**
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* Create the quant index table needed for the envelope.
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*
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* @param q pointer to the COOKContext
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* @param quant_index_table pointer to the array
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*/
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static void decode_envelope(COOKContext *q, COOKSubpacket *p, int* quant_index_table) {
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int i,j, vlc_index;
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quant_index_table[0]= get_bits(&q->gb,6) - 6; //This is used later in categorize
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for (i=1 ; i < p->total_subbands ; i++){
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vlc_index=i;
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if (i >= p->js_subband_start * 2) {
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vlc_index-=p->js_subband_start;
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} else {
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vlc_index/=2;
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if(vlc_index < 1) vlc_index = 1;
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}
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if (vlc_index>13) vlc_index = 13; //the VLC tables >13 are identical to No. 13
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j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index-1].table,
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q->envelope_quant_index[vlc_index-1].bits,2);
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quant_index_table[i] = quant_index_table[i-1] + j - 12; //differential encoding
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}
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}
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/**
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* Calculate the category and category_index vector.
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*
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* @param q pointer to the COOKContext
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* @param quant_index_table pointer to the array
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* @param category pointer to the category array
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* @param category_index pointer to the category_index array
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*/
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static void categorize(COOKContext *q, COOKSubpacket *p, int* quant_index_table,
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int* category, int* category_index){
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int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;
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int exp_index2[102];
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int exp_index1[102];
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int tmp_categorize_array[128*2];
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int tmp_categorize_array1_idx=p->numvector_size;
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int tmp_categorize_array2_idx=p->numvector_size;
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bits_left = p->bits_per_subpacket - get_bits_count(&q->gb);
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if(bits_left > q->samples_per_channel) {
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bits_left = q->samples_per_channel +
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((bits_left - q->samples_per_channel)*5)/8;
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//av_log(q->avctx, AV_LOG_ERROR, "bits_left = %d\n",bits_left);
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}
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memset(&exp_index1,0,102*sizeof(int));
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memset(&exp_index2,0,102*sizeof(int));
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memset(&tmp_categorize_array,0,128*2*sizeof(int));
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bias=-32;
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/* Estimate bias. */
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for (i=32 ; i>0 ; i=i/2){
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num_bits = 0;
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index = 0;
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for (j=p->total_subbands ; j>0 ; j--){
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exp_idx = av_clip((i - quant_index_table[index] + bias) / 2, 0, 7);
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index++;
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num_bits+=expbits_tab[exp_idx];
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}
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if(num_bits >= bits_left - 32){
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bias+=i;
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}
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}
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/* Calculate total number of bits. */
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num_bits=0;
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for (i=0 ; i<p->total_subbands ; i++) {
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exp_idx = av_clip((bias - quant_index_table[i]) / 2, 0, 7);
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num_bits += expbits_tab[exp_idx];
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exp_index1[i] = exp_idx;
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exp_index2[i] = exp_idx;
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}
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tmpbias1 = tmpbias2 = num_bits;
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for (j = 1 ; j < p->numvector_size ; j++) {
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if (tmpbias1 + tmpbias2 > 2*bits_left) { /* ---> */
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int max = -999999;
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index=-1;
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for (i=0 ; i<p->total_subbands ; i++){
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if (exp_index1[i] < 7) {
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v = (-2*exp_index1[i]) - quant_index_table[i] + bias;
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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, ff_random_get_seed());
|
|
|
|
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 = CODEC_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"),
|
|
};
|