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Diffstat (limited to 'contrib/ffmpeg/libavcodec/cook.c')
| -rw-r--r-- | contrib/ffmpeg/libavcodec/cook.c | 1147 |
1 files changed, 1147 insertions, 0 deletions
diff --git a/contrib/ffmpeg/libavcodec/cook.c b/contrib/ffmpeg/libavcodec/cook.c new file mode 100644 index 000000000..32b1081cc --- /dev/null +++ b/contrib/ffmpeg/libavcodec/cook.c @@ -0,0 +1,1147 @@ +/* + * 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.c + * Cook compatible decoder. + * This decoder handles RealNetworks, RealAudio G2 data. + * Cook is identified by the codec name cook in RM files. + * + * To use this decoder, a calling application must supply the extradata + * bytes provided from the RM container; 8+ bytes for mono streams and + * 16+ for stereo streams (maybe more). + * + * Codec technicalities (all this assume a buffer length of 1024): + * Cook works with several different techniques to achieve its compression. + * In the timedomain the buffer is divided into 8 pieces and quantized. If + * two neighboring pieces have different quantization index a smooth + * quantization curve is used to get a smooth overlap between the different + * pieces. + * To get to the transformdomain Cook uses a modulated lapped transform. + * The transform domain has 50 subbands with 20 elements each. This + * means only a maximum of 50*20=1000 coefficients are used out of the 1024 + * available. + */ + +#include <math.h> +#include <stddef.h> +#include <stdio.h> + +#include "avcodec.h" +#include "bitstream.h" +#include "dsputil.h" +#include "common.h" +#include "bytestream.h" +#include "random.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 COOKDEBUG + +typedef struct { + int *now; + int *previous; +} cook_gains; + +typedef struct { + GetBitContext gb; + /* stream data */ + int nb_channels; + int joint_stereo; + int bit_rate; + int sample_rate; + int samples_per_channel; + int samples_per_frame; + int subbands; + int log2_numvector_size; + int numvector_size; //1 << log2_numvector_size; + int js_subband_start; + int total_subbands; + int num_vectors; + int bits_per_subpacket; + int cookversion; + /* states */ + AVRandomState random_state; + + /* transform data */ + MDCTContext mdct_ctx; + DECLARE_ALIGNED_16(FFTSample, mdct_tmp[1024]); /* temporary storage for imlt */ + float* mlt_window; + + /* gain buffers */ + cook_gains gains1; + cook_gains gains2; + int gain_1[9]; + int gain_2[9]; + int gain_3[9]; + int gain_4[9]; + + /* VLC data */ + int js_vlc_bits; + VLC envelope_quant_index[13]; + VLC sqvh[7]; //scalar quantization + VLC ccpl; //channel coupling + + /* generatable tables and related variables */ + int gain_size_factor; + float gain_table[23]; + float pow2tab[127]; + float rootpow2tab[127]; + + /* data buffers */ + + uint8_t* decoded_bytes_buffer; + DECLARE_ALIGNED_16(float,mono_mdct_output[2048]); + float mono_previous_buffer1[1024]; + float mono_previous_buffer2[1024]; + float decode_buffer_1[1024]; + float decode_buffer_2[1024]; +} COOKContext; + +/* debug functions */ + +#ifdef COOKDEBUG +static void dump_float_table(float* table, int size, int delimiter) { + int i=0; + av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i); + for (i=0 ; i<size ; i++) { + av_log(NULL, AV_LOG_ERROR, "%5.1f, ", table[i]); + if ((i+1)%delimiter == 0) av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i+1); + } +} + +static void dump_int_table(int* table, int size, int delimiter) { + int i=0; + av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i); + for (i=0 ; i<size ; i++) { + av_log(NULL, AV_LOG_ERROR, "%d, ", table[i]); + if ((i+1)%delimiter == 0) av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i+1); + } +} + +static void dump_short_table(short* table, int size, int delimiter) { + int i=0; + av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i); + for (i=0 ; i<size ; i++) { + av_log(NULL, AV_LOG_ERROR, "%d, ", table[i]); + if ((i+1)%delimiter == 0) av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i+1); + } +} + +#endif + +/*************** init functions ***************/ + +/* table generator */ +static void init_pow2table(COOKContext *q){ + int i; + q->pow2tab[63] = 1.0; + for (i=1 ; i<64 ; i++){ + q->pow2tab[63+i]=(float)((uint64_t)1<<i); + q->pow2tab[63-i]=1.0/(float)((uint64_t)1<<i); + } +} + +/* table generator */ +static void init_rootpow2table(COOKContext *q){ + int i; + q->rootpow2tab[63] = 1.0; + for (i=1 ; i<64 ; i++){ + q->rootpow2tab[63+i]=sqrt((float)((uint64_t)1<<i)); + q->rootpow2tab[63-i]=sqrt(1.0/(float)((uint64_t)1<<i)); + } +} + +/* table generator */ +static void init_gain_table(COOKContext *q) { + int i; + q->gain_size_factor = q->samples_per_channel/8; + for (i=0 ; i<23 ; i++) { + q->gain_table[i] = pow((double)q->pow2tab[i+52] , + (1.0/(double)q->gain_size_factor)); + } +} + + +static int init_cook_vlc_tables(COOKContext *q) { + int i, result; + + result = 0; + for (i=0 ; i<13 ; i++) { + result &= init_vlc (&q->envelope_quant_index[i], 9, 24, + envelope_quant_index_huffbits[i], 1, 1, + envelope_quant_index_huffcodes[i], 2, 2, 0); + } + av_log(NULL,AV_LOG_DEBUG,"sqvh VLC init\n"); + for (i=0 ; i<7 ; i++) { + result &= init_vlc (&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i], + cvh_huffbits[i], 1, 1, + cvh_huffcodes[i], 2, 2, 0); + } + + if (q->nb_channels==2 && q->joint_stereo==1){ + result &= init_vlc (&q->ccpl, 6, (1<<q->js_vlc_bits)-1, + ccpl_huffbits[q->js_vlc_bits-2], 1, 1, + ccpl_huffcodes[q->js_vlc_bits-2], 2, 2, 0); + av_log(NULL,AV_LOG_DEBUG,"Joint-stereo VLC used.\n"); + } + + av_log(NULL,AV_LOG_DEBUG,"VLC tables initialized.\n"); + return result; +} + +static int init_cook_mlt(COOKContext *q) { + int j; + float alpha; + 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. */ + alpha = M_PI / (2.0 * (float)mlt_size); + for(j=0 ; j<mlt_size ; j++) + q->mlt_window[j] = sin((j + 0.5) * alpha) * sqrt(2.0 / q->samples_per_channel); + + /* Initialize the MDCT. */ + if (ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size)+1, 1)) { + av_free(q->mlt_window); + return -1; + } + av_log(NULL,AV_LOG_DEBUG,"MDCT initialized, order = %d.\n", + av_log2(mlt_size)+1); + + return 0; +} + +/*************** init functions end ***********/ + +/** + * Cook indata decoding, every 32 bits are XORed with 0x37c511f2. + * Why? No idea, some checksum/error detection method maybe. + * + * Out buffer size: extra bytes are needed to cope with + * padding/missalignment. + * Subpackets passed to the decoder can contain two, consecutive + * half-subpackets, of identical but arbitrary size. + * 1234 1234 1234 1234 extraA extraB + * Case 1: AAAA BBBB 0 0 + * Case 2: AAAA ABBB BB-- 3 3 + * Case 3: AAAA AABB BBBB 2 2 + * Case 4: AAAA AAAB BBBB BB-- 1 5 + * + * Nice way to waste CPU cycles. + * + * @param inbuffer pointer to byte array of indata + * @param out pointer to byte array of outdata + * @param bytes number of bytes + */ +#define DECODE_BYTES_PAD1(bytes) (3 - ((bytes)+3) % 4) +#define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes))) + +static inline int decode_bytes(uint8_t* inbuffer, uint8_t* out, int bytes){ + int i, off; + uint32_t c; + 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 = (int)((long)inbuffer & 3); + buf = (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 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]); + } + if(q->nb_channels==2 && q->joint_stereo==1 ){ + free_vlc(&q->ccpl); + } + + av_log(NULL,AV_LOG_DEBUG,"Memory deallocated.\n"); + + return 0; +} + +/** + * Fill the gain array for the timedomain quantization. + * + * @param q pointer to the COOKContext + * @param gaininfo[9] array of gain indices + */ + +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, int* quant_index_table) { + int i,j, vlc_index; + int bitbias; + + bitbias = get_bits_count(&q->gb); + quant_index_table[0]= get_bits(&q->gb,6) - 6; //This is used later in categorize + + for (i=1 ; i < q->total_subbands ; i++){ + vlc_index=i; + if (i >= q->js_subband_start * 2) { + vlc_index-=q->js_subband_start; + } else { + vlc_index/=2; + if(vlc_index < 1) vlc_index = 1; + } + if (vlc_index>13) vlc_index = 13; //the VLC tables >13 are identical to No. 13 + + j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index-1].table, + q->envelope_quant_index[vlc_index-1].bits,2); + quant_index_table[i] = quant_index_table[i-1] + j - 12; //differential encoding + } +} + +/** + * Calculate the category and category_index vector. + * + * @param q pointer to the COOKContext + * @param quant_index_table pointer to the array + * @param category pointer to the category array + * @param category_index pointer to the category_index array + */ + +static void categorize(COOKContext *q, int* quant_index_table, + int* category, int* category_index){ + int exp_idx, bias, tmpbias, bits_left, num_bits, index, v, i, j; + int exp_index2[102]; + int exp_index1[102]; + + int tmp_categorize_array1[128]; + int tmp_categorize_array1_idx=0; + int tmp_categorize_array2[128]; + int tmp_categorize_array2_idx=0; + int category_index_size=0; + + bits_left = q->bits_per_subpacket - get_bits_count(&q->gb); + + if(bits_left > q->samples_per_channel) { + bits_left = q->samples_per_channel + + ((bits_left - q->samples_per_channel)*5)/8; + //av_log(NULL, AV_LOG_ERROR, "bits_left = %d\n",bits_left); + } + + memset(&exp_index1,0,102*sizeof(int)); + memset(&exp_index2,0,102*sizeof(int)); + memset(&tmp_categorize_array1,0,128*sizeof(int)); + memset(&tmp_categorize_array2,0,128*sizeof(int)); + + bias=-32; + + /* Estimate bias. */ + for (i=32 ; i>0 ; i=i/2){ + num_bits = 0; + index = 0; + for (j=q->total_subbands ; j>0 ; j--){ + exp_idx = (i - quant_index_table[index] + bias) / 2; + if (exp_idx<0){ + exp_idx=0; + } else if(exp_idx >7) { + exp_idx=7; + } + index++; + num_bits+=expbits_tab[exp_idx]; + } + if(num_bits >= bits_left - 32){ + bias+=i; + } + } + + /* Calculate total number of bits. */ + num_bits=0; + for (i=0 ; i<q->total_subbands ; i++) { + exp_idx = (bias - quant_index_table[i]) / 2; + if (exp_idx<0) { + exp_idx=0; + } else if(exp_idx >7) { + exp_idx=7; + } + num_bits += expbits_tab[exp_idx]; + exp_index1[i] = exp_idx; + exp_index2[i] = exp_idx; + } + tmpbias = bias = num_bits; + + for (j = 1 ; j < q->numvector_size ; j++) { + if (tmpbias + bias > 2*bits_left) { /* ---> */ + int max = -999999; + index=-1; + for (i=0 ; i<q->total_subbands ; i++){ + if (exp_index1[i] < 7) { + v = (-2*exp_index1[i]) - quant_index_table[i] - 32; + if ( v >= max) { + max = v; + index = i; + } + } + } + if(index==-1)break; + tmp_categorize_array1[tmp_categorize_array1_idx++] = index; + tmpbias -= expbits_tab[exp_index1[index]] - + expbits_tab[exp_index1[index]+1]; + ++exp_index1[index]; + } else { /* <--- */ + int min = 999999; + index=-1; + for (i=0 ; i<q->total_subbands ; i++){ + if(exp_index2[i] > 0){ + v = (-2*exp_index2[i])-quant_index_table[i]; + if ( v < min) { + min = v; + index = i; + } + } + } + if(index == -1)break; + tmp_categorize_array2[tmp_categorize_array2_idx++] = index; + tmpbias -= expbits_tab[exp_index2[index]] - + expbits_tab[exp_index2[index]-1]; + --exp_index2[index]; + } + } + + for(i=0 ; i<q->total_subbands ; i++) + category[i] = exp_index2[i]; + + /* Concatenate the two arrays. */ + for(i=tmp_categorize_array2_idx-1 ; i >= 0; i--) + category_index[category_index_size++] = tmp_categorize_array2[i]; + + for(i=0;i<tmp_categorize_array1_idx;i++) + category_index[category_index_size++ ] = tmp_categorize_array1[i]; + + /* FIXME: mc_sich_ra8_20.rm triggers this, not sure with what we + should fill the remaining bytes. */ + for(i=category_index_size;i<q->numvector_size;i++) + category_index[i]=0; + +} + + +/** + * Expand the category vector. + * + * @param q pointer to the COOKContext + * @param category pointer to the category array + * @param category_index pointer to the category_index array + */ + +static void inline expand_category(COOKContext *q, int* category, + int* category_index){ + int i; + for(i=0 ; i<q->num_vectors ; i++){ + ++category[category_index[i]]; + } +} + +/** + * The real requantization of the mltcoefs + * + * @param q pointer to the COOKContext + * @param index index + * @param 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(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_random(&q->random_state) < 0x80000000) f1 = -f1; + } + mlt_p[i] = f1 * q->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, int category, int* subband_coef_index, + int* subband_coef_sign) { + int i,j; + int vlc, vd ,tmp, result; + int ub; + int cb; + + vd = vd_tab[category]; + result = 0; + for(i=0 ; i<vpr_tab[category] ; i++){ + ub = get_bits_count(&q->gb); + vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3); + cb = get_bits_count(&q->gb); + if (q->bits_per_subpacket < get_bits_count(&q->gb)){ + vlc = 0; + result = 1; + } + for(j=vd-1 ; j>=0 ; j--){ + tmp = (vlc * invradix_tab[category])/0x100000; + subband_coef_index[vd*i+j] = vlc - tmp * (kmax_tab[category]+1); + vlc = tmp; + } + for(j=0 ; j<vd ; j++){ + if (subband_coef_index[i*vd + j]) { + if(get_bits_count(&q->gb) < q->bits_per_subpacket){ + subband_coef_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, 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<q->total_subbands ; band++){ + index = category[band]; + if(category[band] < 7){ + if(unpack_SQVH(q, category[band], subband_coef_index, subband_coef_sign)){ + index=7; + for(j=0 ; j<q->total_subbands ; j++) category[band+j]=7; + } + } + if(index==7) { + memset(subband_coef_index, 0, sizeof(subband_coef_index)); + memset(subband_coef_sign, 0, sizeof(subband_coef_sign)); + } + scalar_dequant(q, index, quant_index_table[band], + subband_coef_index, subband_coef_sign, + &mlt_buffer[band * 20]); + } + + if(q->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_buffer1 pointer to left channel mlt coefficients + * @param mlt_buffer2 pointer to right channel mlt coefficients + */ + +static void mono_decode(COOKContext *q, float* mlt_buffer) { + + int category_index[128]; + int quant_index_table[102]; + int category[128]; + + memset(&category, 0, 128*sizeof(int)); + memset(&category_index, 0, 128*sizeof(int)); + + decode_envelope(q, quant_index_table); + q->num_vectors = get_bits(&q->gb,q->log2_numvector_size); + categorize(q, quant_index_table, category, category_index); + expand_category(q, category, category_index); + decode_vectors(q, 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(COOKContext *q, float* buffer, + int gain_index, int gain_index_next){ + int i; + float fc1, fc2; + fc1 = q->pow2tab[gain_index+63]; + + if(gain_index == gain_index_next){ //static gain + for(i=0 ; i<q->gain_size_factor ; i++){ + buffer[i]*=fc1; + } + return; + } else { //smooth gain + fc2 = q->gain_table[11 + (gain_index_next-gain_index)]; + for(i=0 ; i<q->gain_size_factor ; i++){ + buffer[i]*=fc1; + fc1*=fc2; + } + return; + } +} + + +/** + * 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) +{ + const float fc = q->pow2tab[gains_ptr->previous[0] + 63]; + float *buffer0 = q->mono_mdct_output; + float *buffer1 = q->mono_mdct_output + q->samples_per_channel; + int i; + + /* Inverse modified discrete cosine transform */ + q->mdct_ctx.fft.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, + inbuffer, q->mdct_tmp); + + /* 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]; + } + + /* Apply gain profile */ + for (i = 0; i < 8; i++) { + if (gains_ptr->now[i] || gains_ptr->now[i + 1]) + 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, int* decouple_tab){ + int length, i; + + if(get_bits1(&q->gb)) { + if(cplband[q->js_subband_start] > cplband[q->subbands-1]) return; + + length = cplband[q->subbands-1] - cplband[q->js_subband_start] + 1; + for (i=0 ; i<length ; i++) { + decouple_tab[cplband[q->js_subband_start] + i] = get_vlc2(&q->gb, q->ccpl.table, q->ccpl.bits, 2); + } + return; + } + + if(cplband[q->js_subband_start] > cplband[q->subbands-1]) return; + + length = cplband[q->subbands-1] - cplband[q->js_subband_start] + 1; + for (i=0 ; i<length ; i++) { + decouple_tab[cplband[q->js_subband_start] + i] = get_bits(&q->gb, q->js_vlc_bits); + } + return; +} + + +/** + * function for decoding joint stereo data + * + * @param q pointer to the COOKContext + * @param mlt_buffer1 pointer to left channel mlt coefficients + * @param mlt_buffer2 pointer to right channel mlt coefficients + */ + +static void joint_decode(COOKContext *q, float* mlt_buffer1, + float* mlt_buffer2) { + int i,j; + int decouple_tab[SUBBAND_SIZE]; + float decode_buffer[1060]; + int idx, cpl_tmp,tmp_idx; + float f1,f2; + float* cplscale; + + memset(decouple_tab, 0, sizeof(decouple_tab)); + memset(decode_buffer, 0, sizeof(decode_buffer)); + + /* Make sure the buffers are zeroed out. */ + memset(mlt_buffer1,0, 1024*sizeof(float)); + memset(mlt_buffer2,0, 1024*sizeof(float)); + decouple_info(q, decouple_tab); + mono_decode(q, decode_buffer); + + /* The two channels are stored interleaved in decode_buffer. */ + for (i=0 ; i<q->js_subband_start ; i++) { + for (j=0 ; j<SUBBAND_SIZE ; j++) { + mlt_buffer1[i*20+j] = decode_buffer[i*40+j]; + mlt_buffer2[i*20+j] = decode_buffer[i*40+20+j]; + } + } + + /* When we reach js_subband_start (the higher frequencies) + the coefficients are stored in a coupling scheme. */ + idx = (1 << q->js_vlc_bits) - 1; + for (i=q->js_subband_start ; i<q->subbands ; i++) { + cpl_tmp = cplband[i]; + idx -=decouple_tab[cpl_tmp]; + cplscale = (float*)cplscales[q->js_vlc_bits-2]; //choose decoupler table + f1 = cplscale[decouple_tab[cpl_tmp]]; + f2 = cplscale[idx-1]; + for (j=0 ; j<SUBBAND_SIZE ; j++) { + tmp_idx = ((q->js_subband_start + i)*20)+j; + mlt_buffer1[20*i + j] = f1 * decode_buffer[tmp_idx]; + mlt_buffer2[20*i + j] = f2 * decode_buffer[tmp_idx]; + } + idx = (1 << q->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, uint8_t *inbuffer, + cook_gains *gains_ptr) +{ + int offset; + + offset = decode_bytes(inbuffer, q->decoded_bytes_buffer, + q->bits_per_subpacket/8); + init_get_bits(&q->gb, q->decoded_bytes_buffer + offset, + q->bits_per_subpacket); + decode_gain_info(&q->gb, gains_ptr->now); + + /* Swap current and previous gains */ + FFSWAP(int *, gains_ptr->now, gains_ptr->previous); +} + +/** + * 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) +{ + float *output = q->mono_mdct_output + q->samples_per_channel; + int j; + + imlt_gain(q, decode_buffer, gains, previous_buffer); + + /* Clip and convert floats to 16 bits. + */ + for (j = 0; j < q->samples_per_channel; j++) { + out[chan + q->nb_channels * j] = + av_clip(lrintf(output[j]), -32768, 32767); + } +} + + +/** + * 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 int decode_subpacket(COOKContext *q, uint8_t *inbuffer, + int sub_packet_size, int16_t *outbuffer) { + /* packet dump */ +// for (i=0 ; i<sub_packet_size ; i++) { +// av_log(NULL, AV_LOG_ERROR, "%02x", inbuffer[i]); +// } +// av_log(NULL, AV_LOG_ERROR, "\n"); + + decode_bytes_and_gain(q, inbuffer, &q->gains1); + + if (q->joint_stereo) { + joint_decode(q, q->decode_buffer_1, q->decode_buffer_2); + } else { + mono_decode(q, q->decode_buffer_1); + + if (q->nb_channels == 2) { + decode_bytes_and_gain(q, inbuffer + sub_packet_size/2, &q->gains2); + mono_decode(q, q->decode_buffer_2); + } + } + + mlt_compensate_output(q, q->decode_buffer_1, &q->gains1, + q->mono_previous_buffer1, outbuffer, 0); + + if (q->nb_channels == 2) { + if (q->joint_stereo) { + mlt_compensate_output(q, q->decode_buffer_2, &q->gains1, + q->mono_previous_buffer2, outbuffer, 1); + } else { + mlt_compensate_output(q, q->decode_buffer_2, &q->gains2, + q->mono_previous_buffer2, outbuffer, 1); + } + } + return q->samples_per_frame * sizeof(int16_t); +} + + +/** + * Cook frame decoding + * + * @param avctx pointer to the AVCodecContext + */ + +static int cook_decode_frame(AVCodecContext *avctx, + void *data, int *data_size, + uint8_t *buf, int buf_size) { + COOKContext *q = avctx->priv_data; + + if (buf_size < avctx->block_align) + return buf_size; + + *data_size = decode_subpacket(q, buf, avctx->block_align, data); + + /* 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(NULL,AV_LOG_ERROR," %s = %d\n", a, b); + av_log(NULL,AV_LOG_ERROR,"COOKextradata\n"); + av_log(NULL,AV_LOG_ERROR,"cookversion=%x\n",q->cookversion); + if (q->cookversion > STEREO) { + PRINT("js_subband_start",q->js_subband_start); + PRINT("js_vlc_bits",q->js_vlc_bits); + } + av_log(NULL,AV_LOG_ERROR,"COOKContext\n"); + PRINT("nb_channels",q->nb_channels); + PRINT("bit_rate",q->bit_rate); + PRINT("sample_rate",q->sample_rate); + PRINT("samples_per_channel",q->samples_per_channel); + PRINT("samples_per_frame",q->samples_per_frame); + PRINT("subbands",q->subbands); + PRINT("random_state",q->random_state); + PRINT("js_subband_start",q->js_subband_start); + PRINT("log2_numvector_size",q->log2_numvector_size); + PRINT("numvector_size",q->numvector_size); + PRINT("total_subbands",q->total_subbands); +} +#endif + +/** + * Cook initialization + * + * @param avctx pointer to the AVCodecContext + */ + +static int cook_decode_init(AVCodecContext *avctx) +{ + COOKContext *q = avctx->priv_data; + uint8_t *edata_ptr = avctx->extradata; + + /* Take care of the codec specific extradata. */ + if (avctx->extradata_size <= 0) { + av_log(avctx,AV_LOG_ERROR,"Necessary extradata missing!\n"); + return -1; + } else { + /* 8 for mono, 16 for stereo, ? for multichannel + Swap to right endianness so we don't need to care later on. */ + av_log(avctx,AV_LOG_DEBUG,"codecdata_length=%d\n",avctx->extradata_size); + if (avctx->extradata_size >= 8){ + q->cookversion = bytestream_get_be32(&edata_ptr); + q->samples_per_frame = bytestream_get_be16(&edata_ptr); + q->subbands = bytestream_get_be16(&edata_ptr); + } + if (avctx->extradata_size >= 16){ + bytestream_get_be32(&edata_ptr); //Unknown unused + q->js_subband_start = bytestream_get_be16(&edata_ptr); + q->js_vlc_bits = bytestream_get_be16(&edata_ptr); + } + } + + /* 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_init_random(1, &q->random_state); + + /* Initialize extradata related variables. */ + q->samples_per_channel = q->samples_per_frame / q->nb_channels; + q->bits_per_subpacket = avctx->block_align * 8; + + /* Initialize default data states. */ + q->log2_numvector_size = 5; + q->total_subbands = q->subbands; + + /* Initialize version-dependent variables */ + av_log(NULL,AV_LOG_DEBUG,"q->cookversion=%x\n",q->cookversion); + q->joint_stereo = 0; + switch (q->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->bits_per_subpacket = q->bits_per_subpacket/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->total_subbands = q->subbands + q->js_subband_start; + q->joint_stereo = 1; + } + if (q->samples_per_channel > 256) { + q->log2_numvector_size = 6; + } + if (q->samples_per_channel > 512) { + q->log2_numvector_size = 7; + } + break; + case MC_COOK: + av_log(avctx,AV_LOG_ERROR,"MC_COOK not supported!\n"); + return -1; + break; + default: + av_log(avctx,AV_LOG_ERROR,"Unknown Cook version, report sample!\n"); + return -1; + break; + } + + /* Initialize variable relations */ + q->numvector_size = (1 << q->log2_numvector_size); + + /* Generate tables */ + init_rootpow2table(q); + init_pow2table(q); + init_gain_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. */ + if (q->nb_channels==2 && q->joint_stereo==0) { + q->decoded_bytes_buffer = + av_mallocz(avctx->block_align/2 + + DECODE_BYTES_PAD2(avctx->block_align/2) + + FF_INPUT_BUFFER_PADDING_SIZE); + } else { + 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; + + q->gains1.now = q->gain_1; + q->gains1.previous = q->gain_2; + q->gains2.now = q->gain_3; + q->gains2.previous = q->gain_4; + + /* Initialize transform. */ + if ( init_cook_mlt(q) != 0 ) + return -1; + + /* Try to catch some obviously faulty streams, othervise it might be exploitable */ + if (q->total_subbands > 53) { + av_log(avctx,AV_LOG_ERROR,"total_subbands > 53, report sample!\n"); + return -1; + } + if (q->subbands > 50) { + av_log(avctx,AV_LOG_ERROR,"subbands > 50, report sample!\n"); + return -1; + } + 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; + } + if ((q->js_vlc_bits > 6) || (q->js_vlc_bits < 0)) { + av_log(avctx,AV_LOG_ERROR,"q->js_vlc_bits = %d, only >= 0 and <= 6 allowed!\n",q->js_vlc_bits); + return -1; + } + +#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, +}; |