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libavcodec/qdm2.c

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00001 /*
00002  * QDM2 compatible decoder
00003  * Copyright (c) 2003 Ewald Snel
00004  * Copyright (c) 2005 Benjamin Larsson
00005  * Copyright (c) 2005 Alex Beregszaszi
00006  * Copyright (c) 2005 Roberto Togni
00007  *
00008  * This file is part of FFmpeg.
00009  *
00010  * FFmpeg is free software; you can redistribute it and/or
00011  * modify it under the terms of the GNU Lesser General Public
00012  * License as published by the Free Software Foundation; either
00013  * version 2.1 of the License, or (at your option) any later version.
00014  *
00015  * FFmpeg is distributed in the hope that it will be useful,
00016  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00017  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
00018  * Lesser General Public License for more details.
00019  *
00020  * You should have received a copy of the GNU Lesser General Public
00021  * License along with FFmpeg; if not, write to the Free Software
00022  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
00023  */
00024 
00033 #include <math.h>
00034 #include <stddef.h>
00035 #include <stdio.h>
00036 
00037 #define ALT_BITSTREAM_READER_LE
00038 #include "avcodec.h"
00039 #include "get_bits.h"
00040 #include "dsputil.h"
00041 #include "rdft.h"
00042 #include "mpegaudiodsp.h"
00043 #include "mpegaudio.h"
00044 
00045 #include "qdm2data.h"
00046 #include "qdm2_tablegen.h"
00047 
00048 #undef NDEBUG
00049 #include <assert.h>
00050 
00051 
00052 #define QDM2_LIST_ADD(list, size, packet) \
00053 do { \
00054       if (size > 0) { \
00055     list[size - 1].next = &list[size]; \
00056       } \
00057       list[size].packet = packet; \
00058       list[size].next = NULL; \
00059       size++; \
00060 } while(0)
00061 
00062 // Result is 8, 16 or 30
00063 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
00064 
00065 #define FIX_NOISE_IDX(noise_idx) \
00066   if ((noise_idx) >= 3840) \
00067     (noise_idx) -= 3840; \
00068 
00069 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
00070 
00071 #define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
00072 
00073 #define SAMPLES_NEEDED \
00074      av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
00075 
00076 #define SAMPLES_NEEDED_2(why) \
00077      av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
00078 
00079 #define QDM2_MAX_FRAME_SIZE 512
00080 
00081 typedef int8_t sb_int8_array[2][30][64];
00082 
00086 typedef struct {
00087     int type;            
00088     unsigned int size;   
00089     const uint8_t *data; 
00090 } QDM2SubPacket;
00091 
00095 typedef struct QDM2SubPNode {
00096     QDM2SubPacket *packet;      
00097     struct QDM2SubPNode *next; 
00098 } QDM2SubPNode;
00099 
00100 typedef struct {
00101     float re;
00102     float im;
00103 } QDM2Complex;
00104 
00105 typedef struct {
00106     float level;
00107     QDM2Complex *complex;
00108     const float *table;
00109     int   phase;
00110     int   phase_shift;
00111     int   duration;
00112     short time_index;
00113     short cutoff;
00114 } FFTTone;
00115 
00116 typedef struct {
00117     int16_t sub_packet;
00118     uint8_t channel;
00119     int16_t offset;
00120     int16_t exp;
00121     uint8_t phase;
00122 } FFTCoefficient;
00123 
00124 typedef struct {
00125     DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
00126 } QDM2FFT;
00127 
00131 typedef struct {
00133     int nb_channels;         
00134     int channels;            
00135     int group_size;          
00136     int fft_size;            
00137     int checksum_size;       
00138 
00140     int group_order;         
00141     int fft_order;           
00142     int fft_frame_size;      
00143     int frame_size;          
00144     int frequency_range;
00145     int sub_sampling;        
00146     int coeff_per_sb_select; 
00147     int cm_table_select;     
00148 
00150     QDM2SubPacket sub_packets[16];      
00151     QDM2SubPNode sub_packet_list_A[16]; 
00152     QDM2SubPNode sub_packet_list_B[16]; 
00153     int sub_packets_B;                  
00154     QDM2SubPNode sub_packet_list_C[16]; 
00155     QDM2SubPNode sub_packet_list_D[16]; 
00156 
00158     FFTTone fft_tones[1000];
00159     int fft_tone_start;
00160     int fft_tone_end;
00161     FFTCoefficient fft_coefs[1000];
00162     int fft_coefs_index;
00163     int fft_coefs_min_index[5];
00164     int fft_coefs_max_index[5];
00165     int fft_level_exp[6];
00166     RDFTContext rdft_ctx;
00167     QDM2FFT fft;
00168 
00170     const uint8_t *compressed_data;
00171     int compressed_size;
00172     float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
00173 
00175     MPADSPContext mpadsp;
00176     DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
00177     int synth_buf_offset[MPA_MAX_CHANNELS];
00178     DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
00179     DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
00180 
00182     float tone_level[MPA_MAX_CHANNELS][30][64];
00183     int8_t coding_method[MPA_MAX_CHANNELS][30][64];
00184     int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
00185     int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
00186     int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
00187     int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
00188     int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
00189     int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
00190     int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
00191 
00192     // Flags
00193     int has_errors;         
00194     int superblocktype_2_3; 
00195     int do_synth_filter;    
00196 
00197     int sub_packet;
00198     int noise_idx; 
00199 } QDM2Context;
00200 
00201 
00202 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
00203 
00204 static VLC vlc_tab_level;
00205 static VLC vlc_tab_diff;
00206 static VLC vlc_tab_run;
00207 static VLC fft_level_exp_alt_vlc;
00208 static VLC fft_level_exp_vlc;
00209 static VLC fft_stereo_exp_vlc;
00210 static VLC fft_stereo_phase_vlc;
00211 static VLC vlc_tab_tone_level_idx_hi1;
00212 static VLC vlc_tab_tone_level_idx_mid;
00213 static VLC vlc_tab_tone_level_idx_hi2;
00214 static VLC vlc_tab_type30;
00215 static VLC vlc_tab_type34;
00216 static VLC vlc_tab_fft_tone_offset[5];
00217 
00218 static const uint16_t qdm2_vlc_offs[] = {
00219     0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
00220 };
00221 
00222 static av_cold void qdm2_init_vlc(void)
00223 {
00224     static int vlcs_initialized = 0;
00225     static VLC_TYPE qdm2_table[3838][2];
00226 
00227     if (!vlcs_initialized) {
00228 
00229         vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]];
00230         vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
00231         init_vlc (&vlc_tab_level, 8, 24,
00232             vlc_tab_level_huffbits, 1, 1,
00233             vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00234 
00235         vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
00236         vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
00237         init_vlc (&vlc_tab_diff, 8, 37,
00238             vlc_tab_diff_huffbits, 1, 1,
00239             vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00240 
00241         vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]];
00242         vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
00243         init_vlc (&vlc_tab_run, 5, 6,
00244             vlc_tab_run_huffbits, 1, 1,
00245             vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00246 
00247         fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
00248         fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
00249         init_vlc (&fft_level_exp_alt_vlc, 8, 28,
00250             fft_level_exp_alt_huffbits, 1, 1,
00251             fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00252 
00253 
00254         fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
00255         fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
00256         init_vlc (&fft_level_exp_vlc, 8, 20,
00257             fft_level_exp_huffbits, 1, 1,
00258             fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00259 
00260         fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
00261         fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
00262         init_vlc (&fft_stereo_exp_vlc, 6, 7,
00263             fft_stereo_exp_huffbits, 1, 1,
00264             fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00265 
00266         fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
00267         fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
00268         init_vlc (&fft_stereo_phase_vlc, 6, 9,
00269             fft_stereo_phase_huffbits, 1, 1,
00270             fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00271 
00272         vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]];
00273         vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
00274         init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
00275             vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
00276             vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00277 
00278         vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]];
00279         vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
00280         init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
00281             vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
00282             vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00283 
00284         vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]];
00285         vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
00286         init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
00287             vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
00288             vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00289 
00290         vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
00291         vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
00292         init_vlc (&vlc_tab_type30, 6, 9,
00293             vlc_tab_type30_huffbits, 1, 1,
00294             vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00295 
00296         vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
00297         vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
00298         init_vlc (&vlc_tab_type34, 5, 10,
00299             vlc_tab_type34_huffbits, 1, 1,
00300             vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00301 
00302         vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
00303         vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
00304         init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
00305             vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
00306             vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00307 
00308         vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
00309         vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
00310         init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
00311             vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
00312             vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00313 
00314         vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
00315         vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
00316         init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
00317             vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
00318             vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00319 
00320         vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
00321         vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
00322         init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
00323             vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
00324             vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00325 
00326         vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
00327         vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
00328         init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
00329             vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
00330             vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00331 
00332         vlcs_initialized=1;
00333     }
00334 }
00335 
00336 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
00337 {
00338     int value;
00339 
00340     value = get_vlc2(gb, vlc->table, vlc->bits, depth);
00341 
00342     /* stage-2, 3 bits exponent escape sequence */
00343     if (value-- == 0)
00344         value = get_bits (gb, get_bits (gb, 3) + 1);
00345 
00346     /* stage-3, optional */
00347     if (flag) {
00348         int tmp = vlc_stage3_values[value];
00349 
00350         if ((value & ~3) > 0)
00351             tmp += get_bits (gb, (value >> 2));
00352         value = tmp;
00353     }
00354 
00355     return value;
00356 }
00357 
00358 
00359 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
00360 {
00361     int value = qdm2_get_vlc (gb, vlc, 0, depth);
00362 
00363     return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
00364 }
00365 
00366 
00376 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
00377     int i;
00378 
00379     for (i=0; i < length; i++)
00380         value -= data[i];
00381 
00382     return (uint16_t)(value & 0xffff);
00383 }
00384 
00385 
00392 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
00393 {
00394     sub_packet->type = get_bits (gb, 8);
00395 
00396     if (sub_packet->type == 0) {
00397         sub_packet->size = 0;
00398         sub_packet->data = NULL;
00399     } else {
00400         sub_packet->size = get_bits (gb, 8);
00401 
00402       if (sub_packet->type & 0x80) {
00403           sub_packet->size <<= 8;
00404           sub_packet->size  |= get_bits (gb, 8);
00405           sub_packet->type  &= 0x7f;
00406       }
00407 
00408       if (sub_packet->type == 0x7f)
00409           sub_packet->type |= (get_bits (gb, 8) << 8);
00410 
00411       sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
00412     }
00413 
00414     av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
00415         sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
00416 }
00417 
00418 
00426 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
00427 {
00428     while (list != NULL && list->packet != NULL) {
00429         if (list->packet->type == type)
00430             return list;
00431         list = list->next;
00432     }
00433     return NULL;
00434 }
00435 
00436 
00443 static void average_quantized_coeffs (QDM2Context *q)
00444 {
00445     int i, j, n, ch, sum;
00446 
00447     n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
00448 
00449     for (ch = 0; ch < q->nb_channels; ch++)
00450         for (i = 0; i < n; i++) {
00451             sum = 0;
00452 
00453             for (j = 0; j < 8; j++)
00454                 sum += q->quantized_coeffs[ch][i][j];
00455 
00456             sum /= 8;
00457             if (sum > 0)
00458                 sum--;
00459 
00460             for (j=0; j < 8; j++)
00461                 q->quantized_coeffs[ch][i][j] = sum;
00462         }
00463 }
00464 
00465 
00473 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
00474 {
00475     int ch, j;
00476 
00477     FIX_NOISE_IDX(q->noise_idx);
00478 
00479     if (!q->nb_channels)
00480         return;
00481 
00482     for (ch = 0; ch < q->nb_channels; ch++)
00483         for (j = 0; j < 64; j++) {
00484             q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
00485             q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
00486         }
00487 }
00488 
00489 
00498 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
00499 {
00500     int j,k;
00501     int ch;
00502     int run, case_val;
00503     int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4};
00504 
00505     for (ch = 0; ch < channels; ch++) {
00506         for (j = 0; j < 64; ) {
00507             if((coding_method[ch][sb][j] - 8) > 22) {
00508                 run = 1;
00509                 case_val = 8;
00510             } else {
00511                 switch (switchtable[coding_method[ch][sb][j]-8]) {
00512                     case 0: run = 10; case_val = 10; break;
00513                     case 1: run = 1; case_val = 16; break;
00514                     case 2: run = 5; case_val = 24; break;
00515                     case 3: run = 3; case_val = 30; break;
00516                     case 4: run = 1; case_val = 30; break;
00517                     case 5: run = 1; case_val = 8; break;
00518                     default: run = 1; case_val = 8; break;
00519                 }
00520             }
00521             for (k = 0; k < run; k++)
00522                 if (j + k < 128)
00523                     if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
00524                         if (k > 0) {
00525                            SAMPLES_NEEDED
00526                             //not debugged, almost never used
00527                             memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
00528                             memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
00529                         }
00530             j += run;
00531         }
00532     }
00533 }
00534 
00535 
00543 static void fill_tone_level_array (QDM2Context *q, int flag)
00544 {
00545     int i, sb, ch, sb_used;
00546     int tmp, tab;
00547 
00548     // This should never happen
00549     if (q->nb_channels <= 0)
00550         return;
00551 
00552     for (ch = 0; ch < q->nb_channels; ch++)
00553         for (sb = 0; sb < 30; sb++)
00554             for (i = 0; i < 8; i++) {
00555                 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
00556                     tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
00557                           q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
00558                 else
00559                     tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
00560                 if(tmp < 0)
00561                     tmp += 0xff;
00562                 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
00563             }
00564 
00565     sb_used = QDM2_SB_USED(q->sub_sampling);
00566 
00567     if ((q->superblocktype_2_3 != 0) && !flag) {
00568         for (sb = 0; sb < sb_used; sb++)
00569             for (ch = 0; ch < q->nb_channels; ch++)
00570                 for (i = 0; i < 64; i++) {
00571                     q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
00572                     if (q->tone_level_idx[ch][sb][i] < 0)
00573                         q->tone_level[ch][sb][i] = 0;
00574                     else
00575                         q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
00576                 }
00577     } else {
00578         tab = q->superblocktype_2_3 ? 0 : 1;
00579         for (sb = 0; sb < sb_used; sb++) {
00580             if ((sb >= 4) && (sb <= 23)) {
00581                 for (ch = 0; ch < q->nb_channels; ch++)
00582                     for (i = 0; i < 64; i++) {
00583                         tmp = q->tone_level_idx_base[ch][sb][i / 8] -
00584                               q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
00585                               q->tone_level_idx_mid[ch][sb - 4][i / 8] -
00586                               q->tone_level_idx_hi2[ch][sb - 4];
00587                         q->tone_level_idx[ch][sb][i] = tmp & 0xff;
00588                         if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
00589                             q->tone_level[ch][sb][i] = 0;
00590                         else
00591                             q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
00592                 }
00593             } else {
00594                 if (sb > 4) {
00595                     for (ch = 0; ch < q->nb_channels; ch++)
00596                         for (i = 0; i < 64; i++) {
00597                             tmp = q->tone_level_idx_base[ch][sb][i / 8] -
00598                                   q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
00599                                   q->tone_level_idx_hi2[ch][sb - 4];
00600                             q->tone_level_idx[ch][sb][i] = tmp & 0xff;
00601                             if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
00602                                 q->tone_level[ch][sb][i] = 0;
00603                             else
00604                                 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
00605                     }
00606                 } else {
00607                     for (ch = 0; ch < q->nb_channels; ch++)
00608                         for (i = 0; i < 64; i++) {
00609                             tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
00610                             if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
00611                                 q->tone_level[ch][sb][i] = 0;
00612                             else
00613                                 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
00614                         }
00615                 }
00616             }
00617         }
00618     }
00619 
00620     return;
00621 }
00622 
00623 
00638 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
00639                 sb_int8_array coding_method, int nb_channels,
00640                 int c, int superblocktype_2_3, int cm_table_select)
00641 {
00642     int ch, sb, j;
00643     int tmp, acc, esp_40, comp;
00644     int add1, add2, add3, add4;
00645     int64_t multres;
00646 
00647     // This should never happen
00648     if (nb_channels <= 0)
00649         return;
00650 
00651     if (!superblocktype_2_3) {
00652         /* This case is untested, no samples available */
00653         SAMPLES_NEEDED
00654         for (ch = 0; ch < nb_channels; ch++)
00655             for (sb = 0; sb < 30; sb++) {
00656                 for (j = 1; j < 63; j++) {  // The loop only iterates to 63 so the code doesn't overflow the buffer
00657                     add1 = tone_level_idx[ch][sb][j] - 10;
00658                     if (add1 < 0)
00659                         add1 = 0;
00660                     add2 = add3 = add4 = 0;
00661                     if (sb > 1) {
00662                         add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
00663                         if (add2 < 0)
00664                             add2 = 0;
00665                     }
00666                     if (sb > 0) {
00667                         add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
00668                         if (add3 < 0)
00669                             add3 = 0;
00670                     }
00671                     if (sb < 29) {
00672                         add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
00673                         if (add4 < 0)
00674                             add4 = 0;
00675                     }
00676                     tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
00677                     if (tmp < 0)
00678                         tmp = 0;
00679                     tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
00680                 }
00681                 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
00682             }
00683             acc = 0;
00684             for (ch = 0; ch < nb_channels; ch++)
00685                 for (sb = 0; sb < 30; sb++)
00686                     for (j = 0; j < 64; j++)
00687                         acc += tone_level_idx_temp[ch][sb][j];
00688 
00689             multres = 0x66666667 * (acc * 10);
00690             esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
00691             for (ch = 0;  ch < nb_channels; ch++)
00692                 for (sb = 0; sb < 30; sb++)
00693                     for (j = 0; j < 64; j++) {
00694                         comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
00695                         if (comp < 0)
00696                             comp += 0xff;
00697                         comp /= 256; // signed shift
00698                         switch(sb) {
00699                             case 0:
00700                                 if (comp < 30)
00701                                     comp = 30;
00702                                 comp += 15;
00703                                 break;
00704                             case 1:
00705                                 if (comp < 24)
00706                                     comp = 24;
00707                                 comp += 10;
00708                                 break;
00709                             case 2:
00710                             case 3:
00711                             case 4:
00712                                 if (comp < 16)
00713                                     comp = 16;
00714                         }
00715                         if (comp <= 5)
00716                             tmp = 0;
00717                         else if (comp <= 10)
00718                             tmp = 10;
00719                         else if (comp <= 16)
00720                             tmp = 16;
00721                         else if (comp <= 24)
00722                             tmp = -1;
00723                         else
00724                             tmp = 0;
00725                         coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
00726                     }
00727             for (sb = 0; sb < 30; sb++)
00728                 fix_coding_method_array(sb, nb_channels, coding_method);
00729             for (ch = 0; ch < nb_channels; ch++)
00730                 for (sb = 0; sb < 30; sb++)
00731                     for (j = 0; j < 64; j++)
00732                         if (sb >= 10) {
00733                             if (coding_method[ch][sb][j] < 10)
00734                                 coding_method[ch][sb][j] = 10;
00735                         } else {
00736                             if (sb >= 2) {
00737                                 if (coding_method[ch][sb][j] < 16)
00738                                     coding_method[ch][sb][j] = 16;
00739                             } else {
00740                                 if (coding_method[ch][sb][j] < 30)
00741                                     coding_method[ch][sb][j] = 30;
00742                             }
00743                         }
00744     } else { // superblocktype_2_3 != 0
00745         for (ch = 0; ch < nb_channels; ch++)
00746             for (sb = 0; sb < 30; sb++)
00747                 for (j = 0; j < 64; j++)
00748                     coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
00749     }
00750 
00751     return;
00752 }
00753 
00754 
00766 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
00767 {
00768     int sb, j, k, n, ch, run, channels;
00769     int joined_stereo, zero_encoding, chs;
00770     int type34_first;
00771     float type34_div = 0;
00772     float type34_predictor;
00773     float samples[10], sign_bits[16];
00774 
00775     if (length == 0) {
00776         // If no data use noise
00777         for (sb=sb_min; sb < sb_max; sb++)
00778             build_sb_samples_from_noise (q, sb);
00779 
00780         return;
00781     }
00782 
00783     for (sb = sb_min; sb < sb_max; sb++) {
00784         FIX_NOISE_IDX(q->noise_idx);
00785 
00786         channels = q->nb_channels;
00787 
00788         if (q->nb_channels <= 1 || sb < 12)
00789             joined_stereo = 0;
00790         else if (sb >= 24)
00791             joined_stereo = 1;
00792         else
00793             joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
00794 
00795         if (joined_stereo) {
00796             if (BITS_LEFT(length,gb) >= 16)
00797                 for (j = 0; j < 16; j++)
00798                     sign_bits[j] = get_bits1 (gb);
00799 
00800             for (j = 0; j < 64; j++)
00801                 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
00802                     q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
00803 
00804             fix_coding_method_array(sb, q->nb_channels, q->coding_method);
00805             channels = 1;
00806         }
00807 
00808         for (ch = 0; ch < channels; ch++) {
00809             zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
00810             type34_predictor = 0.0;
00811             type34_first = 1;
00812 
00813             for (j = 0; j < 128; ) {
00814                 switch (q->coding_method[ch][sb][j / 2]) {
00815                     case 8:
00816                         if (BITS_LEFT(length,gb) >= 10) {
00817                             if (zero_encoding) {
00818                                 for (k = 0; k < 5; k++) {
00819                                     if ((j + 2 * k) >= 128)
00820                                         break;
00821                                     samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
00822                                 }
00823                             } else {
00824                                 n = get_bits(gb, 8);
00825                                 for (k = 0; k < 5; k++)
00826                                     samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
00827                             }
00828                             for (k = 0; k < 5; k++)
00829                                 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
00830                         } else {
00831                             for (k = 0; k < 10; k++)
00832                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
00833                         }
00834                         run = 10;
00835                         break;
00836 
00837                     case 10:
00838                         if (BITS_LEFT(length,gb) >= 1) {
00839                             float f = 0.81;
00840 
00841                             if (get_bits1(gb))
00842                                 f = -f;
00843                             f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
00844                             samples[0] = f;
00845                         } else {
00846                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00847                         }
00848                         run = 1;
00849                         break;
00850 
00851                     case 16:
00852                         if (BITS_LEFT(length,gb) >= 10) {
00853                             if (zero_encoding) {
00854                                 for (k = 0; k < 5; k++) {
00855                                     if ((j + k) >= 128)
00856                                         break;
00857                                     samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
00858                                 }
00859                             } else {
00860                                 n = get_bits (gb, 8);
00861                                 for (k = 0; k < 5; k++)
00862                                     samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
00863                             }
00864                         } else {
00865                             for (k = 0; k < 5; k++)
00866                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
00867                         }
00868                         run = 5;
00869                         break;
00870 
00871                     case 24:
00872                         if (BITS_LEFT(length,gb) >= 7) {
00873                             n = get_bits(gb, 7);
00874                             for (k = 0; k < 3; k++)
00875                                 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
00876                         } else {
00877                             for (k = 0; k < 3; k++)
00878                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
00879                         }
00880                         run = 3;
00881                         break;
00882 
00883                     case 30:
00884                         if (BITS_LEFT(length,gb) >= 4)
00885                             samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
00886                         else
00887                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00888 
00889                         run = 1;
00890                         break;
00891 
00892                     case 34:
00893                         if (BITS_LEFT(length,gb) >= 7) {
00894                             if (type34_first) {
00895                                 type34_div = (float)(1 << get_bits(gb, 2));
00896                                 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
00897                                 type34_predictor = samples[0];
00898                                 type34_first = 0;
00899                             } else {
00900                                 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
00901                                 type34_predictor = samples[0];
00902                             }
00903                         } else {
00904                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00905                         }
00906                         run = 1;
00907                         break;
00908 
00909                     default:
00910                         samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00911                         run = 1;
00912                         break;
00913                 }
00914 
00915                 if (joined_stereo) {
00916                     float tmp[10][MPA_MAX_CHANNELS];
00917 
00918                     for (k = 0; k < run; k++) {
00919                         tmp[k][0] = samples[k];
00920                         tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
00921                     }
00922                     for (chs = 0; chs < q->nb_channels; chs++)
00923                         for (k = 0; k < run; k++)
00924                             if ((j + k) < 128)
00925                                 q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs];
00926                 } else {
00927                     for (k = 0; k < run; k++)
00928                         if ((j + k) < 128)
00929                             q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
00930                 }
00931 
00932                 j += run;
00933             } // j loop
00934         } // channel loop
00935     } // subband loop
00936 }
00937 
00938 
00948 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
00949 {
00950     int i, k, run, level, diff;
00951 
00952     if (BITS_LEFT(length,gb) < 16)
00953         return;
00954     level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
00955 
00956     quantized_coeffs[0] = level;
00957 
00958     for (i = 0; i < 7; ) {
00959         if (BITS_LEFT(length,gb) < 16)
00960             break;
00961         run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
00962 
00963         if (BITS_LEFT(length,gb) < 16)
00964             break;
00965         diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
00966 
00967         for (k = 1; k <= run; k++)
00968             quantized_coeffs[i + k] = (level + ((k * diff) / run));
00969 
00970         level += diff;
00971         i += run;
00972     }
00973 }
00974 
00975 
00985 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
00986 {
00987     int sb, j, k, n, ch;
00988 
00989     for (ch = 0; ch < q->nb_channels; ch++) {
00990         init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
00991 
00992         if (BITS_LEFT(length,gb) < 16) {
00993             memset(q->quantized_coeffs[ch][0], 0, 8);
00994             break;
00995         }
00996     }
00997 
00998     n = q->sub_sampling + 1;
00999 
01000     for (sb = 0; sb < n; sb++)
01001         for (ch = 0; ch < q->nb_channels; ch++)
01002             for (j = 0; j < 8; j++) {
01003                 if (BITS_LEFT(length,gb) < 1)
01004                     break;
01005                 if (get_bits1(gb)) {
01006                     for (k=0; k < 8; k++) {
01007                         if (BITS_LEFT(length,gb) < 16)
01008                             break;
01009                         q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
01010                     }
01011                 } else {
01012                     for (k=0; k < 8; k++)
01013                         q->tone_level_idx_hi1[ch][sb][j][k] = 0;
01014                 }
01015             }
01016 
01017     n = QDM2_SB_USED(q->sub_sampling) - 4;
01018 
01019     for (sb = 0; sb < n; sb++)
01020         for (ch = 0; ch < q->nb_channels; ch++) {
01021             if (BITS_LEFT(length,gb) < 16)
01022                 break;
01023             q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
01024             if (sb > 19)
01025                 q->tone_level_idx_hi2[ch][sb] -= 16;
01026             else
01027                 for (j = 0; j < 8; j++)
01028                     q->tone_level_idx_mid[ch][sb][j] = -16;
01029         }
01030 
01031     n = QDM2_SB_USED(q->sub_sampling) - 5;
01032 
01033     for (sb = 0; sb < n; sb++)
01034         for (ch = 0; ch < q->nb_channels; ch++)
01035             for (j = 0; j < 8; j++) {
01036                 if (BITS_LEFT(length,gb) < 16)
01037                     break;
01038                 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
01039             }
01040 }
01041 
01048 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
01049 {
01050     GetBitContext gb;
01051     int i, j, k, n, ch, run, level, diff;
01052 
01053     init_get_bits(&gb, node->packet->data, node->packet->size*8);
01054 
01055     n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
01056 
01057     for (i = 1; i < n; i++)
01058         for (ch=0; ch < q->nb_channels; ch++) {
01059             level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
01060             q->quantized_coeffs[ch][i][0] = level;
01061 
01062             for (j = 0; j < (8 - 1); ) {
01063                 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
01064                 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
01065 
01066                 for (k = 1; k <= run; k++)
01067                     q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
01068 
01069                 level += diff;
01070                 j += run;
01071             }
01072         }
01073 
01074     for (ch = 0; ch < q->nb_channels; ch++)
01075         for (i = 0; i < 8; i++)
01076             q->quantized_coeffs[ch][0][i] = 0;
01077 }
01078 
01079 
01087 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
01088 {
01089     GetBitContext gb;
01090 
01091     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
01092 
01093     if (length != 0) {
01094         init_tone_level_dequantization(q, &gb, length);
01095         fill_tone_level_array(q, 1);
01096     } else {
01097         fill_tone_level_array(q, 0);
01098     }
01099 }
01100 
01101 
01109 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
01110 {
01111     GetBitContext gb;
01112 
01113     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
01114     if (length >= 32) {
01115         int c = get_bits (&gb, 13);
01116 
01117         if (c > 3)
01118             fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
01119                                       q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
01120     }
01121 
01122     synthfilt_build_sb_samples(q, &gb, length, 0, 8);
01123 }
01124 
01125 
01133 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
01134 {
01135     GetBitContext gb;
01136 
01137     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
01138     synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
01139 }
01140 
01141 /*
01142  * Process new subpackets for synthesis filter
01143  *
01144  * @param q       context
01145  * @param list    list with synthesis filter packets (list D)
01146  */
01147 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
01148 {
01149     QDM2SubPNode *nodes[4];
01150 
01151     nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
01152     if (nodes[0] != NULL)
01153         process_subpacket_9(q, nodes[0]);
01154 
01155     nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
01156     if (nodes[1] != NULL)
01157         process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
01158     else
01159         process_subpacket_10(q, NULL, 0);
01160 
01161     nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
01162     if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
01163         process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
01164     else
01165         process_subpacket_11(q, NULL, 0);
01166 
01167     nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
01168     if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
01169         process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
01170     else
01171         process_subpacket_12(q, NULL, 0);
01172 }
01173 
01174 
01175 /*
01176  * Decode superblock, fill packet lists.
01177  *
01178  * @param q    context
01179  */
01180 static void qdm2_decode_super_block (QDM2Context *q)
01181 {
01182     GetBitContext gb;
01183     QDM2SubPacket header, *packet;
01184     int i, packet_bytes, sub_packet_size, sub_packets_D;
01185     unsigned int next_index = 0;
01186 
01187     memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
01188     memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
01189     memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
01190 
01191     q->sub_packets_B = 0;
01192     sub_packets_D = 0;
01193 
01194     average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
01195 
01196     init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
01197     qdm2_decode_sub_packet_header(&gb, &header);
01198 
01199     if (header.type < 2 || header.type >= 8) {
01200         q->has_errors = 1;
01201         av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
01202         return;
01203     }
01204 
01205     q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
01206     packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
01207 
01208     init_get_bits(&gb, header.data, header.size*8);
01209 
01210     if (header.type == 2 || header.type == 4 || header.type == 5) {
01211         int csum  = 257 * get_bits(&gb, 8);
01212             csum +=   2 * get_bits(&gb, 8);
01213 
01214         csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
01215 
01216         if (csum != 0) {
01217             q->has_errors = 1;
01218             av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
01219             return;
01220         }
01221     }
01222 
01223     q->sub_packet_list_B[0].packet = NULL;
01224     q->sub_packet_list_D[0].packet = NULL;
01225 
01226     for (i = 0; i < 6; i++)
01227         if (--q->fft_level_exp[i] < 0)
01228             q->fft_level_exp[i] = 0;
01229 
01230     for (i = 0; packet_bytes > 0; i++) {
01231         int j;
01232 
01233         q->sub_packet_list_A[i].next = NULL;
01234 
01235         if (i > 0) {
01236             q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
01237 
01238             /* seek to next block */
01239             init_get_bits(&gb, header.data, header.size*8);
01240             skip_bits(&gb, next_index*8);
01241 
01242             if (next_index >= header.size)
01243                 break;
01244         }
01245 
01246         /* decode subpacket */
01247         packet = &q->sub_packets[i];
01248         qdm2_decode_sub_packet_header(&gb, packet);
01249         next_index = packet->size + get_bits_count(&gb) / 8;
01250         sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
01251 
01252         if (packet->type == 0)
01253             break;
01254 
01255         if (sub_packet_size > packet_bytes) {
01256             if (packet->type != 10 && packet->type != 11 && packet->type != 12)
01257                 break;
01258             packet->size += packet_bytes - sub_packet_size;
01259         }
01260 
01261         packet_bytes -= sub_packet_size;
01262 
01263         /* add subpacket to 'all subpackets' list */
01264         q->sub_packet_list_A[i].packet = packet;
01265 
01266         /* add subpacket to related list */
01267         if (packet->type == 8) {
01268             SAMPLES_NEEDED_2("packet type 8");
01269             return;
01270         } else if (packet->type >= 9 && packet->type <= 12) {
01271             /* packets for MPEG Audio like Synthesis Filter */
01272             QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
01273         } else if (packet->type == 13) {
01274             for (j = 0; j < 6; j++)
01275                 q->fft_level_exp[j] = get_bits(&gb, 6);
01276         } else if (packet->type == 14) {
01277             for (j = 0; j < 6; j++)
01278                 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
01279         } else if (packet->type == 15) {
01280             SAMPLES_NEEDED_2("packet type 15")
01281             return;
01282         } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
01283             /* packets for FFT */
01284             QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
01285         }
01286     } // Packet bytes loop
01287 
01288 /* **************************************************************** */
01289     if (q->sub_packet_list_D[0].packet != NULL) {
01290         process_synthesis_subpackets(q, q->sub_packet_list_D);
01291         q->do_synth_filter = 1;
01292     } else if (q->do_synth_filter) {
01293         process_subpacket_10(q, NULL, 0);
01294         process_subpacket_11(q, NULL, 0);
01295         process_subpacket_12(q, NULL, 0);
01296     }
01297 /* **************************************************************** */
01298 }
01299 
01300 
01301 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
01302                        int offset, int duration, int channel,
01303                        int exp, int phase)
01304 {
01305     if (q->fft_coefs_min_index[duration] < 0)
01306         q->fft_coefs_min_index[duration] = q->fft_coefs_index;
01307 
01308     q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
01309     q->fft_coefs[q->fft_coefs_index].channel = channel;
01310     q->fft_coefs[q->fft_coefs_index].offset = offset;
01311     q->fft_coefs[q->fft_coefs_index].exp = exp;
01312     q->fft_coefs[q->fft_coefs_index].phase = phase;
01313     q->fft_coefs_index++;
01314 }
01315 
01316 
01317 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
01318 {
01319     int channel, stereo, phase, exp;
01320     int local_int_4,  local_int_8,  stereo_phase,  local_int_10;
01321     int local_int_14, stereo_exp, local_int_20, local_int_28;
01322     int n, offset;
01323 
01324     local_int_4 = 0;
01325     local_int_28 = 0;
01326     local_int_20 = 2;
01327     local_int_8 = (4 - duration);
01328     local_int_10 = 1 << (q->group_order - duration - 1);
01329     offset = 1;
01330 
01331     while (get_bits_left(gb)>0) {
01332         if (q->superblocktype_2_3) {
01333             while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
01334                 offset = 1;
01335                 if (n == 0) {
01336                     local_int_4 += local_int_10;
01337                     local_int_28 += (1 << local_int_8);
01338                 } else {
01339                     local_int_4 += 8*local_int_10;
01340                     local_int_28 += (8 << local_int_8);
01341                 }
01342             }
01343             offset += (n - 2);
01344         } else {
01345             offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
01346             while (offset >= (local_int_10 - 1)) {
01347                 offset += (1 - (local_int_10 - 1));
01348                 local_int_4  += local_int_10;
01349                 local_int_28 += (1 << local_int_8);
01350             }
01351         }
01352 
01353         if (local_int_4 >= q->group_size)
01354             return;
01355 
01356         local_int_14 = (offset >> local_int_8);
01357         if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
01358             return;
01359 
01360         if (q->nb_channels > 1) {
01361             channel = get_bits1(gb);
01362             stereo = get_bits1(gb);
01363         } else {
01364             channel = 0;
01365             stereo = 0;
01366         }
01367 
01368         exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
01369         exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
01370         exp = (exp < 0) ? 0 : exp;
01371 
01372         phase = get_bits(gb, 3);
01373         stereo_exp = 0;
01374         stereo_phase = 0;
01375 
01376         if (stereo) {
01377             stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
01378             stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
01379             if (stereo_phase < 0)
01380                 stereo_phase += 8;
01381         }
01382 
01383         if (q->frequency_range > (local_int_14 + 1)) {
01384             int sub_packet = (local_int_20 + local_int_28);
01385 
01386             qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
01387             if (stereo)
01388                 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
01389         }
01390 
01391         offset++;
01392     }
01393 }
01394 
01395 
01396 static void qdm2_decode_fft_packets (QDM2Context *q)
01397 {
01398     int i, j, min, max, value, type, unknown_flag;
01399     GetBitContext gb;
01400 
01401     if (q->sub_packet_list_B[0].packet == NULL)
01402         return;
01403 
01404     /* reset minimum indexes for FFT coefficients */
01405     q->fft_coefs_index = 0;
01406     for (i=0; i < 5; i++)
01407         q->fft_coefs_min_index[i] = -1;
01408 
01409     /* process subpackets ordered by type, largest type first */
01410     for (i = 0, max = 256; i < q->sub_packets_B; i++) {
01411         QDM2SubPacket *packet= NULL;
01412 
01413         /* find subpacket with largest type less than max */
01414         for (j = 0, min = 0; j < q->sub_packets_B; j++) {
01415             value = q->sub_packet_list_B[j].packet->type;
01416             if (value > min && value < max) {
01417                 min = value;
01418                 packet = q->sub_packet_list_B[j].packet;
01419             }
01420         }
01421 
01422         max = min;
01423 
01424         /* check for errors (?) */
01425         if (!packet)
01426             return;
01427 
01428         if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
01429             return;
01430 
01431         /* decode FFT tones */
01432         init_get_bits (&gb, packet->data, packet->size*8);
01433 
01434         if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
01435             unknown_flag = 1;
01436         else
01437             unknown_flag = 0;
01438 
01439         type = packet->type;
01440 
01441         if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
01442             int duration = q->sub_sampling + 5 - (type & 15);
01443 
01444             if (duration >= 0 && duration < 4)
01445                 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
01446         } else if (type == 31) {
01447             for (j=0; j < 4; j++)
01448                 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
01449         } else if (type == 46) {
01450             for (j=0; j < 6; j++)
01451                 q->fft_level_exp[j] = get_bits(&gb, 6);
01452             for (j=0; j < 4; j++)
01453             qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
01454         }
01455     } // Loop on B packets
01456 
01457     /* calculate maximum indexes for FFT coefficients */
01458     for (i = 0, j = -1; i < 5; i++)
01459         if (q->fft_coefs_min_index[i] >= 0) {
01460             if (j >= 0)
01461                 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
01462             j = i;
01463         }
01464     if (j >= 0)
01465         q->fft_coefs_max_index[j] = q->fft_coefs_index;
01466 }
01467 
01468 
01469 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
01470 {
01471    float level, f[6];
01472    int i;
01473    QDM2Complex c;
01474    const double iscale = 2.0*M_PI / 512.0;
01475 
01476     tone->phase += tone->phase_shift;
01477 
01478     /* calculate current level (maximum amplitude) of tone */
01479     level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
01480     c.im = level * sin(tone->phase*iscale);
01481     c.re = level * cos(tone->phase*iscale);
01482 
01483     /* generate FFT coefficients for tone */
01484     if (tone->duration >= 3 || tone->cutoff >= 3) {
01485         tone->complex[0].im += c.im;
01486         tone->complex[0].re += c.re;
01487         tone->complex[1].im -= c.im;
01488         tone->complex[1].re -= c.re;
01489     } else {
01490         f[1] = -tone->table[4];
01491         f[0] =  tone->table[3] - tone->table[0];
01492         f[2] =  1.0 - tone->table[2] - tone->table[3];
01493         f[3] =  tone->table[1] + tone->table[4] - 1.0;
01494         f[4] =  tone->table[0] - tone->table[1];
01495         f[5] =  tone->table[2];
01496         for (i = 0; i < 2; i++) {
01497             tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
01498             tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
01499         }
01500         for (i = 0; i < 4; i++) {
01501             tone->complex[i].re += c.re * f[i+2];
01502             tone->complex[i].im += c.im * f[i+2];
01503         }
01504     }
01505 
01506     /* copy the tone if it has not yet died out */
01507     if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
01508       memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
01509       q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
01510     }
01511 }
01512 
01513 
01514 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
01515 {
01516     int i, j, ch;
01517     const double iscale = 0.25 * M_PI;
01518 
01519     for (ch = 0; ch < q->channels; ch++) {
01520         memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
01521     }
01522 
01523 
01524     /* apply FFT tones with duration 4 (1 FFT period) */
01525     if (q->fft_coefs_min_index[4] >= 0)
01526         for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
01527             float level;
01528             QDM2Complex c;
01529 
01530             if (q->fft_coefs[i].sub_packet != sub_packet)
01531                 break;
01532 
01533             ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
01534             level = (q->fft_coefs[i].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[i].exp & 63];
01535 
01536             c.re = level * cos(q->fft_coefs[i].phase * iscale);
01537             c.im = level * sin(q->fft_coefs[i].phase * iscale);
01538             q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
01539             q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
01540             q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
01541             q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
01542         }
01543 
01544     /* generate existing FFT tones */
01545     for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
01546         qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
01547         q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
01548     }
01549 
01550     /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
01551     for (i = 0; i < 4; i++)
01552         if (q->fft_coefs_min_index[i] >= 0) {
01553             for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
01554                 int offset, four_i;
01555                 FFTTone tone;
01556 
01557                 if (q->fft_coefs[j].sub_packet != sub_packet)
01558                     break;
01559 
01560                 four_i = (4 - i);
01561                 offset = q->fft_coefs[j].offset >> four_i;
01562                 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
01563 
01564                 if (offset < q->frequency_range) {
01565                     if (offset < 2)
01566                         tone.cutoff = offset;
01567                     else
01568                         tone.cutoff = (offset >= 60) ? 3 : 2;
01569 
01570                     tone.level = (q->fft_coefs[j].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[j].exp & 63];
01571                     tone.complex = &q->fft.complex[ch][offset];
01572                     tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
01573                     tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
01574                     tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
01575                     tone.duration = i;
01576                     tone.time_index = 0;
01577 
01578                     qdm2_fft_generate_tone(q, &tone);
01579                 }
01580             }
01581             q->fft_coefs_min_index[i] = j;
01582         }
01583 }
01584 
01585 
01586 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
01587 {
01588     const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
01589     int i;
01590     q->fft.complex[channel][0].re *= 2.0f;
01591     q->fft.complex[channel][0].im = 0.0f;
01592     q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
01593     /* add samples to output buffer */
01594     for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
01595         q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
01596 }
01597 
01598 
01603 static void qdm2_synthesis_filter (QDM2Context *q, int index)
01604 {
01605     int i, k, ch, sb_used, sub_sampling, dither_state = 0;
01606 
01607     /* copy sb_samples */
01608     sb_used = QDM2_SB_USED(q->sub_sampling);
01609 
01610     for (ch = 0; ch < q->channels; ch++)
01611         for (i = 0; i < 8; i++)
01612             for (k=sb_used; k < SBLIMIT; k++)
01613                 q->sb_samples[ch][(8 * index) + i][k] = 0;
01614 
01615     for (ch = 0; ch < q->nb_channels; ch++) {
01616         float *samples_ptr = q->samples + ch;
01617 
01618         for (i = 0; i < 8; i++) {
01619             ff_mpa_synth_filter_float(&q->mpadsp,
01620                 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
01621                 ff_mpa_synth_window_float, &dither_state,
01622                 samples_ptr, q->nb_channels,
01623                 q->sb_samples[ch][(8 * index) + i]);
01624             samples_ptr += 32 * q->nb_channels;
01625         }
01626     }
01627 
01628     /* add samples to output buffer */
01629     sub_sampling = (4 >> q->sub_sampling);
01630 
01631     for (ch = 0; ch < q->channels; ch++)
01632         for (i = 0; i < q->frame_size; i++)
01633             q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
01634 }
01635 
01636 
01642 static av_cold void qdm2_init(QDM2Context *q) {
01643     static int initialized = 0;
01644 
01645     if (initialized != 0)
01646         return;
01647     initialized = 1;
01648 
01649     qdm2_init_vlc();
01650     ff_mpa_synth_init_float(ff_mpa_synth_window_float);
01651     softclip_table_init();
01652     rnd_table_init();
01653     init_noise_samples();
01654 
01655     av_log(NULL, AV_LOG_DEBUG, "init done\n");
01656 }
01657 
01658 
01659 #if 0
01660 static void dump_context(QDM2Context *q)
01661 {
01662     int i;
01663 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
01664     PRINT("compressed_data",q->compressed_data);
01665     PRINT("compressed_size",q->compressed_size);
01666     PRINT("frame_size",q->frame_size);
01667     PRINT("checksum_size",q->checksum_size);
01668     PRINT("channels",q->channels);
01669     PRINT("nb_channels",q->nb_channels);
01670     PRINT("fft_frame_size",q->fft_frame_size);
01671     PRINT("fft_size",q->fft_size);
01672     PRINT("sub_sampling",q->sub_sampling);
01673     PRINT("fft_order",q->fft_order);
01674     PRINT("group_order",q->group_order);
01675     PRINT("group_size",q->group_size);
01676     PRINT("sub_packet",q->sub_packet);
01677     PRINT("frequency_range",q->frequency_range);
01678     PRINT("has_errors",q->has_errors);
01679     PRINT("fft_tone_end",q->fft_tone_end);
01680     PRINT("fft_tone_start",q->fft_tone_start);
01681     PRINT("fft_coefs_index",q->fft_coefs_index);
01682     PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
01683     PRINT("cm_table_select",q->cm_table_select);
01684     PRINT("noise_idx",q->noise_idx);
01685 
01686     for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
01687     {
01688     FFTTone *t = &q->fft_tones[i];
01689 
01690     av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
01691     av_log(NULL,AV_LOG_DEBUG,"  level = %f\n", t->level);
01692 //  PRINT(" level", t->level);
01693     PRINT(" phase", t->phase);
01694     PRINT(" phase_shift", t->phase_shift);
01695     PRINT(" duration", t->duration);
01696     PRINT(" samples_im", t->samples_im);
01697     PRINT(" samples_re", t->samples_re);
01698     PRINT(" table", t->table);
01699     }
01700 
01701 }
01702 #endif
01703 
01704 
01708 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
01709 {
01710     QDM2Context *s = avctx->priv_data;
01711     uint8_t *extradata;
01712     int extradata_size;
01713     int tmp_val, tmp, size;
01714 
01715     /* extradata parsing
01716 
01717     Structure:
01718     wave {
01719         frma (QDM2)
01720         QDCA
01721         QDCP
01722     }
01723 
01724     32  size (including this field)
01725     32  tag (=frma)
01726     32  type (=QDM2 or QDMC)
01727 
01728     32  size (including this field, in bytes)
01729     32  tag (=QDCA) // maybe mandatory parameters
01730     32  unknown (=1)
01731     32  channels (=2)
01732     32  samplerate (=44100)
01733     32  bitrate (=96000)
01734     32  block size (=4096)
01735     32  frame size (=256) (for one channel)
01736     32  packet size (=1300)
01737 
01738     32  size (including this field, in bytes)
01739     32  tag (=QDCP) // maybe some tuneable parameters
01740     32  float1 (=1.0)
01741     32  zero ?
01742     32  float2 (=1.0)
01743     32  float3 (=1.0)
01744     32  unknown (27)
01745     32  unknown (8)
01746     32  zero ?
01747     */
01748 
01749     if (!avctx->extradata || (avctx->extradata_size < 48)) {
01750         av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
01751         return -1;
01752     }
01753 
01754     extradata = avctx->extradata;
01755     extradata_size = avctx->extradata_size;
01756 
01757     while (extradata_size > 7) {
01758         if (!memcmp(extradata, "frmaQDM", 7))
01759             break;
01760         extradata++;
01761         extradata_size--;
01762     }
01763 
01764     if (extradata_size < 12) {
01765         av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
01766                extradata_size);
01767         return -1;
01768     }
01769 
01770     if (memcmp(extradata, "frmaQDM", 7)) {
01771         av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
01772         return -1;
01773     }
01774 
01775     if (extradata[7] == 'C') {
01776 //        s->is_qdmc = 1;
01777         av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
01778         return -1;
01779     }
01780 
01781     extradata += 8;
01782     extradata_size -= 8;
01783 
01784     size = AV_RB32(extradata);
01785 
01786     if(size > extradata_size){
01787         av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
01788                extradata_size, size);
01789         return -1;
01790     }
01791 
01792     extradata += 4;
01793     av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
01794     if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
01795         av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
01796         return -1;
01797     }
01798 
01799     extradata += 8;
01800 
01801     avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
01802     extradata += 4;
01803     if (s->channels > MPA_MAX_CHANNELS)
01804         return AVERROR_INVALIDDATA;
01805 
01806     avctx->sample_rate = AV_RB32(extradata);
01807     extradata += 4;
01808 
01809     avctx->bit_rate = AV_RB32(extradata);
01810     extradata += 4;
01811 
01812     s->group_size = AV_RB32(extradata);
01813     extradata += 4;
01814 
01815     s->fft_size = AV_RB32(extradata);
01816     extradata += 4;
01817 
01818     s->checksum_size = AV_RB32(extradata);
01819     if (s->checksum_size >= 1U << 28) {
01820         av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
01821         return AVERROR_INVALIDDATA;
01822     }
01823 
01824     s->fft_order = av_log2(s->fft_size) + 1;
01825     s->fft_frame_size = 2 * s->fft_size; // complex has two floats
01826 
01827     // something like max decodable tones
01828     s->group_order = av_log2(s->group_size) + 1;
01829     s->frame_size = s->group_size / 16; // 16 iterations per super block
01830 
01831     if (s->frame_size > QDM2_MAX_FRAME_SIZE)
01832         return AVERROR_INVALIDDATA;
01833 
01834     s->sub_sampling = s->fft_order - 7;
01835     s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
01836 
01837     switch ((s->sub_sampling * 2 + s->channels - 1)) {
01838         case 0: tmp = 40; break;
01839         case 1: tmp = 48; break;
01840         case 2: tmp = 56; break;
01841         case 3: tmp = 72; break;
01842         case 4: tmp = 80; break;
01843         case 5: tmp = 100;break;
01844         default: tmp=s->sub_sampling; break;
01845     }
01846     tmp_val = 0;
01847     if ((tmp * 1000) < avctx->bit_rate)  tmp_val = 1;
01848     if ((tmp * 1440) < avctx->bit_rate)  tmp_val = 2;
01849     if ((tmp * 1760) < avctx->bit_rate)  tmp_val = 3;
01850     if ((tmp * 2240) < avctx->bit_rate)  tmp_val = 4;
01851     s->cm_table_select = tmp_val;
01852 
01853     if (s->sub_sampling == 0)
01854         tmp = 7999;
01855     else
01856         tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
01857     /*
01858     0: 7999 -> 0
01859     1: 20000 -> 2
01860     2: 28000 -> 2
01861     */
01862     if (tmp < 8000)
01863         s->coeff_per_sb_select = 0;
01864     else if (tmp <= 16000)
01865         s->coeff_per_sb_select = 1;
01866     else
01867         s->coeff_per_sb_select = 2;
01868 
01869     // Fail on unknown fft order
01870     if ((s->fft_order < 7) || (s->fft_order > 9)) {
01871         av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
01872         return -1;
01873     }
01874 
01875     ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
01876     ff_mpadsp_init(&s->mpadsp);
01877 
01878     qdm2_init(s);
01879 
01880     avctx->sample_fmt = AV_SAMPLE_FMT_S16;
01881 
01882 //    dump_context(s);
01883     return 0;
01884 }
01885 
01886 
01887 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
01888 {
01889     QDM2Context *s = avctx->priv_data;
01890 
01891     ff_rdft_end(&s->rdft_ctx);
01892 
01893     return 0;
01894 }
01895 
01896 
01897 static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
01898 {
01899     int ch, i;
01900     const int frame_size = (q->frame_size * q->channels);
01901 
01902     if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
01903         return -1;
01904 
01905     /* select input buffer */
01906     q->compressed_data = in;
01907     q->compressed_size = q->checksum_size;
01908 
01909 //  dump_context(q);
01910 
01911     /* copy old block, clear new block of output samples */
01912     memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
01913     memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
01914 
01915     /* decode block of QDM2 compressed data */
01916     if (q->sub_packet == 0) {
01917         q->has_errors = 0; // zero it for a new super block
01918         av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
01919         qdm2_decode_super_block(q);
01920     }
01921 
01922     /* parse subpackets */
01923     if (!q->has_errors) {
01924         if (q->sub_packet == 2)
01925             qdm2_decode_fft_packets(q);
01926 
01927         qdm2_fft_tone_synthesizer(q, q->sub_packet);
01928     }
01929 
01930     /* sound synthesis stage 1 (FFT) */
01931     for (ch = 0; ch < q->channels; ch++) {
01932         qdm2_calculate_fft(q, ch, q->sub_packet);
01933 
01934         if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
01935             SAMPLES_NEEDED_2("has errors, and C list is not empty")
01936             return -1;
01937         }
01938     }
01939 
01940     /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
01941     if (!q->has_errors && q->do_synth_filter)
01942         qdm2_synthesis_filter(q, q->sub_packet);
01943 
01944     q->sub_packet = (q->sub_packet + 1) % 16;
01945 
01946     /* clip and convert output float[] to 16bit signed samples */
01947     for (i = 0; i < frame_size; i++) {
01948         int value = (int)q->output_buffer[i];
01949 
01950         if (value > SOFTCLIP_THRESHOLD)
01951             value = (value >  HARDCLIP_THRESHOLD) ?  32767 :  softclip_table[ value - SOFTCLIP_THRESHOLD];
01952         else if (value < -SOFTCLIP_THRESHOLD)
01953             value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
01954 
01955         out[i] = value;
01956     }
01957 
01958     return 0;
01959 }
01960 
01961 
01962 static int qdm2_decode_frame(AVCodecContext *avctx,
01963             void *data, int *data_size,
01964             AVPacket *avpkt)
01965 {
01966     const uint8_t *buf = avpkt->data;
01967     int buf_size = avpkt->size;
01968     QDM2Context *s = avctx->priv_data;
01969     int16_t *out = data;
01970     int i, out_size;
01971 
01972     if(!buf)
01973         return 0;
01974     if(buf_size < s->checksum_size)
01975         return -1;
01976 
01977     out_size = 16 * s->channels * s->frame_size *
01978                av_get_bytes_per_sample(avctx->sample_fmt);
01979     if (*data_size < out_size) {
01980         av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
01981         return AVERROR(EINVAL);
01982     }
01983 
01984     av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
01985        buf_size, buf, s->checksum_size, data, *data_size);
01986 
01987     for (i = 0; i < 16; i++) {
01988         if (qdm2_decode(s, buf, out) < 0)
01989             return -1;
01990         out += s->channels * s->frame_size;
01991     }
01992 
01993     *data_size = out_size;
01994 
01995     return s->checksum_size;
01996 }
01997 
01998 AVCodec ff_qdm2_decoder =
01999 {
02000     .name = "qdm2",
02001     .type = AVMEDIA_TYPE_AUDIO,
02002     .id = CODEC_ID_QDM2,
02003     .priv_data_size = sizeof(QDM2Context),
02004     .init = qdm2_decode_init,
02005     .close = qdm2_decode_close,
02006     .decode = qdm2_decode_frame,
02007     .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
02008 };

Generated on Wed Apr 11 2012 07:31:34 for FFmpeg by  doxygen 1.7.1