libavcodec/qdm2.c
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00001 /*
00002  * QDM2 compatible decoder
00003  * Copyright (c) 2003 Ewald Snel
00004  * Copyright (c) 2005 Benjamin Larsson
00005  * Copyright (c) 2005 Alex Beregszaszi
00006  * Copyright (c) 2005 Roberto Togni
00007  *
00008  * This file is part of 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 
00034 #include <math.h>
00035 #include <stddef.h>
00036 #include <stdio.h>
00037 
00038 #define BITSTREAM_READER_LE
00039 #include "avcodec.h"
00040 #include "get_bits.h"
00041 #include "dsputil.h"
00042 #include "rdft.h"
00043 #include "mpegaudiodsp.h"
00044 #include "mpegaudio.h"
00045 
00046 #include "qdm2data.h"
00047 #include "qdm2_tablegen.h"
00048 
00049 #undef NDEBUG
00050 #include <assert.h>
00051 
00052 
00053 #define QDM2_LIST_ADD(list, size, packet) \
00054 do { \
00055       if (size > 0) { \
00056     list[size - 1].next = &list[size]; \
00057       } \
00058       list[size].packet = packet; \
00059       list[size].next = NULL; \
00060       size++; \
00061 } while(0)
00062 
00063 // Result is 8, 16 or 30
00064 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
00065 
00066 #define FIX_NOISE_IDX(noise_idx) \
00067   if ((noise_idx) >= 3840) \
00068     (noise_idx) -= 3840; \
00069 
00070 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
00071 
00072 #define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
00073 
00074 #define SAMPLES_NEEDED \
00075      av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
00076 
00077 #define SAMPLES_NEEDED_2(why) \
00078      av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
00079 
00080 #define QDM2_MAX_FRAME_SIZE 512
00081 
00082 typedef int8_t sb_int8_array[2][30][64];
00083 
00087 typedef struct {
00088     int type;            
00089     unsigned int size;   
00090     const uint8_t *data; 
00091 } QDM2SubPacket;
00092 
00096 typedef struct QDM2SubPNode {
00097     QDM2SubPacket *packet;      
00098     struct QDM2SubPNode *next; 
00099 } QDM2SubPNode;
00100 
00101 typedef struct {
00102     float re;
00103     float im;
00104 } QDM2Complex;
00105 
00106 typedef struct {
00107     float level;
00108     QDM2Complex *complex;
00109     const float *table;
00110     int   phase;
00111     int   phase_shift;
00112     int   duration;
00113     short time_index;
00114     short cutoff;
00115 } FFTTone;
00116 
00117 typedef struct {
00118     int16_t sub_packet;
00119     uint8_t channel;
00120     int16_t offset;
00121     int16_t exp;
00122     uint8_t phase;
00123 } FFTCoefficient;
00124 
00125 typedef struct {
00126     DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
00127 } QDM2FFT;
00128 
00132 typedef struct {
00133     AVFrame frame;
00134 
00136     int nb_channels;         
00137     int channels;            
00138     int group_size;          
00139     int fft_size;            
00140     int checksum_size;       
00141 
00143     int group_order;         
00144     int fft_order;           
00145     int fft_frame_size;      
00146     int frame_size;          
00147     int frequency_range;
00148     int sub_sampling;        
00149     int coeff_per_sb_select; 
00150     int cm_table_select;     
00151 
00153     QDM2SubPacket sub_packets[16];      
00154     QDM2SubPNode sub_packet_list_A[16]; 
00155     QDM2SubPNode sub_packet_list_B[16]; 
00156     int sub_packets_B;                  
00157     QDM2SubPNode sub_packet_list_C[16]; 
00158     QDM2SubPNode sub_packet_list_D[16]; 
00159 
00161     FFTTone fft_tones[1000];
00162     int fft_tone_start;
00163     int fft_tone_end;
00164     FFTCoefficient fft_coefs[1000];
00165     int fft_coefs_index;
00166     int fft_coefs_min_index[5];
00167     int fft_coefs_max_index[5];
00168     int fft_level_exp[6];
00169     RDFTContext rdft_ctx;
00170     QDM2FFT fft;
00171 
00173     const uint8_t *compressed_data;
00174     int compressed_size;
00175     float output_buffer[QDM2_MAX_FRAME_SIZE * 2];
00176 
00178     MPADSPContext mpadsp;
00179     DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
00180     int synth_buf_offset[MPA_MAX_CHANNELS];
00181     DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
00182     DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
00183 
00185     float tone_level[MPA_MAX_CHANNELS][30][64];
00186     int8_t coding_method[MPA_MAX_CHANNELS][30][64];
00187     int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
00188     int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
00189     int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
00190     int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
00191     int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
00192     int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
00193     int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
00194 
00195     // Flags
00196     int has_errors;         
00197     int superblocktype_2_3; 
00198     int do_synth_filter;    
00199 
00200     int sub_packet;
00201     int noise_idx; 
00202 } QDM2Context;
00203 
00204 
00205 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
00206 
00207 static VLC vlc_tab_level;
00208 static VLC vlc_tab_diff;
00209 static VLC vlc_tab_run;
00210 static VLC fft_level_exp_alt_vlc;
00211 static VLC fft_level_exp_vlc;
00212 static VLC fft_stereo_exp_vlc;
00213 static VLC fft_stereo_phase_vlc;
00214 static VLC vlc_tab_tone_level_idx_hi1;
00215 static VLC vlc_tab_tone_level_idx_mid;
00216 static VLC vlc_tab_tone_level_idx_hi2;
00217 static VLC vlc_tab_type30;
00218 static VLC vlc_tab_type34;
00219 static VLC vlc_tab_fft_tone_offset[5];
00220 
00221 static const uint16_t qdm2_vlc_offs[] = {
00222     0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
00223 };
00224 
00225 static av_cold void qdm2_init_vlc(void)
00226 {
00227     static int vlcs_initialized = 0;
00228     static VLC_TYPE qdm2_table[3838][2];
00229 
00230     if (!vlcs_initialized) {
00231 
00232         vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]];
00233         vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
00234         init_vlc (&vlc_tab_level, 8, 24,
00235             vlc_tab_level_huffbits, 1, 1,
00236             vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00237 
00238         vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
00239         vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
00240         init_vlc (&vlc_tab_diff, 8, 37,
00241             vlc_tab_diff_huffbits, 1, 1,
00242             vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00243 
00244         vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]];
00245         vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
00246         init_vlc (&vlc_tab_run, 5, 6,
00247             vlc_tab_run_huffbits, 1, 1,
00248             vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00249 
00250         fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
00251         fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
00252         init_vlc (&fft_level_exp_alt_vlc, 8, 28,
00253             fft_level_exp_alt_huffbits, 1, 1,
00254             fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00255 
00256 
00257         fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
00258         fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
00259         init_vlc (&fft_level_exp_vlc, 8, 20,
00260             fft_level_exp_huffbits, 1, 1,
00261             fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00262 
00263         fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
00264         fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
00265         init_vlc (&fft_stereo_exp_vlc, 6, 7,
00266             fft_stereo_exp_huffbits, 1, 1,
00267             fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00268 
00269         fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
00270         fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
00271         init_vlc (&fft_stereo_phase_vlc, 6, 9,
00272             fft_stereo_phase_huffbits, 1, 1,
00273             fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00274 
00275         vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]];
00276         vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
00277         init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
00278             vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
00279             vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00280 
00281         vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]];
00282         vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
00283         init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
00284             vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
00285             vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00286 
00287         vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]];
00288         vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
00289         init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
00290             vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
00291             vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00292 
00293         vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
00294         vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
00295         init_vlc (&vlc_tab_type30, 6, 9,
00296             vlc_tab_type30_huffbits, 1, 1,
00297             vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00298 
00299         vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
00300         vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
00301         init_vlc (&vlc_tab_type34, 5, 10,
00302             vlc_tab_type34_huffbits, 1, 1,
00303             vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00304 
00305         vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
00306         vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
00307         init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
00308             vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
00309             vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00310 
00311         vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
00312         vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
00313         init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
00314             vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
00315             vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00316 
00317         vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
00318         vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
00319         init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
00320             vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
00321             vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00322 
00323         vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
00324         vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
00325         init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
00326             vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
00327             vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00328 
00329         vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
00330         vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
00331         init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
00332             vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
00333             vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00334 
00335         vlcs_initialized=1;
00336     }
00337 }
00338 
00339 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
00340 {
00341     int value;
00342 
00343     value = get_vlc2(gb, vlc->table, vlc->bits, depth);
00344 
00345     /* stage-2, 3 bits exponent escape sequence */
00346     if (value-- == 0)
00347         value = get_bits (gb, get_bits (gb, 3) + 1);
00348 
00349     /* stage-3, optional */
00350     if (flag) {
00351         int tmp = vlc_stage3_values[value];
00352 
00353         if ((value & ~3) > 0)
00354             tmp += get_bits (gb, (value >> 2));
00355         value = tmp;
00356     }
00357 
00358     return value;
00359 }
00360 
00361 
00362 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
00363 {
00364     int value = qdm2_get_vlc (gb, vlc, 0, depth);
00365 
00366     return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
00367 }
00368 
00369 
00379 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
00380     int i;
00381 
00382     for (i=0; i < length; i++)
00383         value -= data[i];
00384 
00385     return (uint16_t)(value & 0xffff);
00386 }
00387 
00388 
00395 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
00396 {
00397     sub_packet->type = get_bits (gb, 8);
00398 
00399     if (sub_packet->type == 0) {
00400         sub_packet->size = 0;
00401         sub_packet->data = NULL;
00402     } else {
00403         sub_packet->size = get_bits (gb, 8);
00404 
00405       if (sub_packet->type & 0x80) {
00406           sub_packet->size <<= 8;
00407           sub_packet->size  |= get_bits (gb, 8);
00408           sub_packet->type  &= 0x7f;
00409       }
00410 
00411       if (sub_packet->type == 0x7f)
00412           sub_packet->type |= (get_bits (gb, 8) << 8);
00413 
00414       sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
00415     }
00416 
00417     av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
00418         sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
00419 }
00420 
00421 
00429 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
00430 {
00431     while (list != NULL && list->packet != NULL) {
00432         if (list->packet->type == type)
00433             return list;
00434         list = list->next;
00435     }
00436     return NULL;
00437 }
00438 
00439 
00446 static void average_quantized_coeffs (QDM2Context *q)
00447 {
00448     int i, j, n, ch, sum;
00449 
00450     n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
00451 
00452     for (ch = 0; ch < q->nb_channels; ch++)
00453         for (i = 0; i < n; i++) {
00454             sum = 0;
00455 
00456             for (j = 0; j < 8; j++)
00457                 sum += q->quantized_coeffs[ch][i][j];
00458 
00459             sum /= 8;
00460             if (sum > 0)
00461                 sum--;
00462 
00463             for (j=0; j < 8; j++)
00464                 q->quantized_coeffs[ch][i][j] = sum;
00465         }
00466 }
00467 
00468 
00476 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
00477 {
00478     int ch, j;
00479 
00480     FIX_NOISE_IDX(q->noise_idx);
00481 
00482     if (!q->nb_channels)
00483         return;
00484 
00485     for (ch = 0; ch < q->nb_channels; ch++)
00486         for (j = 0; j < 64; j++) {
00487             q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
00488             q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
00489         }
00490 }
00491 
00492 
00501 static int fix_coding_method_array(int sb, int channels,
00502                                    sb_int8_array coding_method)
00503 {
00504     int j,k;
00505     int ch;
00506     int run, case_val;
00507     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};
00508 
00509     for (ch = 0; ch < channels; ch++) {
00510         for (j = 0; j < 64; ) {
00511             if (coding_method[ch][sb][j] < 8)
00512                 return -1;
00513             if ((coding_method[ch][sb][j] - 8) > 22) {
00514                 run      = 1;
00515                 case_val = 8;
00516             } else {
00517                 switch (switchtable[coding_method[ch][sb][j]-8]) {
00518                     case 0: run = 10; case_val = 10; break;
00519                     case 1: run = 1; case_val = 16; break;
00520                     case 2: run = 5; case_val = 24; break;
00521                     case 3: run = 3; case_val = 30; break;
00522                     case 4: run = 1; case_val = 30; break;
00523                     case 5: run = 1; case_val = 8; break;
00524                     default: run = 1; case_val = 8; break;
00525                 }
00526             }
00527             for (k = 0; k < run; k++)
00528                 if (j + k < 128)
00529                     if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
00530                         if (k > 0) {
00531                            SAMPLES_NEEDED
00532                             //not debugged, almost never used
00533                             memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
00534                             memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
00535                         }
00536             j += run;
00537         }
00538     }
00539     return 0;
00540 }
00541 
00542 
00550 static void fill_tone_level_array (QDM2Context *q, int flag)
00551 {
00552     int i, sb, ch, sb_used;
00553     int tmp, tab;
00554 
00555     // This should never happen
00556     if (q->nb_channels <= 0)
00557         return;
00558 
00559     for (ch = 0; ch < q->nb_channels; ch++)
00560         for (sb = 0; sb < 30; sb++)
00561             for (i = 0; i < 8; i++) {
00562                 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
00563                     tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
00564                           q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
00565                 else
00566                     tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
00567                 if(tmp < 0)
00568                     tmp += 0xff;
00569                 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
00570             }
00571 
00572     sb_used = QDM2_SB_USED(q->sub_sampling);
00573 
00574     if ((q->superblocktype_2_3 != 0) && !flag) {
00575         for (sb = 0; sb < sb_used; sb++)
00576             for (ch = 0; ch < q->nb_channels; ch++)
00577                 for (i = 0; i < 64; i++) {
00578                     q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
00579                     if (q->tone_level_idx[ch][sb][i] < 0)
00580                         q->tone_level[ch][sb][i] = 0;
00581                     else
00582                         q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
00583                 }
00584     } else {
00585         tab = q->superblocktype_2_3 ? 0 : 1;
00586         for (sb = 0; sb < sb_used; sb++) {
00587             if ((sb >= 4) && (sb <= 23)) {
00588                 for (ch = 0; ch < q->nb_channels; ch++)
00589                     for (i = 0; i < 64; i++) {
00590                         tmp = q->tone_level_idx_base[ch][sb][i / 8] -
00591                               q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
00592                               q->tone_level_idx_mid[ch][sb - 4][i / 8] -
00593                               q->tone_level_idx_hi2[ch][sb - 4];
00594                         q->tone_level_idx[ch][sb][i] = tmp & 0xff;
00595                         if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
00596                             q->tone_level[ch][sb][i] = 0;
00597                         else
00598                             q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
00599                 }
00600             } else {
00601                 if (sb > 4) {
00602                     for (ch = 0; ch < q->nb_channels; ch++)
00603                         for (i = 0; i < 64; i++) {
00604                             tmp = q->tone_level_idx_base[ch][sb][i / 8] -
00605                                   q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
00606                                   q->tone_level_idx_hi2[ch][sb - 4];
00607                             q->tone_level_idx[ch][sb][i] = tmp & 0xff;
00608                             if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
00609                                 q->tone_level[ch][sb][i] = 0;
00610                             else
00611                                 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
00612                     }
00613                 } else {
00614                     for (ch = 0; ch < q->nb_channels; ch++)
00615                         for (i = 0; i < 64; i++) {
00616                             tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
00617                             if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
00618                                 q->tone_level[ch][sb][i] = 0;
00619                             else
00620                                 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
00621                         }
00622                 }
00623             }
00624         }
00625     }
00626 
00627     return;
00628 }
00629 
00630 
00645 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
00646                 sb_int8_array coding_method, int nb_channels,
00647                 int c, int superblocktype_2_3, int cm_table_select)
00648 {
00649     int ch, sb, j;
00650     int tmp, acc, esp_40, comp;
00651     int add1, add2, add3, add4;
00652     int64_t multres;
00653 
00654     // This should never happen
00655     if (nb_channels <= 0)
00656         return;
00657 
00658     if (!superblocktype_2_3) {
00659         /* This case is untested, no samples available */
00660         SAMPLES_NEEDED
00661         for (ch = 0; ch < nb_channels; ch++)
00662             for (sb = 0; sb < 30; sb++) {
00663                 for (j = 1; j < 63; j++) {  // The loop only iterates to 63 so the code doesn't overflow the buffer
00664                     add1 = tone_level_idx[ch][sb][j] - 10;
00665                     if (add1 < 0)
00666                         add1 = 0;
00667                     add2 = add3 = add4 = 0;
00668                     if (sb > 1) {
00669                         add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
00670                         if (add2 < 0)
00671                             add2 = 0;
00672                     }
00673                     if (sb > 0) {
00674                         add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
00675                         if (add3 < 0)
00676                             add3 = 0;
00677                     }
00678                     if (sb < 29) {
00679                         add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
00680                         if (add4 < 0)
00681                             add4 = 0;
00682                     }
00683                     tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
00684                     if (tmp < 0)
00685                         tmp = 0;
00686                     tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
00687                 }
00688                 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
00689             }
00690             acc = 0;
00691             for (ch = 0; ch < nb_channels; ch++)
00692                 for (sb = 0; sb < 30; sb++)
00693                     for (j = 0; j < 64; j++)
00694                         acc += tone_level_idx_temp[ch][sb][j];
00695 
00696             multres = 0x66666667 * (acc * 10);
00697             esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
00698             for (ch = 0;  ch < nb_channels; ch++)
00699                 for (sb = 0; sb < 30; sb++)
00700                     for (j = 0; j < 64; j++) {
00701                         comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
00702                         if (comp < 0)
00703                             comp += 0xff;
00704                         comp /= 256; // signed shift
00705                         switch(sb) {
00706                             case 0:
00707                                 if (comp < 30)
00708                                     comp = 30;
00709                                 comp += 15;
00710                                 break;
00711                             case 1:
00712                                 if (comp < 24)
00713                                     comp = 24;
00714                                 comp += 10;
00715                                 break;
00716                             case 2:
00717                             case 3:
00718                             case 4:
00719                                 if (comp < 16)
00720                                     comp = 16;
00721                         }
00722                         if (comp <= 5)
00723                             tmp = 0;
00724                         else if (comp <= 10)
00725                             tmp = 10;
00726                         else if (comp <= 16)
00727                             tmp = 16;
00728                         else if (comp <= 24)
00729                             tmp = -1;
00730                         else
00731                             tmp = 0;
00732                         coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
00733                     }
00734             for (sb = 0; sb < 30; sb++)
00735                 fix_coding_method_array(sb, nb_channels, coding_method);
00736             for (ch = 0; ch < nb_channels; ch++)
00737                 for (sb = 0; sb < 30; sb++)
00738                     for (j = 0; j < 64; j++)
00739                         if (sb >= 10) {
00740                             if (coding_method[ch][sb][j] < 10)
00741                                 coding_method[ch][sb][j] = 10;
00742                         } else {
00743                             if (sb >= 2) {
00744                                 if (coding_method[ch][sb][j] < 16)
00745                                     coding_method[ch][sb][j] = 16;
00746                             } else {
00747                                 if (coding_method[ch][sb][j] < 30)
00748                                     coding_method[ch][sb][j] = 30;
00749                             }
00750                         }
00751     } else { // superblocktype_2_3 != 0
00752         for (ch = 0; ch < nb_channels; ch++)
00753             for (sb = 0; sb < 30; sb++)
00754                 for (j = 0; j < 64; j++)
00755                     coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
00756     }
00757 
00758     return;
00759 }
00760 
00761 
00773 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
00774 {
00775     int sb, j, k, n, ch, run, channels;
00776     int joined_stereo, zero_encoding;
00777     int type34_first;
00778     float type34_div = 0;
00779     float type34_predictor;
00780     float samples[10], sign_bits[16];
00781 
00782     if (length == 0) {
00783         // If no data use noise
00784         for (sb=sb_min; sb < sb_max; sb++)
00785             build_sb_samples_from_noise (q, sb);
00786 
00787         return;
00788     }
00789 
00790     for (sb = sb_min; sb < sb_max; sb++) {
00791         channels = q->nb_channels;
00792 
00793         if (q->nb_channels <= 1 || sb < 12)
00794             joined_stereo = 0;
00795         else if (sb >= 24)
00796             joined_stereo = 1;
00797         else
00798             joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
00799 
00800         if (joined_stereo) {
00801             if (BITS_LEFT(length,gb) >= 16)
00802                 for (j = 0; j < 16; j++)
00803                     sign_bits[j] = get_bits1 (gb);
00804 
00805             for (j = 0; j < 64; j++)
00806                 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
00807                     q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
00808 
00809             if (fix_coding_method_array(sb, q->nb_channels,
00810                                             q->coding_method)) {
00811                 build_sb_samples_from_noise(q, sb);
00812                 continue;
00813             }
00814             channels = 1;
00815         }
00816 
00817         for (ch = 0; ch < channels; ch++) {
00818             FIX_NOISE_IDX(q->noise_idx);
00819             zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
00820             type34_predictor = 0.0;
00821             type34_first = 1;
00822 
00823             for (j = 0; j < 128; ) {
00824                 switch (q->coding_method[ch][sb][j / 2]) {
00825                     case 8:
00826                         if (BITS_LEFT(length,gb) >= 10) {
00827                             if (zero_encoding) {
00828                                 for (k = 0; k < 5; k++) {
00829                                     if ((j + 2 * k) >= 128)
00830                                         break;
00831                                     samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
00832                                 }
00833                             } else {
00834                                 n = get_bits(gb, 8);
00835                                 for (k = 0; k < 5; k++)
00836                                     samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
00837                             }
00838                             for (k = 0; k < 5; k++)
00839                                 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
00840                         } else {
00841                             for (k = 0; k < 10; k++)
00842                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
00843                         }
00844                         run = 10;
00845                         break;
00846 
00847                     case 10:
00848                         if (BITS_LEFT(length,gb) >= 1) {
00849                             float f = 0.81;
00850 
00851                             if (get_bits1(gb))
00852                                 f = -f;
00853                             f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
00854                             samples[0] = f;
00855                         } else {
00856                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00857                         }
00858                         run = 1;
00859                         break;
00860 
00861                     case 16:
00862                         if (BITS_LEFT(length,gb) >= 10) {
00863                             if (zero_encoding) {
00864                                 for (k = 0; k < 5; k++) {
00865                                     if ((j + k) >= 128)
00866                                         break;
00867                                     samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
00868                                 }
00869                             } else {
00870                                 n = get_bits (gb, 8);
00871                                 for (k = 0; k < 5; k++)
00872                                     samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
00873                             }
00874                         } else {
00875                             for (k = 0; k < 5; k++)
00876                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
00877                         }
00878                         run = 5;
00879                         break;
00880 
00881                     case 24:
00882                         if (BITS_LEFT(length,gb) >= 7) {
00883                             n = get_bits(gb, 7);
00884                             for (k = 0; k < 3; k++)
00885                                 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
00886                         } else {
00887                             for (k = 0; k < 3; k++)
00888                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
00889                         }
00890                         run = 3;
00891                         break;
00892 
00893                     case 30:
00894                         if (BITS_LEFT(length,gb) >= 4) {
00895                             unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
00896                             if (index < FF_ARRAY_ELEMS(type30_dequant)) {
00897                                 samples[0] = type30_dequant[index];
00898                             } else
00899                                 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00900                         } else
00901                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00902 
00903                         run = 1;
00904                         break;
00905 
00906                     case 34:
00907                         if (BITS_LEFT(length,gb) >= 7) {
00908                             if (type34_first) {
00909                                 type34_div = (float)(1 << get_bits(gb, 2));
00910                                 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
00911                                 type34_predictor = samples[0];
00912                                 type34_first = 0;
00913                             } else {
00914                                 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
00915                                 if (index < FF_ARRAY_ELEMS(type34_delta)) {
00916                                     samples[0] = type34_delta[index] / type34_div + type34_predictor;
00917                                     type34_predictor = samples[0];
00918                                 } else
00919                                     samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00920                             }
00921                         } else {
00922                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00923                         }
00924                         run = 1;
00925                         break;
00926 
00927                     default:
00928                         samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00929                         run = 1;
00930                         break;
00931                 }
00932 
00933                 if (joined_stereo) {
00934                     for (k = 0; k < run && j + k < 128; k++) {
00935                         q->sb_samples[0][j + k][sb] =
00936                             q->tone_level[0][sb][(j + k) / 2] * samples[k];
00937                         if (q->nb_channels == 2) {
00938                             if (sign_bits[(j + k) / 8])
00939                                 q->sb_samples[1][j + k][sb] =
00940                                     q->tone_level[1][sb][(j + k) / 2] * -samples[k];
00941                             else
00942                                 q->sb_samples[1][j + k][sb] =
00943                                     q->tone_level[1][sb][(j + k) / 2] * samples[k];
00944                         }
00945                     }
00946                 } else {
00947                     for (k = 0; k < run; k++)
00948                         if ((j + k) < 128)
00949                             q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
00950                 }
00951 
00952                 j += run;
00953             } // j loop
00954         } // channel loop
00955     } // subband loop
00956 }
00957 
00958 
00968 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
00969 {
00970     int i, k, run, level, diff;
00971 
00972     if (BITS_LEFT(length,gb) < 16)
00973         return;
00974     level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
00975 
00976     quantized_coeffs[0] = level;
00977 
00978     for (i = 0; i < 7; ) {
00979         if (BITS_LEFT(length,gb) < 16)
00980             break;
00981         run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
00982 
00983         if (BITS_LEFT(length,gb) < 16)
00984             break;
00985         diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
00986 
00987         for (k = 1; k <= run; k++)
00988             quantized_coeffs[i + k] = (level + ((k * diff) / run));
00989 
00990         level += diff;
00991         i += run;
00992     }
00993 }
00994 
00995 
01005 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
01006 {
01007     int sb, j, k, n, ch;
01008 
01009     for (ch = 0; ch < q->nb_channels; ch++) {
01010         init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
01011 
01012         if (BITS_LEFT(length,gb) < 16) {
01013             memset(q->quantized_coeffs[ch][0], 0, 8);
01014             break;
01015         }
01016     }
01017 
01018     n = q->sub_sampling + 1;
01019 
01020     for (sb = 0; sb < n; sb++)
01021         for (ch = 0; ch < q->nb_channels; ch++)
01022             for (j = 0; j < 8; j++) {
01023                 if (BITS_LEFT(length,gb) < 1)
01024                     break;
01025                 if (get_bits1(gb)) {
01026                     for (k=0; k < 8; k++) {
01027                         if (BITS_LEFT(length,gb) < 16)
01028                             break;
01029                         q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
01030                     }
01031                 } else {
01032                     for (k=0; k < 8; k++)
01033                         q->tone_level_idx_hi1[ch][sb][j][k] = 0;
01034                 }
01035             }
01036 
01037     n = QDM2_SB_USED(q->sub_sampling) - 4;
01038 
01039     for (sb = 0; sb < n; sb++)
01040         for (ch = 0; ch < q->nb_channels; ch++) {
01041             if (BITS_LEFT(length,gb) < 16)
01042                 break;
01043             q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
01044             if (sb > 19)
01045                 q->tone_level_idx_hi2[ch][sb] -= 16;
01046             else
01047                 for (j = 0; j < 8; j++)
01048                     q->tone_level_idx_mid[ch][sb][j] = -16;
01049         }
01050 
01051     n = QDM2_SB_USED(q->sub_sampling) - 5;
01052 
01053     for (sb = 0; sb < n; sb++)
01054         for (ch = 0; ch < q->nb_channels; ch++)
01055             for (j = 0; j < 8; j++) {
01056                 if (BITS_LEFT(length,gb) < 16)
01057                     break;
01058                 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
01059             }
01060 }
01061 
01068 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
01069 {
01070     GetBitContext gb;
01071     int i, j, k, n, ch, run, level, diff;
01072 
01073     init_get_bits(&gb, node->packet->data, node->packet->size*8);
01074 
01075     n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
01076 
01077     for (i = 1; i < n; i++)
01078         for (ch=0; ch < q->nb_channels; ch++) {
01079             level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
01080             q->quantized_coeffs[ch][i][0] = level;
01081 
01082             for (j = 0; j < (8 - 1); ) {
01083                 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
01084                 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
01085 
01086                 for (k = 1; k <= run; k++)
01087                     q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
01088 
01089                 level += diff;
01090                 j += run;
01091             }
01092         }
01093 
01094     for (ch = 0; ch < q->nb_channels; ch++)
01095         for (i = 0; i < 8; i++)
01096             q->quantized_coeffs[ch][0][i] = 0;
01097 }
01098 
01099 
01107 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
01108 {
01109     GetBitContext gb;
01110 
01111     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
01112 
01113     if (length != 0) {
01114         init_tone_level_dequantization(q, &gb, length);
01115         fill_tone_level_array(q, 1);
01116     } else {
01117         fill_tone_level_array(q, 0);
01118     }
01119 }
01120 
01121 
01129 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
01130 {
01131     GetBitContext gb;
01132 
01133     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
01134     if (length >= 32) {
01135         int c = get_bits (&gb, 13);
01136 
01137         if (c > 3)
01138             fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
01139                                       q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
01140     }
01141 
01142     synthfilt_build_sb_samples(q, &gb, length, 0, 8);
01143 }
01144 
01145 
01153 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
01154 {
01155     GetBitContext gb;
01156 
01157     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
01158     synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
01159 }
01160 
01161 /*
01162  * Process new subpackets for synthesis filter
01163  *
01164  * @param q       context
01165  * @param list    list with synthesis filter packets (list D)
01166  */
01167 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
01168 {
01169     QDM2SubPNode *nodes[4];
01170 
01171     nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
01172     if (nodes[0] != NULL)
01173         process_subpacket_9(q, nodes[0]);
01174 
01175     nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
01176     if (nodes[1] != NULL)
01177         process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
01178     else
01179         process_subpacket_10(q, NULL, 0);
01180 
01181     nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
01182     if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
01183         process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
01184     else
01185         process_subpacket_11(q, NULL, 0);
01186 
01187     nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
01188     if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
01189         process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
01190     else
01191         process_subpacket_12(q, NULL, 0);
01192 }
01193 
01194 
01195 /*
01196  * Decode superblock, fill packet lists.
01197  *
01198  * @param q    context
01199  */
01200 static void qdm2_decode_super_block (QDM2Context *q)
01201 {
01202     GetBitContext gb;
01203     QDM2SubPacket header, *packet;
01204     int i, packet_bytes, sub_packet_size, sub_packets_D;
01205     unsigned int next_index = 0;
01206 
01207     memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
01208     memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
01209     memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
01210 
01211     q->sub_packets_B = 0;
01212     sub_packets_D = 0;
01213 
01214     average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
01215 
01216     init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
01217     qdm2_decode_sub_packet_header(&gb, &header);
01218 
01219     if (header.type < 2 || header.type >= 8) {
01220         q->has_errors = 1;
01221         av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
01222         return;
01223     }
01224 
01225     q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
01226     packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
01227 
01228     init_get_bits(&gb, header.data, header.size*8);
01229 
01230     if (header.type == 2 || header.type == 4 || header.type == 5) {
01231         int csum  = 257 * get_bits(&gb, 8);
01232             csum +=   2 * get_bits(&gb, 8);
01233 
01234         csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
01235 
01236         if (csum != 0) {
01237             q->has_errors = 1;
01238             av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
01239             return;
01240         }
01241     }
01242 
01243     q->sub_packet_list_B[0].packet = NULL;
01244     q->sub_packet_list_D[0].packet = NULL;
01245 
01246     for (i = 0; i < 6; i++)
01247         if (--q->fft_level_exp[i] < 0)
01248             q->fft_level_exp[i] = 0;
01249 
01250     for (i = 0; packet_bytes > 0; i++) {
01251         int j;
01252 
01253         q->sub_packet_list_A[i].next = NULL;
01254 
01255         if (i > 0) {
01256             q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
01257 
01258             /* seek to next block */
01259             init_get_bits(&gb, header.data, header.size*8);
01260             skip_bits(&gb, next_index*8);
01261 
01262             if (next_index >= header.size)
01263                 break;
01264         }
01265 
01266         /* decode subpacket */
01267         packet = &q->sub_packets[i];
01268         qdm2_decode_sub_packet_header(&gb, packet);
01269         next_index = packet->size + get_bits_count(&gb) / 8;
01270         sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
01271 
01272         if (packet->type == 0)
01273             break;
01274 
01275         if (sub_packet_size > packet_bytes) {
01276             if (packet->type != 10 && packet->type != 11 && packet->type != 12)
01277                 break;
01278             packet->size += packet_bytes - sub_packet_size;
01279         }
01280 
01281         packet_bytes -= sub_packet_size;
01282 
01283         /* add subpacket to 'all subpackets' list */
01284         q->sub_packet_list_A[i].packet = packet;
01285 
01286         /* add subpacket to related list */
01287         if (packet->type == 8) {
01288             SAMPLES_NEEDED_2("packet type 8");
01289             return;
01290         } else if (packet->type >= 9 && packet->type <= 12) {
01291             /* packets for MPEG Audio like Synthesis Filter */
01292             QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
01293         } else if (packet->type == 13) {
01294             for (j = 0; j < 6; j++)
01295                 q->fft_level_exp[j] = get_bits(&gb, 6);
01296         } else if (packet->type == 14) {
01297             for (j = 0; j < 6; j++)
01298                 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
01299         } else if (packet->type == 15) {
01300             SAMPLES_NEEDED_2("packet type 15")
01301             return;
01302         } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
01303             /* packets for FFT */
01304             QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
01305         }
01306     } // Packet bytes loop
01307 
01308 /* **************************************************************** */
01309     if (q->sub_packet_list_D[0].packet != NULL) {
01310         process_synthesis_subpackets(q, q->sub_packet_list_D);
01311         q->do_synth_filter = 1;
01312     } else if (q->do_synth_filter) {
01313         process_subpacket_10(q, NULL, 0);
01314         process_subpacket_11(q, NULL, 0);
01315         process_subpacket_12(q, NULL, 0);
01316     }
01317 /* **************************************************************** */
01318 }
01319 
01320 
01321 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
01322                        int offset, int duration, int channel,
01323                        int exp, int phase)
01324 {
01325     if (q->fft_coefs_min_index[duration] < 0)
01326         q->fft_coefs_min_index[duration] = q->fft_coefs_index;
01327 
01328     q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
01329     q->fft_coefs[q->fft_coefs_index].channel = channel;
01330     q->fft_coefs[q->fft_coefs_index].offset = offset;
01331     q->fft_coefs[q->fft_coefs_index].exp = exp;
01332     q->fft_coefs[q->fft_coefs_index].phase = phase;
01333     q->fft_coefs_index++;
01334 }
01335 
01336 
01337 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
01338 {
01339     int channel, stereo, phase, exp;
01340     int local_int_4,  local_int_8,  stereo_phase,  local_int_10;
01341     int local_int_14, stereo_exp, local_int_20, local_int_28;
01342     int n, offset;
01343 
01344     local_int_4 = 0;
01345     local_int_28 = 0;
01346     local_int_20 = 2;
01347     local_int_8 = (4 - duration);
01348     local_int_10 = 1 << (q->group_order - duration - 1);
01349     offset = 1;
01350 
01351     while (1) {
01352         if (q->superblocktype_2_3) {
01353             while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
01354                 offset = 1;
01355                 if (n == 0) {
01356                     local_int_4 += local_int_10;
01357                     local_int_28 += (1 << local_int_8);
01358                 } else {
01359                     local_int_4 += 8*local_int_10;
01360                     local_int_28 += (8 << local_int_8);
01361                 }
01362             }
01363             offset += (n - 2);
01364         } else {
01365             offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
01366             while (offset >= (local_int_10 - 1)) {
01367                 offset += (1 - (local_int_10 - 1));
01368                 local_int_4  += local_int_10;
01369                 local_int_28 += (1 << local_int_8);
01370             }
01371         }
01372 
01373         if (local_int_4 >= q->group_size)
01374             return;
01375 
01376         local_int_14 = (offset >> local_int_8);
01377         if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
01378             return;
01379 
01380         if (q->nb_channels > 1) {
01381             channel = get_bits1(gb);
01382             stereo = get_bits1(gb);
01383         } else {
01384             channel = 0;
01385             stereo = 0;
01386         }
01387 
01388         exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
01389         exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
01390         exp = (exp < 0) ? 0 : exp;
01391 
01392         phase = get_bits(gb, 3);
01393         stereo_exp = 0;
01394         stereo_phase = 0;
01395 
01396         if (stereo) {
01397             stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
01398             stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
01399             if (stereo_phase < 0)
01400                 stereo_phase += 8;
01401         }
01402 
01403         if (q->frequency_range > (local_int_14 + 1)) {
01404             int sub_packet = (local_int_20 + local_int_28);
01405 
01406             qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
01407             if (stereo)
01408                 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
01409         }
01410 
01411         offset++;
01412     }
01413 }
01414 
01415 
01416 static void qdm2_decode_fft_packets (QDM2Context *q)
01417 {
01418     int i, j, min, max, value, type, unknown_flag;
01419     GetBitContext gb;
01420 
01421     if (q->sub_packet_list_B[0].packet == NULL)
01422         return;
01423 
01424     /* reset minimum indexes for FFT coefficients */
01425     q->fft_coefs_index = 0;
01426     for (i=0; i < 5; i++)
01427         q->fft_coefs_min_index[i] = -1;
01428 
01429     /* process subpackets ordered by type, largest type first */
01430     for (i = 0, max = 256; i < q->sub_packets_B; i++) {
01431         QDM2SubPacket *packet= NULL;
01432 
01433         /* find subpacket with largest type less than max */
01434         for (j = 0, min = 0; j < q->sub_packets_B; j++) {
01435             value = q->sub_packet_list_B[j].packet->type;
01436             if (value > min && value < max) {
01437                 min = value;
01438                 packet = q->sub_packet_list_B[j].packet;
01439             }
01440         }
01441 
01442         max = min;
01443 
01444         /* check for errors (?) */
01445         if (!packet)
01446             return;
01447 
01448         if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
01449             return;
01450 
01451         /* decode FFT tones */
01452         init_get_bits (&gb, packet->data, packet->size*8);
01453 
01454         if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
01455             unknown_flag = 1;
01456         else
01457             unknown_flag = 0;
01458 
01459         type = packet->type;
01460 
01461         if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
01462             int duration = q->sub_sampling + 5 - (type & 15);
01463 
01464             if (duration >= 0 && duration < 4)
01465                 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
01466         } else if (type == 31) {
01467             for (j=0; j < 4; j++)
01468                 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
01469         } else if (type == 46) {
01470             for (j=0; j < 6; j++)
01471                 q->fft_level_exp[j] = get_bits(&gb, 6);
01472             for (j=0; j < 4; j++)
01473             qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
01474         }
01475     } // Loop on B packets
01476 
01477     /* calculate maximum indexes for FFT coefficients */
01478     for (i = 0, j = -1; i < 5; i++)
01479         if (q->fft_coefs_min_index[i] >= 0) {
01480             if (j >= 0)
01481                 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
01482             j = i;
01483         }
01484     if (j >= 0)
01485         q->fft_coefs_max_index[j] = q->fft_coefs_index;
01486 }
01487 
01488 
01489 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
01490 {
01491    float level, f[6];
01492    int i;
01493    QDM2Complex c;
01494    const double iscale = 2.0*M_PI / 512.0;
01495 
01496     tone->phase += tone->phase_shift;
01497 
01498     /* calculate current level (maximum amplitude) of tone */
01499     level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
01500     c.im = level * sin(tone->phase*iscale);
01501     c.re = level * cos(tone->phase*iscale);
01502 
01503     /* generate FFT coefficients for tone */
01504     if (tone->duration >= 3 || tone->cutoff >= 3) {
01505         tone->complex[0].im += c.im;
01506         tone->complex[0].re += c.re;
01507         tone->complex[1].im -= c.im;
01508         tone->complex[1].re -= c.re;
01509     } else {
01510         f[1] = -tone->table[4];
01511         f[0] =  tone->table[3] - tone->table[0];
01512         f[2] =  1.0 - tone->table[2] - tone->table[3];
01513         f[3] =  tone->table[1] + tone->table[4] - 1.0;
01514         f[4] =  tone->table[0] - tone->table[1];
01515         f[5] =  tone->table[2];
01516         for (i = 0; i < 2; i++) {
01517             tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
01518             tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
01519         }
01520         for (i = 0; i < 4; i++) {
01521             tone->complex[i].re += c.re * f[i+2];
01522             tone->complex[i].im += c.im * f[i+2];
01523         }
01524     }
01525 
01526     /* copy the tone if it has not yet died out */
01527     if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
01528       memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
01529       q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
01530     }
01531 }
01532 
01533 
01534 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
01535 {
01536     int i, j, ch;
01537     const double iscale = 0.25 * M_PI;
01538 
01539     for (ch = 0; ch < q->channels; ch++) {
01540         memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
01541     }
01542 
01543 
01544     /* apply FFT tones with duration 4 (1 FFT period) */
01545     if (q->fft_coefs_min_index[4] >= 0)
01546         for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
01547             float level;
01548             QDM2Complex c;
01549 
01550             if (q->fft_coefs[i].sub_packet != sub_packet)
01551                 break;
01552 
01553             ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
01554             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];
01555 
01556             c.re = level * cos(q->fft_coefs[i].phase * iscale);
01557             c.im = level * sin(q->fft_coefs[i].phase * iscale);
01558             q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
01559             q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
01560             q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
01561             q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
01562         }
01563 
01564     /* generate existing FFT tones */
01565     for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
01566         qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
01567         q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
01568     }
01569 
01570     /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
01571     for (i = 0; i < 4; i++)
01572         if (q->fft_coefs_min_index[i] >= 0) {
01573             for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
01574                 int offset, four_i;
01575                 FFTTone tone;
01576 
01577                 if (q->fft_coefs[j].sub_packet != sub_packet)
01578                     break;
01579 
01580                 four_i = (4 - i);
01581                 offset = q->fft_coefs[j].offset >> four_i;
01582                 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
01583 
01584                 if (offset < q->frequency_range) {
01585                     if (offset < 2)
01586                         tone.cutoff = offset;
01587                     else
01588                         tone.cutoff = (offset >= 60) ? 3 : 2;
01589 
01590                     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];
01591                     tone.complex = &q->fft.complex[ch][offset];
01592                     tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
01593                     tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
01594                     tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
01595                     tone.duration = i;
01596                     tone.time_index = 0;
01597 
01598                     qdm2_fft_generate_tone(q, &tone);
01599                 }
01600             }
01601             q->fft_coefs_min_index[i] = j;
01602         }
01603 }
01604 
01605 
01606 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
01607 {
01608     const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
01609     int i;
01610     q->fft.complex[channel][0].re *= 2.0f;
01611     q->fft.complex[channel][0].im = 0.0f;
01612     q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
01613     /* add samples to output buffer */
01614     for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
01615         q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
01616 }
01617 
01618 
01623 static void qdm2_synthesis_filter (QDM2Context *q, int index)
01624 {
01625     int i, k, ch, sb_used, sub_sampling, dither_state = 0;
01626 
01627     /* copy sb_samples */
01628     sb_used = QDM2_SB_USED(q->sub_sampling);
01629 
01630     for (ch = 0; ch < q->channels; ch++)
01631         for (i = 0; i < 8; i++)
01632             for (k=sb_used; k < SBLIMIT; k++)
01633                 q->sb_samples[ch][(8 * index) + i][k] = 0;
01634 
01635     for (ch = 0; ch < q->nb_channels; ch++) {
01636         float *samples_ptr = q->samples + ch;
01637 
01638         for (i = 0; i < 8; i++) {
01639             ff_mpa_synth_filter_float(&q->mpadsp,
01640                 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
01641                 ff_mpa_synth_window_float, &dither_state,
01642                 samples_ptr, q->nb_channels,
01643                 q->sb_samples[ch][(8 * index) + i]);
01644             samples_ptr += 32 * q->nb_channels;
01645         }
01646     }
01647 
01648     /* add samples to output buffer */
01649     sub_sampling = (4 >> q->sub_sampling);
01650 
01651     for (ch = 0; ch < q->channels; ch++)
01652         for (i = 0; i < q->frame_size; i++)
01653             q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
01654 }
01655 
01656 
01662 static av_cold void qdm2_init(QDM2Context *q) {
01663     static int initialized = 0;
01664 
01665     if (initialized != 0)
01666         return;
01667     initialized = 1;
01668 
01669     qdm2_init_vlc();
01670     ff_mpa_synth_init_float(ff_mpa_synth_window_float);
01671     softclip_table_init();
01672     rnd_table_init();
01673     init_noise_samples();
01674 
01675     av_log(NULL, AV_LOG_DEBUG, "init done\n");
01676 }
01677 
01678 
01679 #if 0
01680 static void dump_context(QDM2Context *q)
01681 {
01682     int i;
01683 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
01684     PRINT("compressed_data",q->compressed_data);
01685     PRINT("compressed_size",q->compressed_size);
01686     PRINT("frame_size",q->frame_size);
01687     PRINT("checksum_size",q->checksum_size);
01688     PRINT("channels",q->channels);
01689     PRINT("nb_channels",q->nb_channels);
01690     PRINT("fft_frame_size",q->fft_frame_size);
01691     PRINT("fft_size",q->fft_size);
01692     PRINT("sub_sampling",q->sub_sampling);
01693     PRINT("fft_order",q->fft_order);
01694     PRINT("group_order",q->group_order);
01695     PRINT("group_size",q->group_size);
01696     PRINT("sub_packet",q->sub_packet);
01697     PRINT("frequency_range",q->frequency_range);
01698     PRINT("has_errors",q->has_errors);
01699     PRINT("fft_tone_end",q->fft_tone_end);
01700     PRINT("fft_tone_start",q->fft_tone_start);
01701     PRINT("fft_coefs_index",q->fft_coefs_index);
01702     PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
01703     PRINT("cm_table_select",q->cm_table_select);
01704     PRINT("noise_idx",q->noise_idx);
01705 
01706     for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
01707     {
01708     FFTTone *t = &q->fft_tones[i];
01709 
01710     av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
01711     av_log(NULL,AV_LOG_DEBUG,"  level = %f\n", t->level);
01712 //  PRINT(" level", t->level);
01713     PRINT(" phase", t->phase);
01714     PRINT(" phase_shift", t->phase_shift);
01715     PRINT(" duration", t->duration);
01716     PRINT(" samples_im", t->samples_im);
01717     PRINT(" samples_re", t->samples_re);
01718     PRINT(" table", t->table);
01719     }
01720 
01721 }
01722 #endif
01723 
01724 
01728 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
01729 {
01730     QDM2Context *s = avctx->priv_data;
01731     uint8_t *extradata;
01732     int extradata_size;
01733     int tmp_val, tmp, size;
01734 
01735     /* extradata parsing
01736 
01737     Structure:
01738     wave {
01739         frma (QDM2)
01740         QDCA
01741         QDCP
01742     }
01743 
01744     32  size (including this field)
01745     32  tag (=frma)
01746     32  type (=QDM2 or QDMC)
01747 
01748     32  size (including this field, in bytes)
01749     32  tag (=QDCA) // maybe mandatory parameters
01750     32  unknown (=1)
01751     32  channels (=2)
01752     32  samplerate (=44100)
01753     32  bitrate (=96000)
01754     32  block size (=4096)
01755     32  frame size (=256) (for one channel)
01756     32  packet size (=1300)
01757 
01758     32  size (including this field, in bytes)
01759     32  tag (=QDCP) // maybe some tuneable parameters
01760     32  float1 (=1.0)
01761     32  zero ?
01762     32  float2 (=1.0)
01763     32  float3 (=1.0)
01764     32  unknown (27)
01765     32  unknown (8)
01766     32  zero ?
01767     */
01768 
01769     if (!avctx->extradata || (avctx->extradata_size < 48)) {
01770         av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
01771         return -1;
01772     }
01773 
01774     extradata = avctx->extradata;
01775     extradata_size = avctx->extradata_size;
01776 
01777     while (extradata_size > 7) {
01778         if (!memcmp(extradata, "frmaQDM", 7))
01779             break;
01780         extradata++;
01781         extradata_size--;
01782     }
01783 
01784     if (extradata_size < 12) {
01785         av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
01786                extradata_size);
01787         return -1;
01788     }
01789 
01790     if (memcmp(extradata, "frmaQDM", 7)) {
01791         av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
01792         return -1;
01793     }
01794 
01795     if (extradata[7] == 'C') {
01796 //        s->is_qdmc = 1;
01797         av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
01798         return -1;
01799     }
01800 
01801     extradata += 8;
01802     extradata_size -= 8;
01803 
01804     size = AV_RB32(extradata);
01805 
01806     if(size > extradata_size){
01807         av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
01808                extradata_size, size);
01809         return -1;
01810     }
01811 
01812     extradata += 4;
01813     av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
01814     if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
01815         av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
01816         return -1;
01817     }
01818 
01819     extradata += 8;
01820 
01821     avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
01822     extradata += 4;
01823     if (s->channels > MPA_MAX_CHANNELS)
01824         return AVERROR_INVALIDDATA;
01825 
01826     avctx->sample_rate = AV_RB32(extradata);
01827     extradata += 4;
01828 
01829     avctx->bit_rate = AV_RB32(extradata);
01830     extradata += 4;
01831 
01832     s->group_size = AV_RB32(extradata);
01833     extradata += 4;
01834 
01835     s->fft_size = AV_RB32(extradata);
01836     extradata += 4;
01837 
01838     s->checksum_size = AV_RB32(extradata);
01839     if (s->checksum_size >= 1U << 28) {
01840         av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
01841         return AVERROR_INVALIDDATA;
01842     }
01843 
01844     s->fft_order = av_log2(s->fft_size) + 1;
01845     s->fft_frame_size = 2 * s->fft_size; // complex has two floats
01846 
01847     // something like max decodable tones
01848     s->group_order = av_log2(s->group_size) + 1;
01849     s->frame_size = s->group_size / 16; // 16 iterations per super block
01850     if (s->frame_size > QDM2_MAX_FRAME_SIZE)
01851         return AVERROR_INVALIDDATA;
01852 
01853     s->sub_sampling = s->fft_order - 7;
01854     s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
01855 
01856     switch ((s->sub_sampling * 2 + s->channels - 1)) {
01857         case 0: tmp = 40; break;
01858         case 1: tmp = 48; break;
01859         case 2: tmp = 56; break;
01860         case 3: tmp = 72; break;
01861         case 4: tmp = 80; break;
01862         case 5: tmp = 100;break;
01863         default: tmp=s->sub_sampling; break;
01864     }
01865     tmp_val = 0;
01866     if ((tmp * 1000) < avctx->bit_rate)  tmp_val = 1;
01867     if ((tmp * 1440) < avctx->bit_rate)  tmp_val = 2;
01868     if ((tmp * 1760) < avctx->bit_rate)  tmp_val = 3;
01869     if ((tmp * 2240) < avctx->bit_rate)  tmp_val = 4;
01870     s->cm_table_select = tmp_val;
01871 
01872     if (s->sub_sampling == 0)
01873         tmp = 7999;
01874     else
01875         tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
01876     /*
01877     0: 7999 -> 0
01878     1: 20000 -> 2
01879     2: 28000 -> 2
01880     */
01881     if (tmp < 8000)
01882         s->coeff_per_sb_select = 0;
01883     else if (tmp <= 16000)
01884         s->coeff_per_sb_select = 1;
01885     else
01886         s->coeff_per_sb_select = 2;
01887 
01888     // Fail on unknown fft order
01889     if ((s->fft_order < 7) || (s->fft_order > 9)) {
01890         av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
01891         return -1;
01892     }
01893     if (s->fft_size != (1 << (s->fft_order - 1))) {
01894         av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
01895         return AVERROR_INVALIDDATA;
01896     }
01897 
01898     ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
01899     ff_mpadsp_init(&s->mpadsp);
01900 
01901     qdm2_init(s);
01902 
01903     avctx->sample_fmt = AV_SAMPLE_FMT_S16;
01904 
01905     avcodec_get_frame_defaults(&s->frame);
01906     avctx->coded_frame = &s->frame;
01907 
01908 //    dump_context(s);
01909     return 0;
01910 }
01911 
01912 
01913 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
01914 {
01915     QDM2Context *s = avctx->priv_data;
01916 
01917     ff_rdft_end(&s->rdft_ctx);
01918 
01919     return 0;
01920 }
01921 
01922 
01923 static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
01924 {
01925     int ch, i;
01926     const int frame_size = (q->frame_size * q->channels);
01927 
01928     /* select input buffer */
01929     q->compressed_data = in;
01930     q->compressed_size = q->checksum_size;
01931 
01932 //  dump_context(q);
01933 
01934     /* copy old block, clear new block of output samples */
01935     memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
01936     memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
01937 
01938     /* decode block of QDM2 compressed data */
01939     if (q->sub_packet == 0) {
01940         q->has_errors = 0; // zero it for a new super block
01941         av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
01942         qdm2_decode_super_block(q);
01943     }
01944 
01945     /* parse subpackets */
01946     if (!q->has_errors) {
01947         if (q->sub_packet == 2)
01948             qdm2_decode_fft_packets(q);
01949 
01950         qdm2_fft_tone_synthesizer(q, q->sub_packet);
01951     }
01952 
01953     /* sound synthesis stage 1 (FFT) */
01954     for (ch = 0; ch < q->channels; ch++) {
01955         qdm2_calculate_fft(q, ch, q->sub_packet);
01956 
01957         if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
01958             SAMPLES_NEEDED_2("has errors, and C list is not empty")
01959             return -1;
01960         }
01961     }
01962 
01963     /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
01964     if (!q->has_errors && q->do_synth_filter)
01965         qdm2_synthesis_filter(q, q->sub_packet);
01966 
01967     q->sub_packet = (q->sub_packet + 1) % 16;
01968 
01969     /* clip and convert output float[] to 16bit signed samples */
01970     for (i = 0; i < frame_size; i++) {
01971         int value = (int)q->output_buffer[i];
01972 
01973         if (value > SOFTCLIP_THRESHOLD)
01974             value = (value >  HARDCLIP_THRESHOLD) ?  32767 :  softclip_table[ value - SOFTCLIP_THRESHOLD];
01975         else if (value < -SOFTCLIP_THRESHOLD)
01976             value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
01977 
01978         out[i] = value;
01979     }
01980 
01981     return 0;
01982 }
01983 
01984 
01985 static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
01986                              int *got_frame_ptr, AVPacket *avpkt)
01987 {
01988     const uint8_t *buf = avpkt->data;
01989     int buf_size = avpkt->size;
01990     QDM2Context *s = avctx->priv_data;
01991     int16_t *out;
01992     int i, ret;
01993 
01994     if(!buf)
01995         return 0;
01996     if(buf_size < s->checksum_size)
01997         return -1;
01998 
01999     /* get output buffer */
02000     s->frame.nb_samples = 16 * s->frame_size;
02001     if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
02002         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
02003         return ret;
02004     }
02005     out = (int16_t *)s->frame.data[0];
02006 
02007     for (i = 0; i < 16; i++) {
02008         if (qdm2_decode(s, buf, out) < 0)
02009             return -1;
02010         out += s->channels * s->frame_size;
02011     }
02012 
02013     *got_frame_ptr   = 1;
02014     *(AVFrame *)data = s->frame;
02015 
02016     return s->checksum_size;
02017 }
02018 
02019 AVCodec ff_qdm2_decoder =
02020 {
02021     .name = "qdm2",
02022     .type = AVMEDIA_TYPE_AUDIO,
02023     .id = CODEC_ID_QDM2,
02024     .priv_data_size = sizeof(QDM2Context),
02025     .init = qdm2_decode_init,
02026     .close = qdm2_decode_close,
02027     .decode = qdm2_decode_frame,
02028     .capabilities = CODEC_CAP_DR1,
02029     .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
02030 };