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 
00080 typedef int8_t sb_int8_array[2][30][64];
00081 
00085 typedef struct {
00086     int type;            
00087     unsigned int size;   
00088     const uint8_t *data; 
00089 } QDM2SubPacket;
00090 
00094 typedef struct QDM2SubPNode {
00095     QDM2SubPacket *packet;      
00096     struct QDM2SubPNode *next; 
00097 } QDM2SubPNode;
00098 
00099 typedef struct {
00100     float re;
00101     float im;
00102 } QDM2Complex;
00103 
00104 typedef struct {
00105     float level;
00106     QDM2Complex *complex;
00107     const float *table;
00108     int   phase;
00109     int   phase_shift;
00110     int   duration;
00111     short time_index;
00112     short cutoff;
00113 } FFTTone;
00114 
00115 typedef struct {
00116     int16_t sub_packet;
00117     uint8_t channel;
00118     int16_t offset;
00119     int16_t exp;
00120     uint8_t phase;
00121 } FFTCoefficient;
00122 
00123 typedef struct {
00124     DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
00125 } QDM2FFT;
00126 
00130 typedef struct {
00132     int nb_channels;         
00133     int channels;            
00134     int group_size;          
00135     int fft_size;            
00136     int checksum_size;       
00137 
00139     int group_order;         
00140     int fft_order;           
00141     int fft_frame_size;      
00142     int frame_size;          
00143     int frequency_range;
00144     int sub_sampling;        
00145     int coeff_per_sb_select; 
00146     int cm_table_select;     
00147 
00149     QDM2SubPacket sub_packets[16];      
00150     QDM2SubPNode sub_packet_list_A[16]; 
00151     QDM2SubPNode sub_packet_list_B[16]; 
00152     int sub_packets_B;                  
00153     QDM2SubPNode sub_packet_list_C[16]; 
00154     QDM2SubPNode sub_packet_list_D[16]; 
00155 
00157     FFTTone fft_tones[1000];
00158     int fft_tone_start;
00159     int fft_tone_end;
00160     FFTCoefficient fft_coefs[1000];
00161     int fft_coefs_index;
00162     int fft_coefs_min_index[5];
00163     int fft_coefs_max_index[5];
00164     int fft_level_exp[6];
00165     RDFTContext rdft_ctx;
00166     QDM2FFT fft;
00167 
00169     const uint8_t *compressed_data;
00170     int compressed_size;
00171     float output_buffer[1024];
00172 
00174     MPADSPContext mpadsp;
00175     DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
00176     int synth_buf_offset[MPA_MAX_CHANNELS];
00177     DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
00178     DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
00179 
00181     float tone_level[MPA_MAX_CHANNELS][30][64];
00182     int8_t coding_method[MPA_MAX_CHANNELS][30][64];
00183     int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
00184     int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
00185     int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
00186     int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
00187     int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
00188     int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
00189     int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
00190 
00191     // Flags
00192     int has_errors;         
00193     int superblocktype_2_3; 
00194     int do_synth_filter;    
00195 
00196     int sub_packet;
00197     int noise_idx; 
00198 } QDM2Context;
00199 
00200 
00201 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
00202 
00203 static VLC vlc_tab_level;
00204 static VLC vlc_tab_diff;
00205 static VLC vlc_tab_run;
00206 static VLC fft_level_exp_alt_vlc;
00207 static VLC fft_level_exp_vlc;
00208 static VLC fft_stereo_exp_vlc;
00209 static VLC fft_stereo_phase_vlc;
00210 static VLC vlc_tab_tone_level_idx_hi1;
00211 static VLC vlc_tab_tone_level_idx_mid;
00212 static VLC vlc_tab_tone_level_idx_hi2;
00213 static VLC vlc_tab_type30;
00214 static VLC vlc_tab_type34;
00215 static VLC vlc_tab_fft_tone_offset[5];
00216 
00217 static const uint16_t qdm2_vlc_offs[] = {
00218     0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
00219 };
00220 
00221 static av_cold void qdm2_init_vlc(void)
00222 {
00223     static int vlcs_initialized = 0;
00224     static VLC_TYPE qdm2_table[3838][2];
00225 
00226     if (!vlcs_initialized) {
00227 
00228         vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]];
00229         vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
00230         init_vlc (&vlc_tab_level, 8, 24,
00231             vlc_tab_level_huffbits, 1, 1,
00232             vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00233 
00234         vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
00235         vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
00236         init_vlc (&vlc_tab_diff, 8, 37,
00237             vlc_tab_diff_huffbits, 1, 1,
00238             vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00239 
00240         vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]];
00241         vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
00242         init_vlc (&vlc_tab_run, 5, 6,
00243             vlc_tab_run_huffbits, 1, 1,
00244             vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00245 
00246         fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
00247         fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
00248         init_vlc (&fft_level_exp_alt_vlc, 8, 28,
00249             fft_level_exp_alt_huffbits, 1, 1,
00250             fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00251 
00252 
00253         fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
00254         fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
00255         init_vlc (&fft_level_exp_vlc, 8, 20,
00256             fft_level_exp_huffbits, 1, 1,
00257             fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00258 
00259         fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
00260         fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
00261         init_vlc (&fft_stereo_exp_vlc, 6, 7,
00262             fft_stereo_exp_huffbits, 1, 1,
00263             fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00264 
00265         fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
00266         fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
00267         init_vlc (&fft_stereo_phase_vlc, 6, 9,
00268             fft_stereo_phase_huffbits, 1, 1,
00269             fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00270 
00271         vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]];
00272         vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
00273         init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
00274             vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
00275             vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00276 
00277         vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]];
00278         vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
00279         init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
00280             vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
00281             vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00282 
00283         vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]];
00284         vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
00285         init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
00286             vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
00287             vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00288 
00289         vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
00290         vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
00291         init_vlc (&vlc_tab_type30, 6, 9,
00292             vlc_tab_type30_huffbits, 1, 1,
00293             vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00294 
00295         vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
00296         vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
00297         init_vlc (&vlc_tab_type34, 5, 10,
00298             vlc_tab_type34_huffbits, 1, 1,
00299             vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00300 
00301         vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
00302         vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
00303         init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
00304             vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
00305             vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00306 
00307         vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
00308         vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
00309         init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
00310             vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
00311             vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00312 
00313         vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
00314         vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
00315         init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
00316             vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
00317             vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00318 
00319         vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
00320         vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
00321         init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
00322             vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
00323             vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00324 
00325         vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
00326         vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
00327         init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
00328             vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
00329             vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00330 
00331         vlcs_initialized=1;
00332     }
00333 }
00334 
00335 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
00336 {
00337     int value;
00338 
00339     value = get_vlc2(gb, vlc->table, vlc->bits, depth);
00340 
00341     /* stage-2, 3 bits exponent escape sequence */
00342     if (value-- == 0)
00343         value = get_bits (gb, get_bits (gb, 3) + 1);
00344 
00345     /* stage-3, optional */
00346     if (flag) {
00347         int tmp = vlc_stage3_values[value];
00348 
00349         if ((value & ~3) > 0)
00350             tmp += get_bits (gb, (value >> 2));
00351         value = tmp;
00352     }
00353 
00354     return value;
00355 }
00356 
00357 
00358 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
00359 {
00360     int value = qdm2_get_vlc (gb, vlc, 0, depth);
00361 
00362     return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
00363 }
00364 
00365 
00375 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
00376     int i;
00377 
00378     for (i=0; i < length; i++)
00379         value -= data[i];
00380 
00381     return (uint16_t)(value & 0xffff);
00382 }
00383 
00384 
00391 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
00392 {
00393     sub_packet->type = get_bits (gb, 8);
00394 
00395     if (sub_packet->type == 0) {
00396         sub_packet->size = 0;
00397         sub_packet->data = NULL;
00398     } else {
00399         sub_packet->size = get_bits (gb, 8);
00400 
00401       if (sub_packet->type & 0x80) {
00402           sub_packet->size <<= 8;
00403           sub_packet->size  |= get_bits (gb, 8);
00404           sub_packet->type  &= 0x7f;
00405       }
00406 
00407       if (sub_packet->type == 0x7f)
00408           sub_packet->type |= (get_bits (gb, 8) << 8);
00409 
00410       sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
00411     }
00412 
00413     av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
00414         sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
00415 }
00416 
00417 
00425 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
00426 {
00427     while (list != NULL && list->packet != NULL) {
00428         if (list->packet->type == type)
00429             return list;
00430         list = list->next;
00431     }
00432     return NULL;
00433 }
00434 
00435 
00442 static void average_quantized_coeffs (QDM2Context *q)
00443 {
00444     int i, j, n, ch, sum;
00445 
00446     n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
00447 
00448     for (ch = 0; ch < q->nb_channels; ch++)
00449         for (i = 0; i < n; i++) {
00450             sum = 0;
00451 
00452             for (j = 0; j < 8; j++)
00453                 sum += q->quantized_coeffs[ch][i][j];
00454 
00455             sum /= 8;
00456             if (sum > 0)
00457                 sum--;
00458 
00459             for (j=0; j < 8; j++)
00460                 q->quantized_coeffs[ch][i][j] = sum;
00461         }
00462 }
00463 
00464 
00472 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
00473 {
00474     int ch, j;
00475 
00476     FIX_NOISE_IDX(q->noise_idx);
00477 
00478     if (!q->nb_channels)
00479         return;
00480 
00481     for (ch = 0; ch < q->nb_channels; ch++)
00482         for (j = 0; j < 64; j++) {
00483             q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
00484             q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
00485         }
00486 }
00487 
00488 
00497 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
00498 {
00499     int j,k;
00500     int ch;
00501     int run, case_val;
00502     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};
00503 
00504     for (ch = 0; ch < channels; ch++) {
00505         for (j = 0; j < 64; ) {
00506             if((coding_method[ch][sb][j] - 8) > 22) {
00507                 run = 1;
00508                 case_val = 8;
00509             } else {
00510                 switch (switchtable[coding_method[ch][sb][j]-8]) {
00511                     case 0: run = 10; case_val = 10; break;
00512                     case 1: run = 1; case_val = 16; break;
00513                     case 2: run = 5; case_val = 24; break;
00514                     case 3: run = 3; case_val = 30; break;
00515                     case 4: run = 1; case_val = 30; break;
00516                     case 5: run = 1; case_val = 8; break;
00517                     default: run = 1; case_val = 8; break;
00518                 }
00519             }
00520             for (k = 0; k < run; k++)
00521                 if (j + k < 128)
00522                     if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
00523                         if (k > 0) {
00524                            SAMPLES_NEEDED
00525                             //not debugged, almost never used
00526                             memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
00527                             memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
00528                         }
00529             j += run;
00530         }
00531     }
00532 }
00533 
00534 
00542 static void fill_tone_level_array (QDM2Context *q, int flag)
00543 {
00544     int i, sb, ch, sb_used;
00545     int tmp, tab;
00546 
00547     // This should never happen
00548     if (q->nb_channels <= 0)
00549         return;
00550 
00551     for (ch = 0; ch < q->nb_channels; ch++)
00552         for (sb = 0; sb < 30; sb++)
00553             for (i = 0; i < 8; i++) {
00554                 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
00555                     tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
00556                           q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
00557                 else
00558                     tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
00559                 if(tmp < 0)
00560                     tmp += 0xff;
00561                 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
00562             }
00563 
00564     sb_used = QDM2_SB_USED(q->sub_sampling);
00565 
00566     if ((q->superblocktype_2_3 != 0) && !flag) {
00567         for (sb = 0; sb < sb_used; sb++)
00568             for (ch = 0; ch < q->nb_channels; ch++)
00569                 for (i = 0; i < 64; i++) {
00570                     q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
00571                     if (q->tone_level_idx[ch][sb][i] < 0)
00572                         q->tone_level[ch][sb][i] = 0;
00573                     else
00574                         q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
00575                 }
00576     } else {
00577         tab = q->superblocktype_2_3 ? 0 : 1;
00578         for (sb = 0; sb < sb_used; sb++) {
00579             if ((sb >= 4) && (sb <= 23)) {
00580                 for (ch = 0; ch < q->nb_channels; ch++)
00581                     for (i = 0; i < 64; i++) {
00582                         tmp = q->tone_level_idx_base[ch][sb][i / 8] -
00583                               q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
00584                               q->tone_level_idx_mid[ch][sb - 4][i / 8] -
00585                               q->tone_level_idx_hi2[ch][sb - 4];
00586                         q->tone_level_idx[ch][sb][i] = tmp & 0xff;
00587                         if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
00588                             q->tone_level[ch][sb][i] = 0;
00589                         else
00590                             q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
00591                 }
00592             } else {
00593                 if (sb > 4) {
00594                     for (ch = 0; ch < q->nb_channels; ch++)
00595                         for (i = 0; i < 64; i++) {
00596                             tmp = q->tone_level_idx_base[ch][sb][i / 8] -
00597                                   q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
00598                                   q->tone_level_idx_hi2[ch][sb - 4];
00599                             q->tone_level_idx[ch][sb][i] = tmp & 0xff;
00600                             if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
00601                                 q->tone_level[ch][sb][i] = 0;
00602                             else
00603                                 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
00604                     }
00605                 } else {
00606                     for (ch = 0; ch < q->nb_channels; ch++)
00607                         for (i = 0; i < 64; i++) {
00608                             tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
00609                             if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
00610                                 q->tone_level[ch][sb][i] = 0;
00611                             else
00612                                 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
00613                         }
00614                 }
00615             }
00616         }
00617     }
00618 
00619     return;
00620 }
00621 
00622 
00637 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
00638                 sb_int8_array coding_method, int nb_channels,
00639                 int c, int superblocktype_2_3, int cm_table_select)
00640 {
00641     int ch, sb, j;
00642     int tmp, acc, esp_40, comp;
00643     int add1, add2, add3, add4;
00644     int64_t multres;
00645 
00646     // This should never happen
00647     if (nb_channels <= 0)
00648         return;
00649 
00650     if (!superblocktype_2_3) {
00651         /* This case is untested, no samples available */
00652         SAMPLES_NEEDED
00653         for (ch = 0; ch < nb_channels; ch++)
00654             for (sb = 0; sb < 30; sb++) {
00655                 for (j = 1; j < 63; j++) {  // The loop only iterates to 63 so the code doesn't overflow the buffer
00656                     add1 = tone_level_idx[ch][sb][j] - 10;
00657                     if (add1 < 0)
00658                         add1 = 0;
00659                     add2 = add3 = add4 = 0;
00660                     if (sb > 1) {
00661                         add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
00662                         if (add2 < 0)
00663                             add2 = 0;
00664                     }
00665                     if (sb > 0) {
00666                         add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
00667                         if (add3 < 0)
00668                             add3 = 0;
00669                     }
00670                     if (sb < 29) {
00671                         add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
00672                         if (add4 < 0)
00673                             add4 = 0;
00674                     }
00675                     tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
00676                     if (tmp < 0)
00677                         tmp = 0;
00678                     tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
00679                 }
00680                 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
00681             }
00682             acc = 0;
00683             for (ch = 0; ch < nb_channels; ch++)
00684                 for (sb = 0; sb < 30; sb++)
00685                     for (j = 0; j < 64; j++)
00686                         acc += tone_level_idx_temp[ch][sb][j];
00687 
00688             multres = 0x66666667 * (acc * 10);
00689             esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
00690             for (ch = 0;  ch < nb_channels; ch++)
00691                 for (sb = 0; sb < 30; sb++)
00692                     for (j = 0; j < 64; j++) {
00693                         comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
00694                         if (comp < 0)
00695                             comp += 0xff;
00696                         comp /= 256; // signed shift
00697                         switch(sb) {
00698                             case 0:
00699                                 if (comp < 30)
00700                                     comp = 30;
00701                                 comp += 15;
00702                                 break;
00703                             case 1:
00704                                 if (comp < 24)
00705                                     comp = 24;
00706                                 comp += 10;
00707                                 break;
00708                             case 2:
00709                             case 3:
00710                             case 4:
00711                                 if (comp < 16)
00712                                     comp = 16;
00713                         }
00714                         if (comp <= 5)
00715                             tmp = 0;
00716                         else if (comp <= 10)
00717                             tmp = 10;
00718                         else if (comp <= 16)
00719                             tmp = 16;
00720                         else if (comp <= 24)
00721                             tmp = -1;
00722                         else
00723                             tmp = 0;
00724                         coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
00725                     }
00726             for (sb = 0; sb < 30; sb++)
00727                 fix_coding_method_array(sb, nb_channels, coding_method);
00728             for (ch = 0; ch < nb_channels; ch++)
00729                 for (sb = 0; sb < 30; sb++)
00730                     for (j = 0; j < 64; j++)
00731                         if (sb >= 10) {
00732                             if (coding_method[ch][sb][j] < 10)
00733                                 coding_method[ch][sb][j] = 10;
00734                         } else {
00735                             if (sb >= 2) {
00736                                 if (coding_method[ch][sb][j] < 16)
00737                                     coding_method[ch][sb][j] = 16;
00738                             } else {
00739                                 if (coding_method[ch][sb][j] < 30)
00740                                     coding_method[ch][sb][j] = 30;
00741                             }
00742                         }
00743     } else { // superblocktype_2_3 != 0
00744         for (ch = 0; ch < nb_channels; ch++)
00745             for (sb = 0; sb < 30; sb++)
00746                 for (j = 0; j < 64; j++)
00747                     coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
00748     }
00749 
00750     return;
00751 }
00752 
00753 
00765 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
00766 {
00767     int sb, j, k, n, ch, run, channels;
00768     int joined_stereo, zero_encoding, chs;
00769     int type34_first;
00770     float type34_div = 0;
00771     float type34_predictor;
00772     float samples[10], sign_bits[16];
00773 
00774     if (length == 0) {
00775         // If no data use noise
00776         for (sb=sb_min; sb < sb_max; sb++)
00777             build_sb_samples_from_noise (q, sb);
00778 
00779         return;
00780     }
00781 
00782     for (sb = sb_min; sb < sb_max; sb++) {
00783         FIX_NOISE_IDX(q->noise_idx);
00784 
00785         channels = q->nb_channels;
00786 
00787         if (q->nb_channels <= 1 || sb < 12)
00788             joined_stereo = 0;
00789         else if (sb >= 24)
00790             joined_stereo = 1;
00791         else
00792             joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
00793 
00794         if (joined_stereo) {
00795             if (BITS_LEFT(length,gb) >= 16)
00796                 for (j = 0; j < 16; j++)
00797                     sign_bits[j] = get_bits1 (gb);
00798 
00799             for (j = 0; j < 64; j++)
00800                 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
00801                     q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
00802 
00803             fix_coding_method_array(sb, q->nb_channels, q->coding_method);
00804             channels = 1;
00805         }
00806 
00807         for (ch = 0; ch < channels; ch++) {
00808             zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
00809             type34_predictor = 0.0;
00810             type34_first = 1;
00811 
00812             for (j = 0; j < 128; ) {
00813                 switch (q->coding_method[ch][sb][j / 2]) {
00814                     case 8:
00815                         if (BITS_LEFT(length,gb) >= 10) {
00816                             if (zero_encoding) {
00817                                 for (k = 0; k < 5; k++) {
00818                                     if ((j + 2 * k) >= 128)
00819                                         break;
00820                                     samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
00821                                 }
00822                             } else {
00823                                 n = get_bits(gb, 8);
00824                                 for (k = 0; k < 5; k++)
00825                                     samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
00826                             }
00827                             for (k = 0; k < 5; k++)
00828                                 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
00829                         } else {
00830                             for (k = 0; k < 10; k++)
00831                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
00832                         }
00833                         run = 10;
00834                         break;
00835 
00836                     case 10:
00837                         if (BITS_LEFT(length,gb) >= 1) {
00838                             float f = 0.81;
00839 
00840                             if (get_bits1(gb))
00841                                 f = -f;
00842                             f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
00843                             samples[0] = f;
00844                         } else {
00845                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00846                         }
00847                         run = 1;
00848                         break;
00849 
00850                     case 16:
00851                         if (BITS_LEFT(length,gb) >= 10) {
00852                             if (zero_encoding) {
00853                                 for (k = 0; k < 5; k++) {
00854                                     if ((j + k) >= 128)
00855                                         break;
00856                                     samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
00857                                 }
00858                             } else {
00859                                 n = get_bits (gb, 8);
00860                                 for (k = 0; k < 5; k++)
00861                                     samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
00862                             }
00863                         } else {
00864                             for (k = 0; k < 5; k++)
00865                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
00866                         }
00867                         run = 5;
00868                         break;
00869 
00870                     case 24:
00871                         if (BITS_LEFT(length,gb) >= 7) {
00872                             n = get_bits(gb, 7);
00873                             for (k = 0; k < 3; k++)
00874                                 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
00875                         } else {
00876                             for (k = 0; k < 3; k++)
00877                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
00878                         }
00879                         run = 3;
00880                         break;
00881 
00882                     case 30:
00883                         if (BITS_LEFT(length,gb) >= 4)
00884                             samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
00885                         else
00886                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00887 
00888                         run = 1;
00889                         break;
00890 
00891                     case 34:
00892                         if (BITS_LEFT(length,gb) >= 7) {
00893                             if (type34_first) {
00894                                 type34_div = (float)(1 << get_bits(gb, 2));
00895                                 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
00896                                 type34_predictor = samples[0];
00897                                 type34_first = 0;
00898                             } else {
00899                                 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
00900                                 type34_predictor = samples[0];
00901                             }
00902                         } else {
00903                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00904                         }
00905                         run = 1;
00906                         break;
00907 
00908                     default:
00909                         samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00910                         run = 1;
00911                         break;
00912                 }
00913 
00914                 if (joined_stereo) {
00915                     float tmp[10][MPA_MAX_CHANNELS];
00916 
00917                     for (k = 0; k < run; k++) {
00918                         tmp[k][0] = samples[k];
00919                         tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
00920                     }
00921                     for (chs = 0; chs < q->nb_channels; chs++)
00922                         for (k = 0; k < run; k++)
00923                             if ((j + k) < 128)
00924                                 q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs];
00925                 } else {
00926                     for (k = 0; k < run; k++)
00927                         if ((j + k) < 128)
00928                             q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
00929                 }
00930 
00931                 j += run;
00932             } // j loop
00933         } // channel loop
00934     } // subband loop
00935 }
00936 
00937 
00947 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
00948 {
00949     int i, k, run, level, diff;
00950 
00951     if (BITS_LEFT(length,gb) < 16)
00952         return;
00953     level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
00954 
00955     quantized_coeffs[0] = level;
00956 
00957     for (i = 0; i < 7; ) {
00958         if (BITS_LEFT(length,gb) < 16)
00959             break;
00960         run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
00961 
00962         if (BITS_LEFT(length,gb) < 16)
00963             break;
00964         diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
00965 
00966         for (k = 1; k <= run; k++)
00967             quantized_coeffs[i + k] = (level + ((k * diff) / run));
00968 
00969         level += diff;
00970         i += run;
00971     }
00972 }
00973 
00974 
00984 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
00985 {
00986     int sb, j, k, n, ch;
00987 
00988     for (ch = 0; ch < q->nb_channels; ch++) {
00989         init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
00990 
00991         if (BITS_LEFT(length,gb) < 16) {
00992             memset(q->quantized_coeffs[ch][0], 0, 8);
00993             break;
00994         }
00995     }
00996 
00997     n = q->sub_sampling + 1;
00998 
00999     for (sb = 0; sb < n; sb++)
01000         for (ch = 0; ch < q->nb_channels; ch++)
01001             for (j = 0; j < 8; j++) {
01002                 if (BITS_LEFT(length,gb) < 1)
01003                     break;
01004                 if (get_bits1(gb)) {
01005                     for (k=0; k < 8; k++) {
01006                         if (BITS_LEFT(length,gb) < 16)
01007                             break;
01008                         q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
01009                     }
01010                 } else {
01011                     for (k=0; k < 8; k++)
01012                         q->tone_level_idx_hi1[ch][sb][j][k] = 0;
01013                 }
01014             }
01015 
01016     n = QDM2_SB_USED(q->sub_sampling) - 4;
01017 
01018     for (sb = 0; sb < n; sb++)
01019         for (ch = 0; ch < q->nb_channels; ch++) {
01020             if (BITS_LEFT(length,gb) < 16)
01021                 break;
01022             q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
01023             if (sb > 19)
01024                 q->tone_level_idx_hi2[ch][sb] -= 16;
01025             else
01026                 for (j = 0; j < 8; j++)
01027                     q->tone_level_idx_mid[ch][sb][j] = -16;
01028         }
01029 
01030     n = QDM2_SB_USED(q->sub_sampling) - 5;
01031 
01032     for (sb = 0; sb < n; sb++)
01033         for (ch = 0; ch < q->nb_channels; ch++)
01034             for (j = 0; j < 8; j++) {
01035                 if (BITS_LEFT(length,gb) < 16)
01036                     break;
01037                 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
01038             }
01039 }
01040 
01047 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
01048 {
01049     GetBitContext gb;
01050     int i, j, k, n, ch, run, level, diff;
01051 
01052     init_get_bits(&gb, node->packet->data, node->packet->size*8);
01053 
01054     n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
01055 
01056     for (i = 1; i < n; i++)
01057         for (ch=0; ch < q->nb_channels; ch++) {
01058             level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
01059             q->quantized_coeffs[ch][i][0] = level;
01060 
01061             for (j = 0; j < (8 - 1); ) {
01062                 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
01063                 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
01064 
01065                 for (k = 1; k <= run; k++)
01066                     q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
01067 
01068                 level += diff;
01069                 j += run;
01070             }
01071         }
01072 
01073     for (ch = 0; ch < q->nb_channels; ch++)
01074         for (i = 0; i < 8; i++)
01075             q->quantized_coeffs[ch][0][i] = 0;
01076 }
01077 
01078 
01086 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
01087 {
01088     GetBitContext gb;
01089 
01090     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
01091 
01092     if (length != 0) {
01093         init_tone_level_dequantization(q, &gb, length);
01094         fill_tone_level_array(q, 1);
01095     } else {
01096         fill_tone_level_array(q, 0);
01097     }
01098 }
01099 
01100 
01108 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
01109 {
01110     GetBitContext gb;
01111 
01112     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
01113     if (length >= 32) {
01114         int c = get_bits (&gb, 13);
01115 
01116         if (c > 3)
01117             fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
01118                                       q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
01119     }
01120 
01121     synthfilt_build_sb_samples(q, &gb, length, 0, 8);
01122 }
01123 
01124 
01132 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
01133 {
01134     GetBitContext gb;
01135 
01136     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
01137     synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
01138 }
01139 
01140 /*
01141  * Process new subpackets for synthesis filter
01142  *
01143  * @param q       context
01144  * @param list    list with synthesis filter packets (list D)
01145  */
01146 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
01147 {
01148     QDM2SubPNode *nodes[4];
01149 
01150     nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
01151     if (nodes[0] != NULL)
01152         process_subpacket_9(q, nodes[0]);
01153 
01154     nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
01155     if (nodes[1] != NULL)
01156         process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
01157     else
01158         process_subpacket_10(q, NULL, 0);
01159 
01160     nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
01161     if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
01162         process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
01163     else
01164         process_subpacket_11(q, NULL, 0);
01165 
01166     nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
01167     if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
01168         process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
01169     else
01170         process_subpacket_12(q, NULL, 0);
01171 }
01172 
01173 
01174 /*
01175  * Decode superblock, fill packet lists.
01176  *
01177  * @param q    context
01178  */
01179 static void qdm2_decode_super_block (QDM2Context *q)
01180 {
01181     GetBitContext gb;
01182     QDM2SubPacket header, *packet;
01183     int i, packet_bytes, sub_packet_size, sub_packets_D;
01184     unsigned int next_index = 0;
01185 
01186     memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
01187     memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
01188     memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
01189 
01190     q->sub_packets_B = 0;
01191     sub_packets_D = 0;
01192 
01193     average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
01194 
01195     init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
01196     qdm2_decode_sub_packet_header(&gb, &header);
01197 
01198     if (header.type < 2 || header.type >= 8) {
01199         q->has_errors = 1;
01200         av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
01201         return;
01202     }
01203 
01204     q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
01205     packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
01206 
01207     init_get_bits(&gb, header.data, header.size*8);
01208 
01209     if (header.type == 2 || header.type == 4 || header.type == 5) {
01210         int csum  = 257 * get_bits(&gb, 8);
01211             csum +=   2 * get_bits(&gb, 8);
01212 
01213         csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
01214 
01215         if (csum != 0) {
01216             q->has_errors = 1;
01217             av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
01218             return;
01219         }
01220     }
01221 
01222     q->sub_packet_list_B[0].packet = NULL;
01223     q->sub_packet_list_D[0].packet = NULL;
01224 
01225     for (i = 0; i < 6; i++)
01226         if (--q->fft_level_exp[i] < 0)
01227             q->fft_level_exp[i] = 0;
01228 
01229     for (i = 0; packet_bytes > 0; i++) {
01230         int j;
01231 
01232         q->sub_packet_list_A[i].next = NULL;
01233 
01234         if (i > 0) {
01235             q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
01236 
01237             /* seek to next block */
01238             init_get_bits(&gb, header.data, header.size*8);
01239             skip_bits(&gb, next_index*8);
01240 
01241             if (next_index >= header.size)
01242                 break;
01243         }
01244 
01245         /* decode subpacket */
01246         packet = &q->sub_packets[i];
01247         qdm2_decode_sub_packet_header(&gb, packet);
01248         next_index = packet->size + get_bits_count(&gb) / 8;
01249         sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
01250 
01251         if (packet->type == 0)
01252             break;
01253 
01254         if (sub_packet_size > packet_bytes) {
01255             if (packet->type != 10 && packet->type != 11 && packet->type != 12)
01256                 break;
01257             packet->size += packet_bytes - sub_packet_size;
01258         }
01259 
01260         packet_bytes -= sub_packet_size;
01261 
01262         /* add subpacket to 'all subpackets' list */
01263         q->sub_packet_list_A[i].packet = packet;
01264 
01265         /* add subpacket to related list */
01266         if (packet->type == 8) {
01267             SAMPLES_NEEDED_2("packet type 8");
01268             return;
01269         } else if (packet->type >= 9 && packet->type <= 12) {
01270             /* packets for MPEG Audio like Synthesis Filter */
01271             QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
01272         } else if (packet->type == 13) {
01273             for (j = 0; j < 6; j++)
01274                 q->fft_level_exp[j] = get_bits(&gb, 6);
01275         } else if (packet->type == 14) {
01276             for (j = 0; j < 6; j++)
01277                 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
01278         } else if (packet->type == 15) {
01279             SAMPLES_NEEDED_2("packet type 15")
01280             return;
01281         } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
01282             /* packets for FFT */
01283             QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
01284         }
01285     } // Packet bytes loop
01286 
01287 /* **************************************************************** */
01288     if (q->sub_packet_list_D[0].packet != NULL) {
01289         process_synthesis_subpackets(q, q->sub_packet_list_D);
01290         q->do_synth_filter = 1;
01291     } else if (q->do_synth_filter) {
01292         process_subpacket_10(q, NULL, 0);
01293         process_subpacket_11(q, NULL, 0);
01294         process_subpacket_12(q, NULL, 0);
01295     }
01296 /* **************************************************************** */
01297 }
01298 
01299 
01300 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
01301                        int offset, int duration, int channel,
01302                        int exp, int phase)
01303 {
01304     if (q->fft_coefs_min_index[duration] < 0)
01305         q->fft_coefs_min_index[duration] = q->fft_coefs_index;
01306 
01307     q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
01308     q->fft_coefs[q->fft_coefs_index].channel = channel;
01309     q->fft_coefs[q->fft_coefs_index].offset = offset;
01310     q->fft_coefs[q->fft_coefs_index].exp = exp;
01311     q->fft_coefs[q->fft_coefs_index].phase = phase;
01312     q->fft_coefs_index++;
01313 }
01314 
01315 
01316 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
01317 {
01318     int channel, stereo, phase, exp;
01319     int local_int_4,  local_int_8,  stereo_phase,  local_int_10;
01320     int local_int_14, stereo_exp, local_int_20, local_int_28;
01321     int n, offset;
01322 
01323     local_int_4 = 0;
01324     local_int_28 = 0;
01325     local_int_20 = 2;
01326     local_int_8 = (4 - duration);
01327     local_int_10 = 1 << (q->group_order - duration - 1);
01328     offset = 1;
01329 
01330     while (1) {
01331         if (q->superblocktype_2_3) {
01332             while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
01333                 offset = 1;
01334                 if (n == 0) {
01335                     local_int_4 += local_int_10;
01336                     local_int_28 += (1 << local_int_8);
01337                 } else {
01338                     local_int_4 += 8*local_int_10;
01339                     local_int_28 += (8 << local_int_8);
01340                 }
01341             }
01342             offset += (n - 2);
01343         } else {
01344             offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
01345             while (offset >= (local_int_10 - 1)) {
01346                 offset += (1 - (local_int_10 - 1));
01347                 local_int_4  += local_int_10;
01348                 local_int_28 += (1 << local_int_8);
01349             }
01350         }
01351 
01352         if (local_int_4 >= q->group_size)
01353             return;
01354 
01355         local_int_14 = (offset >> local_int_8);
01356 
01357         if (q->nb_channels > 1) {
01358             channel = get_bits1(gb);
01359             stereo = get_bits1(gb);
01360         } else {
01361             channel = 0;
01362             stereo = 0;
01363         }
01364 
01365         exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
01366         exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
01367         exp = (exp < 0) ? 0 : exp;
01368 
01369         phase = get_bits(gb, 3);
01370         stereo_exp = 0;
01371         stereo_phase = 0;
01372 
01373         if (stereo) {
01374             stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
01375             stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
01376             if (stereo_phase < 0)
01377                 stereo_phase += 8;
01378         }
01379 
01380         if (q->frequency_range > (local_int_14 + 1)) {
01381             int sub_packet = (local_int_20 + local_int_28);
01382 
01383             qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
01384             if (stereo)
01385                 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
01386         }
01387 
01388         offset++;
01389     }
01390 }
01391 
01392 
01393 static void qdm2_decode_fft_packets (QDM2Context *q)
01394 {
01395     int i, j, min, max, value, type, unknown_flag;
01396     GetBitContext gb;
01397 
01398     if (q->sub_packet_list_B[0].packet == NULL)
01399         return;
01400 
01401     /* reset minimum indexes for FFT coefficients */
01402     q->fft_coefs_index = 0;
01403     for (i=0; i < 5; i++)
01404         q->fft_coefs_min_index[i] = -1;
01405 
01406     /* process subpackets ordered by type, largest type first */
01407     for (i = 0, max = 256; i < q->sub_packets_B; i++) {
01408         QDM2SubPacket *packet= NULL;
01409 
01410         /* find subpacket with largest type less than max */
01411         for (j = 0, min = 0; j < q->sub_packets_B; j++) {
01412             value = q->sub_packet_list_B[j].packet->type;
01413             if (value > min && value < max) {
01414                 min = value;
01415                 packet = q->sub_packet_list_B[j].packet;
01416             }
01417         }
01418 
01419         max = min;
01420 
01421         /* check for errors (?) */
01422         if (!packet)
01423             return;
01424 
01425         if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
01426             return;
01427 
01428         /* decode FFT tones */
01429         init_get_bits (&gb, packet->data, packet->size*8);
01430 
01431         if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
01432             unknown_flag = 1;
01433         else
01434             unknown_flag = 0;
01435 
01436         type = packet->type;
01437 
01438         if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
01439             int duration = q->sub_sampling + 5 - (type & 15);
01440 
01441             if (duration >= 0 && duration < 4)
01442                 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
01443         } else if (type == 31) {
01444             for (j=0; j < 4; j++)
01445                 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
01446         } else if (type == 46) {
01447             for (j=0; j < 6; j++)
01448                 q->fft_level_exp[j] = get_bits(&gb, 6);
01449             for (j=0; j < 4; j++)
01450             qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
01451         }
01452     } // Loop on B packets
01453 
01454     /* calculate maximum indexes for FFT coefficients */
01455     for (i = 0, j = -1; i < 5; i++)
01456         if (q->fft_coefs_min_index[i] >= 0) {
01457             if (j >= 0)
01458                 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
01459             j = i;
01460         }
01461     if (j >= 0)
01462         q->fft_coefs_max_index[j] = q->fft_coefs_index;
01463 }
01464 
01465 
01466 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
01467 {
01468    float level, f[6];
01469    int i;
01470    QDM2Complex c;
01471    const double iscale = 2.0*M_PI / 512.0;
01472 
01473     tone->phase += tone->phase_shift;
01474 
01475     /* calculate current level (maximum amplitude) of tone */
01476     level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
01477     c.im = level * sin(tone->phase*iscale);
01478     c.re = level * cos(tone->phase*iscale);
01479 
01480     /* generate FFT coefficients for tone */
01481     if (tone->duration >= 3 || tone->cutoff >= 3) {
01482         tone->complex[0].im += c.im;
01483         tone->complex[0].re += c.re;
01484         tone->complex[1].im -= c.im;
01485         tone->complex[1].re -= c.re;
01486     } else {
01487         f[1] = -tone->table[4];
01488         f[0] =  tone->table[3] - tone->table[0];
01489         f[2] =  1.0 - tone->table[2] - tone->table[3];
01490         f[3] =  tone->table[1] + tone->table[4] - 1.0;
01491         f[4] =  tone->table[0] - tone->table[1];
01492         f[5] =  tone->table[2];
01493         for (i = 0; i < 2; i++) {
01494             tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
01495             tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
01496         }
01497         for (i = 0; i < 4; i++) {
01498             tone->complex[i].re += c.re * f[i+2];
01499             tone->complex[i].im += c.im * f[i+2];
01500         }
01501     }
01502 
01503     /* copy the tone if it has not yet died out */
01504     if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
01505       memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
01506       q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
01507     }
01508 }
01509 
01510 
01511 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
01512 {
01513     int i, j, ch;
01514     const double iscale = 0.25 * M_PI;
01515 
01516     for (ch = 0; ch < q->channels; ch++) {
01517         memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
01518     }
01519 
01520 
01521     /* apply FFT tones with duration 4 (1 FFT period) */
01522     if (q->fft_coefs_min_index[4] >= 0)
01523         for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
01524             float level;
01525             QDM2Complex c;
01526 
01527             if (q->fft_coefs[i].sub_packet != sub_packet)
01528                 break;
01529 
01530             ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
01531             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];
01532 
01533             c.re = level * cos(q->fft_coefs[i].phase * iscale);
01534             c.im = level * sin(q->fft_coefs[i].phase * iscale);
01535             q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
01536             q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
01537             q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
01538             q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
01539         }
01540 
01541     /* generate existing FFT tones */
01542     for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
01543         qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
01544         q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
01545     }
01546 
01547     /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
01548     for (i = 0; i < 4; i++)
01549         if (q->fft_coefs_min_index[i] >= 0) {
01550             for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
01551                 int offset, four_i;
01552                 FFTTone tone;
01553 
01554                 if (q->fft_coefs[j].sub_packet != sub_packet)
01555                     break;
01556 
01557                 four_i = (4 - i);
01558                 offset = q->fft_coefs[j].offset >> four_i;
01559                 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
01560 
01561                 if (offset < q->frequency_range) {
01562                     if (offset < 2)
01563                         tone.cutoff = offset;
01564                     else
01565                         tone.cutoff = (offset >= 60) ? 3 : 2;
01566 
01567                     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];
01568                     tone.complex = &q->fft.complex[ch][offset];
01569                     tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
01570                     tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
01571                     tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
01572                     tone.duration = i;
01573                     tone.time_index = 0;
01574 
01575                     qdm2_fft_generate_tone(q, &tone);
01576                 }
01577             }
01578             q->fft_coefs_min_index[i] = j;
01579         }
01580 }
01581 
01582 
01583 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
01584 {
01585     const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
01586     int i;
01587     q->fft.complex[channel][0].re *= 2.0f;
01588     q->fft.complex[channel][0].im = 0.0f;
01589     q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
01590     /* add samples to output buffer */
01591     for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
01592         q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
01593 }
01594 
01595 
01600 static void qdm2_synthesis_filter (QDM2Context *q, int index)
01601 {
01602     int i, k, ch, sb_used, sub_sampling, dither_state = 0;
01603 
01604     /* copy sb_samples */
01605     sb_used = QDM2_SB_USED(q->sub_sampling);
01606 
01607     for (ch = 0; ch < q->channels; ch++)
01608         for (i = 0; i < 8; i++)
01609             for (k=sb_used; k < SBLIMIT; k++)
01610                 q->sb_samples[ch][(8 * index) + i][k] = 0;
01611 
01612     for (ch = 0; ch < q->nb_channels; ch++) {
01613         float *samples_ptr = q->samples + ch;
01614 
01615         for (i = 0; i < 8; i++) {
01616             ff_mpa_synth_filter_float(&q->mpadsp,
01617                 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
01618                 ff_mpa_synth_window_float, &dither_state,
01619                 samples_ptr, q->nb_channels,
01620                 q->sb_samples[ch][(8 * index) + i]);
01621             samples_ptr += 32 * q->nb_channels;
01622         }
01623     }
01624 
01625     /* add samples to output buffer */
01626     sub_sampling = (4 >> q->sub_sampling);
01627 
01628     for (ch = 0; ch < q->channels; ch++)
01629         for (i = 0; i < q->frame_size; i++)
01630             q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
01631 }
01632 
01633 
01639 static av_cold void qdm2_init(QDM2Context *q) {
01640     static int initialized = 0;
01641 
01642     if (initialized != 0)
01643         return;
01644     initialized = 1;
01645 
01646     qdm2_init_vlc();
01647     ff_mpa_synth_init_float(ff_mpa_synth_window_float);
01648     softclip_table_init();
01649     rnd_table_init();
01650     init_noise_samples();
01651 
01652     av_log(NULL, AV_LOG_DEBUG, "init done\n");
01653 }
01654 
01655 
01656 #if 0
01657 static void dump_context(QDM2Context *q)
01658 {
01659     int i;
01660 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
01661     PRINT("compressed_data",q->compressed_data);
01662     PRINT("compressed_size",q->compressed_size);
01663     PRINT("frame_size",q->frame_size);
01664     PRINT("checksum_size",q->checksum_size);
01665     PRINT("channels",q->channels);
01666     PRINT("nb_channels",q->nb_channels);
01667     PRINT("fft_frame_size",q->fft_frame_size);
01668     PRINT("fft_size",q->fft_size);
01669     PRINT("sub_sampling",q->sub_sampling);
01670     PRINT("fft_order",q->fft_order);
01671     PRINT("group_order",q->group_order);
01672     PRINT("group_size",q->group_size);
01673     PRINT("sub_packet",q->sub_packet);
01674     PRINT("frequency_range",q->frequency_range);
01675     PRINT("has_errors",q->has_errors);
01676     PRINT("fft_tone_end",q->fft_tone_end);
01677     PRINT("fft_tone_start",q->fft_tone_start);
01678     PRINT("fft_coefs_index",q->fft_coefs_index);
01679     PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
01680     PRINT("cm_table_select",q->cm_table_select);
01681     PRINT("noise_idx",q->noise_idx);
01682 
01683     for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
01684     {
01685     FFTTone *t = &q->fft_tones[i];
01686 
01687     av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
01688     av_log(NULL,AV_LOG_DEBUG,"  level = %f\n", t->level);
01689 //  PRINT(" level", t->level);
01690     PRINT(" phase", t->phase);
01691     PRINT(" phase_shift", t->phase_shift);
01692     PRINT(" duration", t->duration);
01693     PRINT(" samples_im", t->samples_im);
01694     PRINT(" samples_re", t->samples_re);
01695     PRINT(" table", t->table);
01696     }
01697 
01698 }
01699 #endif
01700 
01701 
01705 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
01706 {
01707     QDM2Context *s = avctx->priv_data;
01708     uint8_t *extradata;
01709     int extradata_size;
01710     int tmp_val, tmp, size;
01711 
01712     /* extradata parsing
01713 
01714     Structure:
01715     wave {
01716         frma (QDM2)
01717         QDCA
01718         QDCP
01719     }
01720 
01721     32  size (including this field)
01722     32  tag (=frma)
01723     32  type (=QDM2 or QDMC)
01724 
01725     32  size (including this field, in bytes)
01726     32  tag (=QDCA) // maybe mandatory parameters
01727     32  unknown (=1)
01728     32  channels (=2)
01729     32  samplerate (=44100)
01730     32  bitrate (=96000)
01731     32  block size (=4096)
01732     32  frame size (=256) (for one channel)
01733     32  packet size (=1300)
01734 
01735     32  size (including this field, in bytes)
01736     32  tag (=QDCP) // maybe some tuneable parameters
01737     32  float1 (=1.0)
01738     32  zero ?
01739     32  float2 (=1.0)
01740     32  float3 (=1.0)
01741     32  unknown (27)
01742     32  unknown (8)
01743     32  zero ?
01744     */
01745 
01746     if (!avctx->extradata || (avctx->extradata_size < 48)) {
01747         av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
01748         return -1;
01749     }
01750 
01751     extradata = avctx->extradata;
01752     extradata_size = avctx->extradata_size;
01753 
01754     while (extradata_size > 7) {
01755         if (!memcmp(extradata, "frmaQDM", 7))
01756             break;
01757         extradata++;
01758         extradata_size--;
01759     }
01760 
01761     if (extradata_size < 12) {
01762         av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
01763                extradata_size);
01764         return -1;
01765     }
01766 
01767     if (memcmp(extradata, "frmaQDM", 7)) {
01768         av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
01769         return -1;
01770     }
01771 
01772     if (extradata[7] == 'C') {
01773 //        s->is_qdmc = 1;
01774         av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
01775         return -1;
01776     }
01777 
01778     extradata += 8;
01779     extradata_size -= 8;
01780 
01781     size = AV_RB32(extradata);
01782 
01783     if(size > extradata_size){
01784         av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
01785                extradata_size, size);
01786         return -1;
01787     }
01788 
01789     extradata += 4;
01790     av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
01791     if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
01792         av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
01793         return -1;
01794     }
01795 
01796     extradata += 8;
01797 
01798     avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
01799     extradata += 4;
01800 
01801     avctx->sample_rate = AV_RB32(extradata);
01802     extradata += 4;
01803 
01804     avctx->bit_rate = AV_RB32(extradata);
01805     extradata += 4;
01806 
01807     s->group_size = AV_RB32(extradata);
01808     extradata += 4;
01809 
01810     s->fft_size = AV_RB32(extradata);
01811     extradata += 4;
01812 
01813     s->checksum_size = AV_RB32(extradata);
01814 
01815     s->fft_order = av_log2(s->fft_size) + 1;
01816     s->fft_frame_size = 2 * s->fft_size; // complex has two floats
01817 
01818     // something like max decodable tones
01819     s->group_order = av_log2(s->group_size) + 1;
01820     s->frame_size = s->group_size / 16; // 16 iterations per super block
01821 
01822     s->sub_sampling = s->fft_order - 7;
01823     s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
01824 
01825     switch ((s->sub_sampling * 2 + s->channels - 1)) {
01826         case 0: tmp = 40; break;
01827         case 1: tmp = 48; break;
01828         case 2: tmp = 56; break;
01829         case 3: tmp = 72; break;
01830         case 4: tmp = 80; break;
01831         case 5: tmp = 100;break;
01832         default: tmp=s->sub_sampling; break;
01833     }
01834     tmp_val = 0;
01835     if ((tmp * 1000) < avctx->bit_rate)  tmp_val = 1;
01836     if ((tmp * 1440) < avctx->bit_rate)  tmp_val = 2;
01837     if ((tmp * 1760) < avctx->bit_rate)  tmp_val = 3;
01838     if ((tmp * 2240) < avctx->bit_rate)  tmp_val = 4;
01839     s->cm_table_select = tmp_val;
01840 
01841     if (s->sub_sampling == 0)
01842         tmp = 7999;
01843     else
01844         tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
01845     /*
01846     0: 7999 -> 0
01847     1: 20000 -> 2
01848     2: 28000 -> 2
01849     */
01850     if (tmp < 8000)
01851         s->coeff_per_sb_select = 0;
01852     else if (tmp <= 16000)
01853         s->coeff_per_sb_select = 1;
01854     else
01855         s->coeff_per_sb_select = 2;
01856 
01857     // Fail on unknown fft order
01858     if ((s->fft_order < 7) || (s->fft_order > 9)) {
01859         av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
01860         return -1;
01861     }
01862 
01863     ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
01864     ff_mpadsp_init(&s->mpadsp);
01865 
01866     qdm2_init(s);
01867 
01868     avctx->sample_fmt = AV_SAMPLE_FMT_S16;
01869 
01870 //    dump_context(s);
01871     return 0;
01872 }
01873 
01874 
01875 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
01876 {
01877     QDM2Context *s = avctx->priv_data;
01878 
01879     ff_rdft_end(&s->rdft_ctx);
01880 
01881     return 0;
01882 }
01883 
01884 
01885 static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
01886 {
01887     int ch, i;
01888     const int frame_size = (q->frame_size * q->channels);
01889 
01890     /* select input buffer */
01891     q->compressed_data = in;
01892     q->compressed_size = q->checksum_size;
01893 
01894 //  dump_context(q);
01895 
01896     /* copy old block, clear new block of output samples */
01897     memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
01898     memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
01899 
01900     /* decode block of QDM2 compressed data */
01901     if (q->sub_packet == 0) {
01902         q->has_errors = 0; // zero it for a new super block
01903         av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
01904         qdm2_decode_super_block(q);
01905     }
01906 
01907     /* parse subpackets */
01908     if (!q->has_errors) {
01909         if (q->sub_packet == 2)
01910             qdm2_decode_fft_packets(q);
01911 
01912         qdm2_fft_tone_synthesizer(q, q->sub_packet);
01913     }
01914 
01915     /* sound synthesis stage 1 (FFT) */
01916     for (ch = 0; ch < q->channels; ch++) {
01917         qdm2_calculate_fft(q, ch, q->sub_packet);
01918 
01919         if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
01920             SAMPLES_NEEDED_2("has errors, and C list is not empty")
01921             return -1;
01922         }
01923     }
01924 
01925     /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
01926     if (!q->has_errors && q->do_synth_filter)
01927         qdm2_synthesis_filter(q, q->sub_packet);
01928 
01929     q->sub_packet = (q->sub_packet + 1) % 16;
01930 
01931     /* clip and convert output float[] to 16bit signed samples */
01932     for (i = 0; i < frame_size; i++) {
01933         int value = (int)q->output_buffer[i];
01934 
01935         if (value > SOFTCLIP_THRESHOLD)
01936             value = (value >  HARDCLIP_THRESHOLD) ?  32767 :  softclip_table[ value - SOFTCLIP_THRESHOLD];
01937         else if (value < -SOFTCLIP_THRESHOLD)
01938             value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
01939 
01940         out[i] = value;
01941     }
01942 
01943     return 0;
01944 }
01945 
01946 
01947 static int qdm2_decode_frame(AVCodecContext *avctx,
01948             void *data, int *data_size,
01949             AVPacket *avpkt)
01950 {
01951     const uint8_t *buf = avpkt->data;
01952     int buf_size = avpkt->size;
01953     QDM2Context *s = avctx->priv_data;
01954     int16_t *out = data;
01955     int i;
01956 
01957     if(!buf)
01958         return 0;
01959     if(buf_size < s->checksum_size)
01960         return -1;
01961 
01962     av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
01963        buf_size, buf, s->checksum_size, data, *data_size);
01964 
01965     for (i = 0; i < 16; i++) {
01966         if (qdm2_decode(s, buf, out) < 0)
01967             return -1;
01968         out += s->channels * s->frame_size;
01969     }
01970 
01971     *data_size = (uint8_t*)out - (uint8_t*)data;
01972 
01973     return s->checksum_size;
01974 }
01975 
01976 AVCodec ff_qdm2_decoder =
01977 {
01978     .name = "qdm2",
01979     .type = AVMEDIA_TYPE_AUDIO,
01980     .id = CODEC_ID_QDM2,
01981     .priv_data_size = sizeof(QDM2Context),
01982     .init = qdm2_decode_init,
01983     .close = qdm2_decode_close,
01984     .decode = qdm2_decode_frame,
01985     .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
01986 };