Mercurial > libavcodec.hg
view cook.c @ 12105:d6e87496883b libavcodec
ARM: set section to .text in 'function' macro
This ensures code always goes into the .text section and avoids the
need to specify it explicitly after changing sections.
| author | mru |
|---|---|
| date | Wed, 07 Jul 2010 20:09:41 +0000 |
| parents | 313167689ce8 |
| children | 495b13d86191 |
line wrap: on
line source
/* * COOK compatible decoder * Copyright (c) 2003 Sascha Sommer * Copyright (c) 2005 Benjamin Larsson * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * Cook compatible decoder. Bastardization of the G.722.1 standard. * This decoder handles RealNetworks, RealAudio G2 data. * Cook is identified by the codec name cook in RM files. * * To use this decoder, a calling application must supply the extradata * bytes provided from the RM container; 8+ bytes for mono streams and * 16+ for stereo streams (maybe more). * * Codec technicalities (all this assume a buffer length of 1024): * Cook works with several different techniques to achieve its compression. * In the timedomain the buffer is divided into 8 pieces and quantized. If * two neighboring pieces have different quantization index a smooth * quantization curve is used to get a smooth overlap between the different * pieces. * To get to the transformdomain Cook uses a modulated lapped transform. * The transform domain has 50 subbands with 20 elements each. This * means only a maximum of 50*20=1000 coefficients are used out of the 1024 * available. */ #include <math.h> #include <stddef.h> #include <stdio.h> #include "libavutil/lfg.h" #include "libavutil/random_seed.h" #include "avcodec.h" #include "get_bits.h" #include "dsputil.h" #include "bytestream.h" #include "fft.h" #include "cookdata.h" /* the different Cook versions */ #define MONO 0x1000001 #define STEREO 0x1000002 #define JOINT_STEREO 0x1000003 #define MC_COOK 0x2000000 //multichannel Cook, not supported #define SUBBAND_SIZE 20 #define MAX_SUBPACKETS 5 //#define COOKDEBUG typedef struct { int *now; int *previous; } cook_gains; typedef struct { int ch_idx; int size; int num_channels; int cookversion; int samples_per_frame; int subbands; int js_subband_start; int js_vlc_bits; int samples_per_channel; int log2_numvector_size; unsigned int channel_mask; VLC ccpl; ///< channel coupling int joint_stereo; int bits_per_subpacket; int bits_per_subpdiv; int total_subbands; int numvector_size; ///< 1 << log2_numvector_size; float mono_previous_buffer1[1024]; float mono_previous_buffer2[1024]; /** gain buffers */ cook_gains gains1; cook_gains gains2; int gain_1[9]; int gain_2[9]; int gain_3[9]; int gain_4[9]; } COOKSubpacket; typedef struct cook { /* * The following 5 functions provide the lowlevel arithmetic on * the internal audio buffers. */ void (* scalar_dequant)(struct cook *q, int index, int quant_index, int* subband_coef_index, int* subband_coef_sign, float* mlt_p); void (* decouple) (struct cook *q, COOKSubpacket *p, int subband, float f1, float f2, float *decode_buffer, float *mlt_buffer1, float *mlt_buffer2); void (* imlt_window) (struct cook *q, float *buffer1, cook_gains *gains_ptr, float *previous_buffer); void (* interpolate) (struct cook *q, float* buffer, int gain_index, int gain_index_next); void (* saturate_output) (struct cook *q, int chan, int16_t *out); AVCodecContext* avctx; GetBitContext gb; /* stream data */ int nb_channels; int bit_rate; int sample_rate; int num_vectors; int samples_per_channel; /* states */ AVLFG random_state; /* transform data */ FFTContext mdct_ctx; float* mlt_window; /* VLC data */ VLC envelope_quant_index[13]; VLC sqvh[7]; //scalar quantization /* generatable tables and related variables */ int gain_size_factor; float gain_table[23]; /* data buffers */ uint8_t* decoded_bytes_buffer; DECLARE_ALIGNED(16, float,mono_mdct_output)[2048]; float decode_buffer_1[1024]; float decode_buffer_2[1024]; float decode_buffer_0[1060]; /* static allocation for joint decode */ const float *cplscales[5]; int num_subpackets; COOKSubpacket subpacket[MAX_SUBPACKETS]; } COOKContext; static float pow2tab[127]; static float rootpow2tab[127]; /* debug functions */ #ifdef COOKDEBUG static void dump_float_table(float* table, int size, int delimiter) { int i=0; av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i); for (i=0 ; i<size ; i++) { av_log(NULL, AV_LOG_ERROR, "%5.1f, ", table[i]); if ((i+1)%delimiter == 0) av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i+1); } } static void dump_int_table(int* table, int size, int delimiter) { int i=0; av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i); for (i=0 ; i<size ; i++) { av_log(NULL, AV_LOG_ERROR, "%d, ", table[i]); if ((i+1)%delimiter == 0) av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i+1); } } static void dump_short_table(short* table, int size, int delimiter) { int i=0; av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i); for (i=0 ; i<size ; i++) { av_log(NULL, AV_LOG_ERROR, "%d, ", table[i]); if ((i+1)%delimiter == 0) av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i+1); } } #endif /*************** init functions ***************/ /* table generator */ static av_cold void init_pow2table(void){ int i; for (i=-63 ; i<64 ; i++){ pow2tab[63+i]= pow(2, i); rootpow2tab[63+i]=sqrt(pow(2, i)); } } /* table generator */ static av_cold void init_gain_table(COOKContext *q) { int i; q->gain_size_factor = q->samples_per_channel/8; for (i=0 ; i<23 ; i++) { q->gain_table[i] = pow(pow2tab[i+52] , (1.0/(double)q->gain_size_factor)); } } static av_cold int init_cook_vlc_tables(COOKContext *q) { int i, result; result = 0; for (i=0 ; i<13 ; i++) { result |= init_vlc (&q->envelope_quant_index[i], 9, 24, envelope_quant_index_huffbits[i], 1, 1, envelope_quant_index_huffcodes[i], 2, 2, 0); } av_log(q->avctx,AV_LOG_DEBUG,"sqvh VLC init\n"); for (i=0 ; i<7 ; i++) { result |= init_vlc (&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i], cvh_huffbits[i], 1, 1, cvh_huffcodes[i], 2, 2, 0); } for(i=0;i<q->num_subpackets;i++){ if (q->subpacket[i].joint_stereo==1){ result |= init_vlc (&q->subpacket[i].ccpl, 6, (1<<q->subpacket[i].js_vlc_bits)-1, ccpl_huffbits[q->subpacket[i].js_vlc_bits-2], 1, 1, ccpl_huffcodes[q->subpacket[i].js_vlc_bits-2], 2, 2, 0); av_log(q->avctx,AV_LOG_DEBUG,"subpacket %i Joint-stereo VLC used.\n",i); } } av_log(q->avctx,AV_LOG_DEBUG,"VLC tables initialized.\n"); return result; } static av_cold int init_cook_mlt(COOKContext *q) { int j; int mlt_size = q->samples_per_channel; if ((q->mlt_window = av_malloc(sizeof(float)*mlt_size)) == 0) return -1; /* Initialize the MLT window: simple sine window. */ ff_sine_window_init(q->mlt_window, mlt_size); for(j=0 ; j<mlt_size ; j++) q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel); /* Initialize the MDCT. */ if (ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size)+1, 1, 1.0)) { av_free(q->mlt_window); return -1; } av_log(q->avctx,AV_LOG_DEBUG,"MDCT initialized, order = %d.\n", av_log2(mlt_size)+1); return 0; } static const float *maybe_reformat_buffer32 (COOKContext *q, const float *ptr, int n) { if (1) return ptr; } static av_cold void init_cplscales_table (COOKContext *q) { int i; for (i=0;i<5;i++) q->cplscales[i] = maybe_reformat_buffer32 (q, cplscales[i], (1<<(i+2))-1); } /*************** init functions end ***********/ #define DECODE_BYTES_PAD1(bytes) (3 - ((bytes)+3) % 4) #define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes))) /** * Cook indata decoding, every 32 bits are XORed with 0x37c511f2. * Why? No idea, some checksum/error detection method maybe. * * Out buffer size: extra bytes are needed to cope with * padding/misalignment. * Subpackets passed to the decoder can contain two, consecutive * half-subpackets, of identical but arbitrary size. * 1234 1234 1234 1234 extraA extraB * Case 1: AAAA BBBB 0 0 * Case 2: AAAA ABBB BB-- 3 3 * Case 3: AAAA AABB BBBB 2 2 * Case 4: AAAA AAAB BBBB BB-- 1 5 * * Nice way to waste CPU cycles. * * @param inbuffer pointer to byte array of indata * @param out pointer to byte array of outdata * @param bytes number of bytes */ static inline int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){ int i, off; uint32_t c; const uint32_t* buf; uint32_t* obuf = (uint32_t*) out; /* FIXME: 64 bit platforms would be able to do 64 bits at a time. * I'm too lazy though, should be something like * for(i=0 ; i<bitamount/64 ; i++) * (int64_t)out[i] = 0x37c511f237c511f2^be2me_64(int64_t)in[i]); * Buffer alignment needs to be checked. */ off = (intptr_t)inbuffer & 3; buf = (const uint32_t*) (inbuffer - off); c = be2me_32((0x37c511f2 >> (off*8)) | (0x37c511f2 << (32-(off*8)))); bytes += 3 + off; for (i = 0; i < bytes/4; i++) obuf[i] = c ^ buf[i]; return off; } /** * Cook uninit */ static av_cold int cook_decode_close(AVCodecContext *avctx) { int i; COOKContext *q = avctx->priv_data; av_log(avctx,AV_LOG_DEBUG, "Deallocating memory.\n"); /* Free allocated memory buffers. */ av_free(q->mlt_window); av_free(q->decoded_bytes_buffer); /* Free the transform. */ ff_mdct_end(&q->mdct_ctx); /* Free the VLC tables. */ for (i=0 ; i<13 ; i++) { free_vlc(&q->envelope_quant_index[i]); } for (i=0 ; i<7 ; i++) { free_vlc(&q->sqvh[i]); } for (i=0 ; i<q->num_subpackets ; i++) { free_vlc(&q->subpacket[i].ccpl); } av_log(avctx,AV_LOG_DEBUG,"Memory deallocated.\n"); return 0; } /** * Fill the gain array for the timedomain quantization. * * @param gb pointer to the GetBitContext * @param gaininfo array of gain indexes */ static void decode_gain_info(GetBitContext *gb, int *gaininfo) { int i, n; while (get_bits1(gb)) {} n = get_bits_count(gb) - 1; //amount of elements*2 to update i = 0; while (n--) { int index = get_bits(gb, 3); int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1; while (i <= index) gaininfo[i++] = gain; } while (i <= 8) gaininfo[i++] = 0; } /** * Create the quant index table needed for the envelope. * * @param q pointer to the COOKContext * @param quant_index_table pointer to the array */ static void decode_envelope(COOKContext *q, COOKSubpacket *p, int* quant_index_table) { int i,j, vlc_index; quant_index_table[0]= get_bits(&q->gb,6) - 6; //This is used later in categorize for (i=1 ; i < p->total_subbands ; i++){ vlc_index=i; if (i >= p->js_subband_start * 2) { vlc_index-=p->js_subband_start; } else { vlc_index/=2; if(vlc_index < 1) vlc_index = 1; } if (vlc_index>13) vlc_index = 13; //the VLC tables >13 are identical to No. 13 j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index-1].table, q->envelope_quant_index[vlc_index-1].bits,2); quant_index_table[i] = quant_index_table[i-1] + j - 12; //differential encoding } } /** * Calculate the category and category_index vector. * * @param q pointer to the COOKContext * @param quant_index_table pointer to the array * @param category pointer to the category array * @param category_index pointer to the category_index array */ static void categorize(COOKContext *q, COOKSubpacket *p, int* quant_index_table, int* category, int* category_index){ int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j; int exp_index2[102]; int exp_index1[102]; int tmp_categorize_array[128*2]; int tmp_categorize_array1_idx=p->numvector_size; int tmp_categorize_array2_idx=p->numvector_size; bits_left = p->bits_per_subpacket - get_bits_count(&q->gb); if(bits_left > q->samples_per_channel) { bits_left = q->samples_per_channel + ((bits_left - q->samples_per_channel)*5)/8; //av_log(q->avctx, AV_LOG_ERROR, "bits_left = %d\n",bits_left); } memset(&exp_index1,0,102*sizeof(int)); memset(&exp_index2,0,102*sizeof(int)); memset(&tmp_categorize_array,0,128*2*sizeof(int)); bias=-32; /* Estimate bias. */ for (i=32 ; i>0 ; i=i/2){ num_bits = 0; index = 0; for (j=p->total_subbands ; j>0 ; j--){ exp_idx = av_clip((i - quant_index_table[index] + bias) / 2, 0, 7); index++; num_bits+=expbits_tab[exp_idx]; } if(num_bits >= bits_left - 32){ bias+=i; } } /* Calculate total number of bits. */ num_bits=0; for (i=0 ; i<p->total_subbands ; i++) { exp_idx = av_clip((bias - quant_index_table[i]) / 2, 0, 7); num_bits += expbits_tab[exp_idx]; exp_index1[i] = exp_idx; exp_index2[i] = exp_idx; } tmpbias1 = tmpbias2 = num_bits; for (j = 1 ; j < p->numvector_size ; j++) { if (tmpbias1 + tmpbias2 > 2*bits_left) { /* ---> */ int max = -999999; index=-1; for (i=0 ; i<p->total_subbands ; i++){ if (exp_index1[i] < 7) { v = (-2*exp_index1[i]) - quant_index_table[i] + bias; if ( v >= max) { max = v; index = i; } } } if(index==-1)break; tmp_categorize_array[tmp_categorize_array1_idx++] = index; tmpbias1 -= expbits_tab[exp_index1[index]] - expbits_tab[exp_index1[index]+1]; ++exp_index1[index]; } else { /* <--- */ int min = 999999; index=-1; for (i=0 ; i<p->total_subbands ; i++){ if(exp_index2[i] > 0){ v = (-2*exp_index2[i])-quant_index_table[i]+bias; if ( v < min) { min = v; index = i; } } } if(index == -1)break; tmp_categorize_array[--tmp_categorize_array2_idx] = index; tmpbias2 -= expbits_tab[exp_index2[index]] - expbits_tab[exp_index2[index]-1]; --exp_index2[index]; } } for(i=0 ; i<p->total_subbands ; i++) category[i] = exp_index2[i]; for(i=0 ; i<p->numvector_size-1 ; i++) category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++]; } /** * Expand the category vector. * * @param q pointer to the COOKContext * @param category pointer to the category array * @param category_index pointer to the category_index array */ static inline void expand_category(COOKContext *q, int* category, int* category_index){ int i; for(i=0 ; i<q->num_vectors ; i++){ ++category[category_index[i]]; } } /** * The real requantization of the mltcoefs * * @param q pointer to the COOKContext * @param index index * @param quant_index quantisation index * @param subband_coef_index array of indexes to quant_centroid_tab * @param subband_coef_sign signs of coefficients * @param mlt_p pointer into the mlt buffer */ static void scalar_dequant_float(COOKContext *q, int index, int quant_index, int* subband_coef_index, int* subband_coef_sign, float* mlt_p){ int i; float f1; for(i=0 ; i<SUBBAND_SIZE ; i++) { if (subband_coef_index[i]) { f1 = quant_centroid_tab[index][subband_coef_index[i]]; if (subband_coef_sign[i]) f1 = -f1; } else { /* noise coding if subband_coef_index[i] == 0 */ f1 = dither_tab[index]; if (av_lfg_get(&q->random_state) < 0x80000000) f1 = -f1; } mlt_p[i] = f1 * rootpow2tab[quant_index+63]; } } /** * Unpack the subband_coef_index and subband_coef_sign vectors. * * @param q pointer to the COOKContext * @param category pointer to the category array * @param subband_coef_index array of indexes to quant_centroid_tab * @param subband_coef_sign signs of coefficients */ static int unpack_SQVH(COOKContext *q, COOKSubpacket *p, int category, int* subband_coef_index, int* subband_coef_sign) { int i,j; int vlc, vd ,tmp, result; vd = vd_tab[category]; result = 0; for(i=0 ; i<vpr_tab[category] ; i++){ vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3); if (p->bits_per_subpacket < get_bits_count(&q->gb)){ vlc = 0; result = 1; } for(j=vd-1 ; j>=0 ; j--){ tmp = (vlc * invradix_tab[category])/0x100000; subband_coef_index[vd*i+j] = vlc - tmp * (kmax_tab[category]+1); vlc = tmp; } for(j=0 ; j<vd ; j++){ if (subband_coef_index[i*vd + j]) { if(get_bits_count(&q->gb) < p->bits_per_subpacket){ subband_coef_sign[i*vd+j] = get_bits1(&q->gb); } else { result=1; subband_coef_sign[i*vd+j]=0; } } else { subband_coef_sign[i*vd+j]=0; } } } return result; } /** * Fill the mlt_buffer with mlt coefficients. * * @param q pointer to the COOKContext * @param category pointer to the category array * @param quant_index_table pointer to the array * @param mlt_buffer pointer to mlt coefficients */ static void decode_vectors(COOKContext* q, COOKSubpacket* p, int* category, int *quant_index_table, float* mlt_buffer){ /* A zero in this table means that the subband coefficient is random noise coded. */ int subband_coef_index[SUBBAND_SIZE]; /* A zero in this table means that the subband coefficient is a positive multiplicator. */ int subband_coef_sign[SUBBAND_SIZE]; int band, j; int index=0; for(band=0 ; band<p->total_subbands ; band++){ index = category[band]; if(category[band] < 7){ if(unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)){ index=7; for(j=0 ; j<p->total_subbands ; j++) category[band+j]=7; } } if(index>=7) { memset(subband_coef_index, 0, sizeof(subband_coef_index)); memset(subband_coef_sign, 0, sizeof(subband_coef_sign)); } q->scalar_dequant(q, index, quant_index_table[band], subband_coef_index, subband_coef_sign, &mlt_buffer[band * SUBBAND_SIZE]); } if(p->total_subbands*SUBBAND_SIZE >= q->samples_per_channel){ return; } /* FIXME: should this be removed, or moved into loop above? */ } /** * function for decoding mono data * * @param q pointer to the COOKContext * @param mlt_buffer pointer to mlt coefficients */ static void mono_decode(COOKContext *q, COOKSubpacket *p, float* mlt_buffer) { int category_index[128]; int quant_index_table[102]; int category[128]; memset(&category, 0, 128*sizeof(int)); memset(&category_index, 0, 128*sizeof(int)); decode_envelope(q, p, quant_index_table); q->num_vectors = get_bits(&q->gb,p->log2_numvector_size); categorize(q, p, quant_index_table, category, category_index); expand_category(q, category, category_index); decode_vectors(q, p, category, quant_index_table, mlt_buffer); } /** * the actual requantization of the timedomain samples * * @param q pointer to the COOKContext * @param buffer pointer to the timedomain buffer * @param gain_index index for the block multiplier * @param gain_index_next index for the next block multiplier */ static void interpolate_float(COOKContext *q, float* buffer, int gain_index, int gain_index_next){ int i; float fc1, fc2; fc1 = pow2tab[gain_index+63]; if(gain_index == gain_index_next){ //static gain for(i=0 ; i<q->gain_size_factor ; i++){ buffer[i]*=fc1; } return; } else { //smooth gain fc2 = q->gain_table[11 + (gain_index_next-gain_index)]; for(i=0 ; i<q->gain_size_factor ; i++){ buffer[i]*=fc1; fc1*=fc2; } return; } } /** * Apply transform window, overlap buffers. * * @param q pointer to the COOKContext * @param buffer1 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 gains_ptr array of current/prev gain pointers */ static inline void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p, const uint8_t *inbuffer, cook_gains *gains_ptr) { int offset; offset = decode_bytes(inbuffer, q->decoded_bytes_buffer, p->bits_per_subpacket/8); init_get_bits(&q->gb, q->decoded_bytes_buffer + offset, p->bits_per_subpacket); decode_gain_info(&q->gb, gains_ptr->now); /* Swap current and previous gains */ FFSWAP(int *, gains_ptr->now, gains_ptr->previous); } /** * Saturate the output signal to signed 16bit integers. * * @param q pointer to the COOKContext * @param chan channel to saturate * @param out pointer to the output vector */ static void saturate_output_float (COOKContext *q, int chan, int16_t *out) { int j; float *output = q->mono_mdct_output + q->samples_per_channel; /* Clip and convert floats to 16 bits. */ for (j = 0; j < q->samples_per_channel; j++) { out[chan + q->nb_channels * j] = av_clip_int16(lrintf(output[j])); } } /** * Final part of subpacket decoding: * Apply modulated lapped transform, gain compensation, * clip and convert to integer. * * @param q pointer to the COOKContext * @param decode_buffer pointer to the mlt coefficients * @param gains 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 outbuffer pointer to the outbuffer */ static void decode_subpacket(COOKContext *q, COOKSubpacket* p, const uint8_t *inbuffer, int16_t *outbuffer) { int sub_packet_size = p->size; /* packet dump */ // for (i=0 ; i<sub_packet_size ; i++) { // av_log(q->avctx, AV_LOG_ERROR, "%02x", inbuffer[i]); // } // av_log(q->avctx, AV_LOG_ERROR, "\n"); memset(q->decode_buffer_1,0,sizeof(q->decode_buffer_1)); decode_bytes_and_gain(q, p, inbuffer, &p->gains1); if (p->joint_stereo) { joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2); } else { mono_decode(q, p, q->decode_buffer_1); if (p->num_channels == 2) { decode_bytes_and_gain(q, p, inbuffer + sub_packet_size/2, &p->gains2); mono_decode(q, p, q->decode_buffer_2); } } mlt_compensate_output(q, q->decode_buffer_1, &p->gains1, p->mono_previous_buffer1, outbuffer, p->ch_idx); if (p->num_channels == 2) { if (p->joint_stereo) { mlt_compensate_output(q, q->decode_buffer_2, &p->gains1, p->mono_previous_buffer2, outbuffer, p->ch_idx + 1); } else { mlt_compensate_output(q, q->decode_buffer_2, &p->gains2, p->mono_previous_buffer2, outbuffer, p->ch_idx + 1); } } } /** * Cook frame decoding * * @param avctx pointer to the AVCodecContext */ static int cook_decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; COOKContext *q = avctx->priv_data; int i; int offset = 0; int chidx = 0; if (buf_size < avctx->block_align) return buf_size; /* estimate subpacket sizes */ q->subpacket[0].size = avctx->block_align; for(i=1;i<q->num_subpackets;i++){ q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i]; q->subpacket[0].size -= q->subpacket[i].size + 1; if (q->subpacket[0].size < 0) { av_log(avctx,AV_LOG_DEBUG,"frame subpacket size total > avctx->block_align!\n"); return -1; } } /* decode supbackets */ *data_size = 0; for(i=0;i<q->num_subpackets;i++){ q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size*8)>>q->subpacket[i].bits_per_subpdiv; q->subpacket[i].ch_idx = chidx; av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] size %i js %i %i block_align %i\n",i,q->subpacket[i].size,q->subpacket[i].joint_stereo,offset,avctx->block_align); decode_subpacket(q, &q->subpacket[i], buf + offset, (int16_t*)data); offset += q->subpacket[i].size; chidx += q->subpacket[i].num_channels; av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] %i %i\n",i,q->subpacket[i].size * 8,get_bits_count(&q->gb)); } *data_size = sizeof(int16_t) * q->nb_channels * q->samples_per_channel; /* Discard the first two frames: no valid audio. */ if (avctx->frame_number < 2) *data_size = 0; return avctx->block_align; } #ifdef COOKDEBUG static void dump_cook_context(COOKContext *q) { //int i=0; #define PRINT(a,b) av_log(q->avctx,AV_LOG_ERROR," %s = %d\n", a, b); av_log(q->avctx,AV_LOG_ERROR,"COOKextradata\n"); av_log(q->avctx,AV_LOG_ERROR,"cookversion=%x\n",q->subpacket[0].cookversion); if (q->subpacket[0].cookversion > STEREO) { PRINT("js_subband_start",q->subpacket[0].js_subband_start); PRINT("js_vlc_bits",q->subpacket[0].js_vlc_bits); } av_log(q->avctx,AV_LOG_ERROR,"COOKContext\n"); PRINT("nb_channels",q->nb_channels); PRINT("bit_rate",q->bit_rate); PRINT("sample_rate",q->sample_rate); PRINT("samples_per_channel",q->subpacket[0].samples_per_channel); PRINT("samples_per_frame",q->subpacket[0].samples_per_frame); PRINT("subbands",q->subpacket[0].subbands); PRINT("random_state",q->random_state); PRINT("js_subband_start",q->subpacket[0].js_subband_start); PRINT("log2_numvector_size",q->subpacket[0].log2_numvector_size); PRINT("numvector_size",q->subpacket[0].numvector_size); PRINT("total_subbands",q->subpacket[0].total_subbands); } #endif static av_cold int cook_count_channels(unsigned int mask){ int i; int channels = 0; for(i = 0;i<32;i++){ if(mask & (1<<i)) ++channels; } return channels; } /** * Cook initialization * * @param avctx pointer to the AVCodecContext */ static av_cold int cook_decode_init(AVCodecContext *avctx) { COOKContext *q = avctx->priv_data; const uint8_t *edata_ptr = avctx->extradata; const uint8_t *edata_ptr_end = edata_ptr + avctx->extradata_size; int extradata_size = avctx->extradata_size; int s = 0; unsigned int channel_mask = 0; q->avctx = avctx; /* Take care of the codec specific extradata. */ if (extradata_size <= 0) { av_log(avctx,AV_LOG_ERROR,"Necessary extradata missing!\n"); return -1; } av_log(avctx,AV_LOG_DEBUG,"codecdata_length=%d\n",avctx->extradata_size); /* Take data from the AVCodecContext (RM container). */ q->sample_rate = avctx->sample_rate; q->nb_channels = avctx->channels; q->bit_rate = avctx->bit_rate; /* Initialize RNG. */ av_lfg_init(&q->random_state, 0); while(edata_ptr < edata_ptr_end){ /* 8 for mono, 16 for stereo, ? for multichannel Swap to right endianness so we don't need to care later on. */ if (extradata_size >= 8){ q->subpacket[s].cookversion = bytestream_get_be32(&edata_ptr); q->subpacket[s].samples_per_frame = bytestream_get_be16(&edata_ptr); q->subpacket[s].subbands = bytestream_get_be16(&edata_ptr); extradata_size -= 8; } if (avctx->extradata_size >= 8){ bytestream_get_be32(&edata_ptr); //Unknown unused q->subpacket[s].js_subband_start = bytestream_get_be16(&edata_ptr); q->subpacket[s].js_vlc_bits = bytestream_get_be16(&edata_ptr); extradata_size -= 8; } /* Initialize extradata related variables. */ q->subpacket[s].samples_per_channel = q->subpacket[s].samples_per_frame / q->nb_channels; q->subpacket[s].bits_per_subpacket = avctx->block_align * 8; /* Initialize default data states. */ q->subpacket[s].log2_numvector_size = 5; q->subpacket[s].total_subbands = q->subpacket[s].subbands; q->subpacket[s].num_channels = 1; /* Initialize version-dependent variables */ av_log(avctx,AV_LOG_DEBUG,"subpacket[%i].cookversion=%x\n",s,q->subpacket[s].cookversion); q->subpacket[s].joint_stereo = 0; switch (q->subpacket[s].cookversion) { case MONO: if (q->nb_channels != 1) { av_log(avctx,AV_LOG_ERROR,"Container channels != 1, report sample!\n"); return -1; } av_log(avctx,AV_LOG_DEBUG,"MONO\n"); break; case STEREO: if (q->nb_channels != 1) { q->subpacket[s].bits_per_subpdiv = 1; q->subpacket[s].num_channels = 2; } av_log(avctx,AV_LOG_DEBUG,"STEREO\n"); break; case JOINT_STEREO: if (q->nb_channels != 2) { av_log(avctx,AV_LOG_ERROR,"Container channels != 2, report sample!\n"); return -1; } av_log(avctx,AV_LOG_DEBUG,"JOINT_STEREO\n"); if (avctx->extradata_size >= 16){ q->subpacket[s].total_subbands = q->subpacket[s].subbands + q->subpacket[s].js_subband_start; q->subpacket[s].joint_stereo = 1; q->subpacket[s].num_channels = 2; } if (q->subpacket[s].samples_per_channel > 256) { q->subpacket[s].log2_numvector_size = 6; } if (q->subpacket[s].samples_per_channel > 512) { q->subpacket[s].log2_numvector_size = 7; } break; case MC_COOK: av_log(avctx,AV_LOG_DEBUG,"MULTI_CHANNEL\n"); if(extradata_size >= 4) channel_mask |= q->subpacket[s].channel_mask = bytestream_get_be32(&edata_ptr); if(cook_count_channels(q->subpacket[s].channel_mask) > 1){ q->subpacket[s].total_subbands = q->subpacket[s].subbands + q->subpacket[s].js_subband_start; q->subpacket[s].joint_stereo = 1; q->subpacket[s].num_channels = 2; q->subpacket[s].samples_per_channel = q->subpacket[s].samples_per_frame >> 1; if (q->subpacket[s].samples_per_channel > 256) { q->subpacket[s].log2_numvector_size = 6; } if (q->subpacket[s].samples_per_channel > 512) { q->subpacket[s].log2_numvector_size = 7; } }else q->subpacket[s].samples_per_channel = q->subpacket[s].samples_per_frame; break; default: av_log(avctx,AV_LOG_ERROR,"Unknown Cook version, report sample!\n"); return -1; break; } if(s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) { av_log(avctx,AV_LOG_ERROR,"different number of samples per channel!\n"); return -1; } else q->samples_per_channel = q->subpacket[0].samples_per_channel; /* Initialize variable relations */ q->subpacket[s].numvector_size = (1 << q->subpacket[s].log2_numvector_size); /* Try to catch some obviously faulty streams, othervise it might be exploitable */ if (q->subpacket[s].total_subbands > 53) { av_log(avctx,AV_LOG_ERROR,"total_subbands > 53, report sample!\n"); return -1; } if ((q->subpacket[s].js_vlc_bits > 6) || (q->subpacket[s].js_vlc_bits < 0)) { av_log(avctx,AV_LOG_ERROR,"js_vlc_bits = %d, only >= 0 and <= 6 allowed!\n",q->subpacket[s].js_vlc_bits); return -1; } if (q->subpacket[s].subbands > 50) { av_log(avctx,AV_LOG_ERROR,"subbands > 50, report sample!\n"); return -1; } q->subpacket[s].gains1.now = q->subpacket[s].gain_1; q->subpacket[s].gains1.previous = q->subpacket[s].gain_2; q->subpacket[s].gains2.now = q->subpacket[s].gain_3; q->subpacket[s].gains2.previous = q->subpacket[s].gain_4; q->num_subpackets++; s++; if (s > MAX_SUBPACKETS) { av_log(avctx,AV_LOG_ERROR,"Too many subpackets > 5, report file!\n"); return -1; } } /* Generate tables */ init_pow2table(); init_gain_table(q); init_cplscales_table(q); if (init_cook_vlc_tables(q) != 0) return -1; if(avctx->block_align >= UINT_MAX/2) return -1; /* Pad the databuffer with: DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(), FF_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */ q->decoded_bytes_buffer = av_mallocz(avctx->block_align + DECODE_BYTES_PAD1(avctx->block_align) + FF_INPUT_BUFFER_PADDING_SIZE); if (q->decoded_bytes_buffer == NULL) return -1; /* Initialize transform. */ if ( init_cook_mlt(q) != 0 ) return -1; /* Initialize COOK signal arithmetic handling */ if (1) { q->scalar_dequant = scalar_dequant_float; q->decouple = decouple_float; q->imlt_window = imlt_window_float; q->interpolate = interpolate_float; q->saturate_output = saturate_output_float; } /* Try to catch some obviously faulty streams, othervise it might be exploitable */ if ((q->samples_per_channel == 256) || (q->samples_per_channel == 512) || (q->samples_per_channel == 1024)) { } else { av_log(avctx,AV_LOG_ERROR,"unknown amount of samples_per_channel = %d, report sample!\n",q->samples_per_channel); return -1; } avctx->sample_fmt = SAMPLE_FMT_S16; if (channel_mask) avctx->channel_layout = channel_mask; else avctx->channel_layout = (avctx->channels==2) ? CH_LAYOUT_STEREO : CH_LAYOUT_MONO; #ifdef COOKDEBUG dump_cook_context(q); #endif return 0; } AVCodec cook_decoder = { .name = "cook", .type = AVMEDIA_TYPE_AUDIO, .id = CODEC_ID_COOK, .priv_data_size = sizeof(COOKContext), .init = cook_decode_init, .close = cook_decode_close, .decode = cook_decode_frame, .long_name = NULL_IF_CONFIG_SMALL("COOK"), };