Libav 0.7.1
libavcodec/qdm2.c
Go to the documentation of this file.
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 Libav.
00009  *
00010  * Libav 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  * Libav 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 Libav; 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 * 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                             unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
00886                             if (index < FF_ARRAY_ELEMS(type30_dequant)) {
00887                                 samples[0] = type30_dequant[index];
00888                             } else
00889                                 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00890                         } else
00891                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00892 
00893                         run = 1;
00894                         break;
00895 
00896                     case 34:
00897                         if (BITS_LEFT(length,gb) >= 7) {
00898                             if (type34_first) {
00899                                 type34_div = (float)(1 << get_bits(gb, 2));
00900                                 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
00901                                 type34_predictor = samples[0];
00902                                 type34_first = 0;
00903                             } else {
00904                                 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
00905                                 if (index < FF_ARRAY_ELEMS(type34_delta)) {
00906                                     samples[0] = type34_delta[index] / type34_div + type34_predictor;
00907                                     type34_predictor = samples[0];
00908                                 } else
00909                                     samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00910                             }
00911                         } else {
00912                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00913                         }
00914                         run = 1;
00915                         break;
00916 
00917                     default:
00918                         samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00919                         run = 1;
00920                         break;
00921                 }
00922 
00923                 if (joined_stereo) {
00924                     float tmp[10][MPA_MAX_CHANNELS];
00925 
00926                     for (k = 0; k < run; k++) {
00927                         tmp[k][0] = samples[k];
00928                         tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
00929                     }
00930                     for (chs = 0; chs < q->nb_channels; chs++)
00931                         for (k = 0; k < run; k++)
00932                             if ((j + k) < 128)
00933                                 q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs];
00934                 } else {
00935                     for (k = 0; k < run; k++)
00936                         if ((j + k) < 128)
00937                             q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
00938                 }
00939 
00940                 j += run;
00941             } // j loop
00942         } // channel loop
00943     } // subband loop
00944 }
00945 
00946 
00956 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
00957 {
00958     int i, k, run, level, diff;
00959 
00960     if (BITS_LEFT(length,gb) < 16)
00961         return;
00962     level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
00963 
00964     quantized_coeffs[0] = level;
00965 
00966     for (i = 0; i < 7; ) {
00967         if (BITS_LEFT(length,gb) < 16)
00968             break;
00969         run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
00970 
00971         if (BITS_LEFT(length,gb) < 16)
00972             break;
00973         diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
00974 
00975         for (k = 1; k <= run; k++)
00976             quantized_coeffs[i + k] = (level + ((k * diff) / run));
00977 
00978         level += diff;
00979         i += run;
00980     }
00981 }
00982 
00983 
00993 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
00994 {
00995     int sb, j, k, n, ch;
00996 
00997     for (ch = 0; ch < q->nb_channels; ch++) {
00998         init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
00999 
01000         if (BITS_LEFT(length,gb) < 16) {
01001             memset(q->quantized_coeffs[ch][0], 0, 8);
01002             break;
01003         }
01004     }
01005 
01006     n = q->sub_sampling + 1;
01007 
01008     for (sb = 0; sb < n; sb++)
01009         for (ch = 0; ch < q->nb_channels; ch++)
01010             for (j = 0; j < 8; j++) {
01011                 if (BITS_LEFT(length,gb) < 1)
01012                     break;
01013                 if (get_bits1(gb)) {
01014                     for (k=0; k < 8; k++) {
01015                         if (BITS_LEFT(length,gb) < 16)
01016                             break;
01017                         q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
01018                     }
01019                 } else {
01020                     for (k=0; k < 8; k++)
01021                         q->tone_level_idx_hi1[ch][sb][j][k] = 0;
01022                 }
01023             }
01024 
01025     n = QDM2_SB_USED(q->sub_sampling) - 4;
01026 
01027     for (sb = 0; sb < n; sb++)
01028         for (ch = 0; ch < q->nb_channels; ch++) {
01029             if (BITS_LEFT(length,gb) < 16)
01030                 break;
01031             q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
01032             if (sb > 19)
01033                 q->tone_level_idx_hi2[ch][sb] -= 16;
01034             else
01035                 for (j = 0; j < 8; j++)
01036                     q->tone_level_idx_mid[ch][sb][j] = -16;
01037         }
01038 
01039     n = QDM2_SB_USED(q->sub_sampling) - 5;
01040 
01041     for (sb = 0; sb < n; sb++)
01042         for (ch = 0; ch < q->nb_channels; ch++)
01043             for (j = 0; j < 8; j++) {
01044                 if (BITS_LEFT(length,gb) < 16)
01045                     break;
01046                 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
01047             }
01048 }
01049 
01056 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
01057 {
01058     GetBitContext gb;
01059     int i, j, k, n, ch, run, level, diff;
01060 
01061     init_get_bits(&gb, node->packet->data, node->packet->size*8);
01062 
01063     n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
01064 
01065     for (i = 1; i < n; i++)
01066         for (ch=0; ch < q->nb_channels; ch++) {
01067             level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
01068             q->quantized_coeffs[ch][i][0] = level;
01069 
01070             for (j = 0; j < (8 - 1); ) {
01071                 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
01072                 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
01073 
01074                 for (k = 1; k <= run; k++)
01075                     q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
01076 
01077                 level += diff;
01078                 j += run;
01079             }
01080         }
01081 
01082     for (ch = 0; ch < q->nb_channels; ch++)
01083         for (i = 0; i < 8; i++)
01084             q->quantized_coeffs[ch][0][i] = 0;
01085 }
01086 
01087 
01095 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
01096 {
01097     GetBitContext gb;
01098 
01099     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
01100 
01101     if (length != 0) {
01102         init_tone_level_dequantization(q, &gb, length);
01103         fill_tone_level_array(q, 1);
01104     } else {
01105         fill_tone_level_array(q, 0);
01106     }
01107 }
01108 
01109 
01117 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
01118 {
01119     GetBitContext gb;
01120 
01121     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
01122     if (length >= 32) {
01123         int c = get_bits (&gb, 13);
01124 
01125         if (c > 3)
01126             fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
01127                                       q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
01128     }
01129 
01130     synthfilt_build_sb_samples(q, &gb, length, 0, 8);
01131 }
01132 
01133 
01141 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
01142 {
01143     GetBitContext gb;
01144 
01145     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
01146     synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
01147 }
01148 
01149 /*
01150  * Process new subpackets for synthesis filter
01151  *
01152  * @param q       context
01153  * @param list    list with synthesis filter packets (list D)
01154  */
01155 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
01156 {
01157     QDM2SubPNode *nodes[4];
01158 
01159     nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
01160     if (nodes[0] != NULL)
01161         process_subpacket_9(q, nodes[0]);
01162 
01163     nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
01164     if (nodes[1] != NULL)
01165         process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
01166     else
01167         process_subpacket_10(q, NULL, 0);
01168 
01169     nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
01170     if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
01171         process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
01172     else
01173         process_subpacket_11(q, NULL, 0);
01174 
01175     nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
01176     if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
01177         process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
01178     else
01179         process_subpacket_12(q, NULL, 0);
01180 }
01181 
01182 
01183 /*
01184  * Decode superblock, fill packet lists.
01185  *
01186  * @param q    context
01187  */
01188 static void qdm2_decode_super_block (QDM2Context *q)
01189 {
01190     GetBitContext gb;
01191     QDM2SubPacket header, *packet;
01192     int i, packet_bytes, sub_packet_size, sub_packets_D;
01193     unsigned int next_index = 0;
01194 
01195     memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
01196     memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
01197     memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
01198 
01199     q->sub_packets_B = 0;
01200     sub_packets_D = 0;
01201 
01202     average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
01203 
01204     init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
01205     qdm2_decode_sub_packet_header(&gb, &header);
01206 
01207     if (header.type < 2 || header.type >= 8) {
01208         q->has_errors = 1;
01209         av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
01210         return;
01211     }
01212 
01213     q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
01214     packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
01215 
01216     init_get_bits(&gb, header.data, header.size*8);
01217 
01218     if (header.type == 2 || header.type == 4 || header.type == 5) {
01219         int csum  = 257 * get_bits(&gb, 8);
01220             csum +=   2 * get_bits(&gb, 8);
01221 
01222         csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
01223 
01224         if (csum != 0) {
01225             q->has_errors = 1;
01226             av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
01227             return;
01228         }
01229     }
01230 
01231     q->sub_packet_list_B[0].packet = NULL;
01232     q->sub_packet_list_D[0].packet = NULL;
01233 
01234     for (i = 0; i < 6; i++)
01235         if (--q->fft_level_exp[i] < 0)
01236             q->fft_level_exp[i] = 0;
01237 
01238     for (i = 0; packet_bytes > 0; i++) {
01239         int j;
01240 
01241         q->sub_packet_list_A[i].next = NULL;
01242 
01243         if (i > 0) {
01244             q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
01245 
01246             /* seek to next block */
01247             init_get_bits(&gb, header.data, header.size*8);
01248             skip_bits(&gb, next_index*8);
01249 
01250             if (next_index >= header.size)
01251                 break;
01252         }
01253 
01254         /* decode subpacket */
01255         packet = &q->sub_packets[i];
01256         qdm2_decode_sub_packet_header(&gb, packet);
01257         next_index = packet->size + get_bits_count(&gb) / 8;
01258         sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
01259 
01260         if (packet->type == 0)
01261             break;
01262 
01263         if (sub_packet_size > packet_bytes) {
01264             if (packet->type != 10 && packet->type != 11 && packet->type != 12)
01265                 break;
01266             packet->size += packet_bytes - sub_packet_size;
01267         }
01268 
01269         packet_bytes -= sub_packet_size;
01270 
01271         /* add subpacket to 'all subpackets' list */
01272         q->sub_packet_list_A[i].packet = packet;
01273 
01274         /* add subpacket to related list */
01275         if (packet->type == 8) {
01276             SAMPLES_NEEDED_2("packet type 8");
01277             return;
01278         } else if (packet->type >= 9 && packet->type <= 12) {
01279             /* packets for MPEG Audio like Synthesis Filter */
01280             QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
01281         } else if (packet->type == 13) {
01282             for (j = 0; j < 6; j++)
01283                 q->fft_level_exp[j] = get_bits(&gb, 6);
01284         } else if (packet->type == 14) {
01285             for (j = 0; j < 6; j++)
01286                 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
01287         } else if (packet->type == 15) {
01288             SAMPLES_NEEDED_2("packet type 15")
01289             return;
01290         } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
01291             /* packets for FFT */
01292             QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
01293         }
01294     } // Packet bytes loop
01295 
01296 /* **************************************************************** */
01297     if (q->sub_packet_list_D[0].packet != NULL) {
01298         process_synthesis_subpackets(q, q->sub_packet_list_D);
01299         q->do_synth_filter = 1;
01300     } else if (q->do_synth_filter) {
01301         process_subpacket_10(q, NULL, 0);
01302         process_subpacket_11(q, NULL, 0);
01303         process_subpacket_12(q, NULL, 0);
01304     }
01305 /* **************************************************************** */
01306 }
01307 
01308 
01309 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
01310                        int offset, int duration, int channel,
01311                        int exp, int phase)
01312 {
01313     if (q->fft_coefs_min_index[duration] < 0)
01314         q->fft_coefs_min_index[duration] = q->fft_coefs_index;
01315 
01316     q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
01317     q->fft_coefs[q->fft_coefs_index].channel = channel;
01318     q->fft_coefs[q->fft_coefs_index].offset = offset;
01319     q->fft_coefs[q->fft_coefs_index].exp = exp;
01320     q->fft_coefs[q->fft_coefs_index].phase = phase;
01321     q->fft_coefs_index++;
01322 }
01323 
01324 
01325 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
01326 {
01327     int channel, stereo, phase, exp;
01328     int local_int_4,  local_int_8,  stereo_phase,  local_int_10;
01329     int local_int_14, stereo_exp, local_int_20, local_int_28;
01330     int n, offset;
01331 
01332     local_int_4 = 0;
01333     local_int_28 = 0;
01334     local_int_20 = 2;
01335     local_int_8 = (4 - duration);
01336     local_int_10 = 1 << (q->group_order - duration - 1);
01337     offset = 1;
01338 
01339     while (1) {
01340         if (q->superblocktype_2_3) {
01341             while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
01342                 offset = 1;
01343                 if (n == 0) {
01344                     local_int_4 += local_int_10;
01345                     local_int_28 += (1 << local_int_8);
01346                 } else {
01347                     local_int_4 += 8*local_int_10;
01348                     local_int_28 += (8 << local_int_8);
01349                 }
01350             }
01351             offset += (n - 2);
01352         } else {
01353             offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
01354             while (offset >= (local_int_10 - 1)) {
01355                 offset += (1 - (local_int_10 - 1));
01356                 local_int_4  += local_int_10;
01357                 local_int_28 += (1 << local_int_8);
01358             }
01359         }
01360 
01361         if (local_int_4 >= q->group_size)
01362             return;
01363 
01364         local_int_14 = (offset >> local_int_8);
01365         if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
01366             return;
01367 
01368         if (q->nb_channels > 1) {
01369             channel = get_bits1(gb);
01370             stereo = get_bits1(gb);
01371         } else {
01372             channel = 0;
01373             stereo = 0;
01374         }
01375 
01376         exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
01377         exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
01378         exp = (exp < 0) ? 0 : exp;
01379 
01380         phase = get_bits(gb, 3);
01381         stereo_exp = 0;
01382         stereo_phase = 0;
01383 
01384         if (stereo) {
01385             stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
01386             stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
01387             if (stereo_phase < 0)
01388                 stereo_phase += 8;
01389         }
01390 
01391         if (q->frequency_range > (local_int_14 + 1)) {
01392             int sub_packet = (local_int_20 + local_int_28);
01393 
01394             qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
01395             if (stereo)
01396                 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
01397         }
01398 
01399         offset++;
01400     }
01401 }
01402 
01403 
01404 static void qdm2_decode_fft_packets (QDM2Context *q)
01405 {
01406     int i, j, min, max, value, type, unknown_flag;
01407     GetBitContext gb;
01408 
01409     if (q->sub_packet_list_B[0].packet == NULL)
01410         return;
01411 
01412     /* reset minimum indexes for FFT coefficients */
01413     q->fft_coefs_index = 0;
01414     for (i=0; i < 5; i++)
01415         q->fft_coefs_min_index[i] = -1;
01416 
01417     /* process subpackets ordered by type, largest type first */
01418     for (i = 0, max = 256; i < q->sub_packets_B; i++) {
01419         QDM2SubPacket *packet= NULL;
01420 
01421         /* find subpacket with largest type less than max */
01422         for (j = 0, min = 0; j < q->sub_packets_B; j++) {
01423             value = q->sub_packet_list_B[j].packet->type;
01424             if (value > min && value < max) {
01425                 min = value;
01426                 packet = q->sub_packet_list_B[j].packet;
01427             }
01428         }
01429 
01430         max = min;
01431 
01432         /* check for errors (?) */
01433         if (!packet)
01434             return;
01435 
01436         if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
01437             return;
01438 
01439         /* decode FFT tones */
01440         init_get_bits (&gb, packet->data, packet->size*8);
01441 
01442         if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
01443             unknown_flag = 1;
01444         else
01445             unknown_flag = 0;
01446 
01447         type = packet->type;
01448 
01449         if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
01450             int duration = q->sub_sampling + 5 - (type & 15);
01451 
01452             if (duration >= 0 && duration < 4)
01453                 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
01454         } else if (type == 31) {
01455             for (j=0; j < 4; j++)
01456                 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
01457         } else if (type == 46) {
01458             for (j=0; j < 6; j++)
01459                 q->fft_level_exp[j] = get_bits(&gb, 6);
01460             for (j=0; j < 4; j++)
01461             qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
01462         }
01463     } // Loop on B packets
01464 
01465     /* calculate maximum indexes for FFT coefficients */
01466     for (i = 0, j = -1; i < 5; i++)
01467         if (q->fft_coefs_min_index[i] >= 0) {
01468             if (j >= 0)
01469                 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
01470             j = i;
01471         }
01472     if (j >= 0)
01473         q->fft_coefs_max_index[j] = q->fft_coefs_index;
01474 }
01475 
01476 
01477 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
01478 {
01479    float level, f[6];
01480    int i;
01481    QDM2Complex c;
01482    const double iscale = 2.0*M_PI / 512.0;
01483 
01484     tone->phase += tone->phase_shift;
01485 
01486     /* calculate current level (maximum amplitude) of tone */
01487     level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
01488     c.im = level * sin(tone->phase*iscale);
01489     c.re = level * cos(tone->phase*iscale);
01490 
01491     /* generate FFT coefficients for tone */
01492     if (tone->duration >= 3 || tone->cutoff >= 3) {
01493         tone->complex[0].im += c.im;
01494         tone->complex[0].re += c.re;
01495         tone->complex[1].im -= c.im;
01496         tone->complex[1].re -= c.re;
01497     } else {
01498         f[1] = -tone->table[4];
01499         f[0] =  tone->table[3] - tone->table[0];
01500         f[2] =  1.0 - tone->table[2] - tone->table[3];
01501         f[3] =  tone->table[1] + tone->table[4] - 1.0;
01502         f[4] =  tone->table[0] - tone->table[1];
01503         f[5] =  tone->table[2];
01504         for (i = 0; i < 2; i++) {
01505             tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
01506             tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
01507         }
01508         for (i = 0; i < 4; i++) {
01509             tone->complex[i].re += c.re * f[i+2];
01510             tone->complex[i].im += c.im * f[i+2];
01511         }
01512     }
01513 
01514     /* copy the tone if it has not yet died out */
01515     if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
01516       memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
01517       q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
01518     }
01519 }
01520 
01521 
01522 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
01523 {
01524     int i, j, ch;
01525     const double iscale = 0.25 * M_PI;
01526 
01527     for (ch = 0; ch < q->channels; ch++) {
01528         memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
01529     }
01530 
01531 
01532     /* apply FFT tones with duration 4 (1 FFT period) */
01533     if (q->fft_coefs_min_index[4] >= 0)
01534         for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
01535             float level;
01536             QDM2Complex c;
01537 
01538             if (q->fft_coefs[i].sub_packet != sub_packet)
01539                 break;
01540 
01541             ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
01542             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];
01543 
01544             c.re = level * cos(q->fft_coefs[i].phase * iscale);
01545             c.im = level * sin(q->fft_coefs[i].phase * iscale);
01546             q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
01547             q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
01548             q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
01549             q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
01550         }
01551 
01552     /* generate existing FFT tones */
01553     for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
01554         qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
01555         q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
01556     }
01557 
01558     /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
01559     for (i = 0; i < 4; i++)
01560         if (q->fft_coefs_min_index[i] >= 0) {
01561             for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
01562                 int offset, four_i;
01563                 FFTTone tone;
01564 
01565                 if (q->fft_coefs[j].sub_packet != sub_packet)
01566                     break;
01567 
01568                 four_i = (4 - i);
01569                 offset = q->fft_coefs[j].offset >> four_i;
01570                 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
01571 
01572                 if (offset < q->frequency_range) {
01573                     if (offset < 2)
01574                         tone.cutoff = offset;
01575                     else
01576                         tone.cutoff = (offset >= 60) ? 3 : 2;
01577 
01578                     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];
01579                     tone.complex = &q->fft.complex[ch][offset];
01580                     tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
01581                     tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
01582                     tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
01583                     tone.duration = i;
01584                     tone.time_index = 0;
01585 
01586                     qdm2_fft_generate_tone(q, &tone);
01587                 }
01588             }
01589             q->fft_coefs_min_index[i] = j;
01590         }
01591 }
01592 
01593 
01594 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
01595 {
01596     const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
01597     int i;
01598     q->fft.complex[channel][0].re *= 2.0f;
01599     q->fft.complex[channel][0].im = 0.0f;
01600     q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
01601     /* add samples to output buffer */
01602     for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
01603         q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
01604 }
01605 
01606 
01611 static void qdm2_synthesis_filter (QDM2Context *q, int index)
01612 {
01613     int i, k, ch, sb_used, sub_sampling, dither_state = 0;
01614 
01615     /* copy sb_samples */
01616     sb_used = QDM2_SB_USED(q->sub_sampling);
01617 
01618     for (ch = 0; ch < q->channels; ch++)
01619         for (i = 0; i < 8; i++)
01620             for (k=sb_used; k < SBLIMIT; k++)
01621                 q->sb_samples[ch][(8 * index) + i][k] = 0;
01622 
01623     for (ch = 0; ch < q->nb_channels; ch++) {
01624         float *samples_ptr = q->samples + ch;
01625 
01626         for (i = 0; i < 8; i++) {
01627             ff_mpa_synth_filter_float(&q->mpadsp,
01628                 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
01629                 ff_mpa_synth_window_float, &dither_state,
01630                 samples_ptr, q->nb_channels,
01631                 q->sb_samples[ch][(8 * index) + i]);
01632             samples_ptr += 32 * q->nb_channels;
01633         }
01634     }
01635 
01636     /* add samples to output buffer */
01637     sub_sampling = (4 >> q->sub_sampling);
01638 
01639     for (ch = 0; ch < q->channels; ch++)
01640         for (i = 0; i < q->frame_size; i++)
01641             q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
01642 }
01643 
01644 
01650 static av_cold void qdm2_init(QDM2Context *q) {
01651     static int initialized = 0;
01652 
01653     if (initialized != 0)
01654         return;
01655     initialized = 1;
01656 
01657     qdm2_init_vlc();
01658     ff_mpa_synth_init_float(ff_mpa_synth_window_float);
01659     softclip_table_init();
01660     rnd_table_init();
01661     init_noise_samples();
01662 
01663     av_log(NULL, AV_LOG_DEBUG, "init done\n");
01664 }
01665 
01666 
01667 #if 0
01668 static void dump_context(QDM2Context *q)
01669 {
01670     int i;
01671 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
01672     PRINT("compressed_data",q->compressed_data);
01673     PRINT("compressed_size",q->compressed_size);
01674     PRINT("frame_size",q->frame_size);
01675     PRINT("checksum_size",q->checksum_size);
01676     PRINT("channels",q->channels);
01677     PRINT("nb_channels",q->nb_channels);
01678     PRINT("fft_frame_size",q->fft_frame_size);
01679     PRINT("fft_size",q->fft_size);
01680     PRINT("sub_sampling",q->sub_sampling);
01681     PRINT("fft_order",q->fft_order);
01682     PRINT("group_order",q->group_order);
01683     PRINT("group_size",q->group_size);
01684     PRINT("sub_packet",q->sub_packet);
01685     PRINT("frequency_range",q->frequency_range);
01686     PRINT("has_errors",q->has_errors);
01687     PRINT("fft_tone_end",q->fft_tone_end);
01688     PRINT("fft_tone_start",q->fft_tone_start);
01689     PRINT("fft_coefs_index",q->fft_coefs_index);
01690     PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
01691     PRINT("cm_table_select",q->cm_table_select);
01692     PRINT("noise_idx",q->noise_idx);
01693 
01694     for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
01695     {
01696     FFTTone *t = &q->fft_tones[i];
01697 
01698     av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
01699     av_log(NULL,AV_LOG_DEBUG,"  level = %f\n", t->level);
01700 //  PRINT(" level", t->level);
01701     PRINT(" phase", t->phase);
01702     PRINT(" phase_shift", t->phase_shift);
01703     PRINT(" duration", t->duration);
01704     PRINT(" samples_im", t->samples_im);
01705     PRINT(" samples_re", t->samples_re);
01706     PRINT(" table", t->table);
01707     }
01708 
01709 }
01710 #endif
01711 
01712 
01716 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
01717 {
01718     QDM2Context *s = avctx->priv_data;
01719     uint8_t *extradata;
01720     int extradata_size;
01721     int tmp_val, tmp, size;
01722 
01723     /* extradata parsing
01724 
01725     Structure:
01726     wave {
01727         frma (QDM2)
01728         QDCA
01729         QDCP
01730     }
01731 
01732     32  size (including this field)
01733     32  tag (=frma)
01734     32  type (=QDM2 or QDMC)
01735 
01736     32  size (including this field, in bytes)
01737     32  tag (=QDCA) // maybe mandatory parameters
01738     32  unknown (=1)
01739     32  channels (=2)
01740     32  samplerate (=44100)
01741     32  bitrate (=96000)
01742     32  block size (=4096)
01743     32  frame size (=256) (for one channel)
01744     32  packet size (=1300)
01745 
01746     32  size (including this field, in bytes)
01747     32  tag (=QDCP) // maybe some tuneable parameters
01748     32  float1 (=1.0)
01749     32  zero ?
01750     32  float2 (=1.0)
01751     32  float3 (=1.0)
01752     32  unknown (27)
01753     32  unknown (8)
01754     32  zero ?
01755     */
01756 
01757     if (!avctx->extradata || (avctx->extradata_size < 48)) {
01758         av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
01759         return -1;
01760     }
01761 
01762     extradata = avctx->extradata;
01763     extradata_size = avctx->extradata_size;
01764 
01765     while (extradata_size > 7) {
01766         if (!memcmp(extradata, "frmaQDM", 7))
01767             break;
01768         extradata++;
01769         extradata_size--;
01770     }
01771 
01772     if (extradata_size < 12) {
01773         av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
01774                extradata_size);
01775         return -1;
01776     }
01777 
01778     if (memcmp(extradata, "frmaQDM", 7)) {
01779         av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
01780         return -1;
01781     }
01782 
01783     if (extradata[7] == 'C') {
01784 //        s->is_qdmc = 1;
01785         av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
01786         return -1;
01787     }
01788 
01789     extradata += 8;
01790     extradata_size -= 8;
01791 
01792     size = AV_RB32(extradata);
01793 
01794     if(size > extradata_size){
01795         av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
01796                extradata_size, size);
01797         return -1;
01798     }
01799 
01800     extradata += 4;
01801     av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
01802     if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
01803         av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
01804         return -1;
01805     }
01806 
01807     extradata += 8;
01808 
01809     avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
01810     extradata += 4;
01811     if (s->channels > MPA_MAX_CHANNELS)
01812         return AVERROR_INVALIDDATA;
01813 
01814     avctx->sample_rate = AV_RB32(extradata);
01815     extradata += 4;
01816 
01817     avctx->bit_rate = AV_RB32(extradata);
01818     extradata += 4;
01819 
01820     s->group_size = AV_RB32(extradata);
01821     extradata += 4;
01822 
01823     s->fft_size = AV_RB32(extradata);
01824     extradata += 4;
01825 
01826     s->checksum_size = AV_RB32(extradata);
01827     if (s->checksum_size >= 1U << 28) {
01828         av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
01829         return AVERROR_INVALIDDATA;
01830     }
01831 
01832     s->fft_order = av_log2(s->fft_size) + 1;
01833     s->fft_frame_size = 2 * s->fft_size; // complex has two floats
01834 
01835     // something like max decodable tones
01836     s->group_order = av_log2(s->group_size) + 1;
01837     s->frame_size = s->group_size / 16; // 16 iterations per super block
01838     if (s->frame_size > QDM2_MAX_FRAME_SIZE)
01839         return AVERROR_INVALIDDATA;
01840 
01841     s->sub_sampling = s->fft_order - 7;
01842     s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
01843 
01844     switch ((s->sub_sampling * 2 + s->channels - 1)) {
01845         case 0: tmp = 40; break;
01846         case 1: tmp = 48; break;
01847         case 2: tmp = 56; break;
01848         case 3: tmp = 72; break;
01849         case 4: tmp = 80; break;
01850         case 5: tmp = 100;break;
01851         default: tmp=s->sub_sampling; break;
01852     }
01853     tmp_val = 0;
01854     if ((tmp * 1000) < avctx->bit_rate)  tmp_val = 1;
01855     if ((tmp * 1440) < avctx->bit_rate)  tmp_val = 2;
01856     if ((tmp * 1760) < avctx->bit_rate)  tmp_val = 3;
01857     if ((tmp * 2240) < avctx->bit_rate)  tmp_val = 4;
01858     s->cm_table_select = tmp_val;
01859 
01860     if (s->sub_sampling == 0)
01861         tmp = 7999;
01862     else
01863         tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
01864     /*
01865     0: 7999 -> 0
01866     1: 20000 -> 2
01867     2: 28000 -> 2
01868     */
01869     if (tmp < 8000)
01870         s->coeff_per_sb_select = 0;
01871     else if (tmp <= 16000)
01872         s->coeff_per_sb_select = 1;
01873     else
01874         s->coeff_per_sb_select = 2;
01875 
01876     // Fail on unknown fft order
01877     if ((s->fft_order < 7) || (s->fft_order > 9)) {
01878         av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
01879         return -1;
01880     }
01881 
01882     ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
01883     ff_mpadsp_init(&s->mpadsp);
01884 
01885     qdm2_init(s);
01886 
01887     avctx->sample_fmt = AV_SAMPLE_FMT_S16;
01888 
01889 //    dump_context(s);
01890     return 0;
01891 }
01892 
01893 
01894 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
01895 {
01896     QDM2Context *s = avctx->priv_data;
01897 
01898     ff_rdft_end(&s->rdft_ctx);
01899 
01900     return 0;
01901 }
01902 
01903 
01904 static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
01905 {
01906     int ch, i;
01907     const int frame_size = (q->frame_size * q->channels);
01908 
01909     /* select input buffer */
01910     q->compressed_data = in;
01911     q->compressed_size = q->checksum_size;
01912 
01913 //  dump_context(q);
01914 
01915     /* copy old block, clear new block of output samples */
01916     memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
01917     memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
01918 
01919     /* decode block of QDM2 compressed data */
01920     if (q->sub_packet == 0) {
01921         q->has_errors = 0; // zero it for a new super block
01922         av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
01923         qdm2_decode_super_block(q);
01924     }
01925 
01926     /* parse subpackets */
01927     if (!q->has_errors) {
01928         if (q->sub_packet == 2)
01929             qdm2_decode_fft_packets(q);
01930 
01931         qdm2_fft_tone_synthesizer(q, q->sub_packet);
01932     }
01933 
01934     /* sound synthesis stage 1 (FFT) */
01935     for (ch = 0; ch < q->channels; ch++) {
01936         qdm2_calculate_fft(q, ch, q->sub_packet);
01937 
01938         if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
01939             SAMPLES_NEEDED_2("has errors, and C list is not empty")
01940             return -1;
01941         }
01942     }
01943 
01944     /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
01945     if (!q->has_errors && q->do_synth_filter)
01946         qdm2_synthesis_filter(q, q->sub_packet);
01947 
01948     q->sub_packet = (q->sub_packet + 1) % 16;
01949 
01950     /* clip and convert output float[] to 16bit signed samples */
01951     for (i = 0; i < frame_size; i++) {
01952         int value = (int)q->output_buffer[i];
01953 
01954         if (value > SOFTCLIP_THRESHOLD)
01955             value = (value >  HARDCLIP_THRESHOLD) ?  32767 :  softclip_table[ value - SOFTCLIP_THRESHOLD];
01956         else if (value < -SOFTCLIP_THRESHOLD)
01957             value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
01958 
01959         out[i] = value;
01960     }
01961 
01962     return 0;
01963 }
01964 
01965 
01966 static int qdm2_decode_frame(AVCodecContext *avctx,
01967             void *data, int *data_size,
01968             AVPacket *avpkt)
01969 {
01970     const uint8_t *buf = avpkt->data;
01971     int buf_size = avpkt->size;
01972     QDM2Context *s = avctx->priv_data;
01973     int16_t *out = data;
01974     int i, out_size;
01975 
01976     if(!buf)
01977         return 0;
01978     if(buf_size < s->checksum_size)
01979         return -1;
01980 
01981     out_size = 16 * s->channels * s->frame_size *
01982                av_get_bytes_per_sample(avctx->sample_fmt);
01983     if (*data_size < out_size) {
01984         av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
01985         return AVERROR(EINVAL);
01986     }
01987 
01988     av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
01989        buf_size, buf, s->checksum_size, data, *data_size);
01990 
01991     for (i = 0; i < 16; i++) {
01992         if (qdm2_decode(s, buf, out) < 0)
01993             return -1;
01994         out += s->channels * s->frame_size;
01995     }
01996 
01997     *data_size = out_size;
01998 
01999     return s->checksum_size;
02000 }
02001 
02002 AVCodec ff_qdm2_decoder =
02003 {
02004     .name = "qdm2",
02005     .type = AVMEDIA_TYPE_AUDIO,
02006     .id = CODEC_ID_QDM2,
02007     .priv_data_size = sizeof(QDM2Context),
02008     .init = qdm2_decode_init,
02009     .close = qdm2_decode_close,
02010     .decode = qdm2_decode_frame,
02011     .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
02012 };