Libav

libavcodec/qdm2.c

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00001 /*
00002  * QDM2 compatible decoder
00003  * Copyright (c) 2003 Ewald Snel
00004  * Copyright (c) 2005 Benjamin Larsson
00005  * Copyright (c) 2005 Alex Beregszaszi
00006  * Copyright (c) 2005 Roberto Togni
00007  *
00008  * This file is part of FFmpeg.
00009  *
00010  * FFmpeg is free software; you can redistribute it and/or
00011  * modify it under the terms of the GNU Lesser General Public
00012  * License as published by the Free Software Foundation; either
00013  * version 2.1 of the License, or (at your option) any later version.
00014  *
00015  * FFmpeg is distributed in the hope that it will be useful,
00016  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00017  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
00018  * Lesser General Public License for more details.
00019  *
00020  * You should have received a copy of the GNU Lesser General Public
00021  * License along with FFmpeg; if not, write to the Free Software
00022  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
00023  */
00024 
00033 #include <math.h>
00034 #include <stddef.h>
00035 #include <stdio.h>
00036 
00037 #define ALT_BITSTREAM_READER_LE
00038 #include "avcodec.h"
00039 #include "get_bits.h"
00040 #include "dsputil.h"
00041 #include "fft.h"
00042 #include "mpegaudio.h"
00043 
00044 #include "qdm2data.h"
00045 #include "qdm2_tablegen.h"
00046 
00047 #undef NDEBUG
00048 #include <assert.h>
00049 
00050 
00051 #define QDM2_LIST_ADD(list, size, packet) \
00052 do { \
00053       if (size > 0) { \
00054     list[size - 1].next = &list[size]; \
00055       } \
00056       list[size].packet = packet; \
00057       list[size].next = NULL; \
00058       size++; \
00059 } while(0)
00060 
00061 // Result is 8, 16 or 30
00062 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
00063 
00064 #define FIX_NOISE_IDX(noise_idx) \
00065   if ((noise_idx) >= 3840) \
00066     (noise_idx) -= 3840; \
00067 
00068 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
00069 
00070 #define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
00071 
00072 #define SAMPLES_NEEDED \
00073      av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
00074 
00075 #define SAMPLES_NEEDED_2(why) \
00076      av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
00077 
00078 #define QDM2_MAX_FRAME_SIZE 512
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(16, 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[QDM2_MAX_FRAME_SIZE * 2];
00172 
00174     DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512*2];
00175     int synth_buf_offset[MPA_MAX_CHANNELS];
00176     DECLARE_ALIGNED(16, int32_t, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
00177 
00179     float tone_level[MPA_MAX_CHANNELS][30][64];
00180     int8_t coding_method[MPA_MAX_CHANNELS][30][64];
00181     int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
00182     int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
00183     int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
00184     int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
00185     int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
00186     int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
00187     int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
00188 
00189     // Flags
00190     int has_errors;         
00191     int superblocktype_2_3; 
00192     int do_synth_filter;    
00193 
00194     int sub_packet;
00195     int noise_idx; 
00196 } QDM2Context;
00197 
00198 
00199 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
00200 
00201 static VLC vlc_tab_level;
00202 static VLC vlc_tab_diff;
00203 static VLC vlc_tab_run;
00204 static VLC fft_level_exp_alt_vlc;
00205 static VLC fft_level_exp_vlc;
00206 static VLC fft_stereo_exp_vlc;
00207 static VLC fft_stereo_phase_vlc;
00208 static VLC vlc_tab_tone_level_idx_hi1;
00209 static VLC vlc_tab_tone_level_idx_mid;
00210 static VLC vlc_tab_tone_level_idx_hi2;
00211 static VLC vlc_tab_type30;
00212 static VLC vlc_tab_type34;
00213 static VLC vlc_tab_fft_tone_offset[5];
00214 
00215 static const uint16_t qdm2_vlc_offs[] = {
00216     0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
00217 };
00218 
00219 static av_cold void qdm2_init_vlc(void)
00220 {
00221     static int vlcs_initialized = 0;
00222     static VLC_TYPE qdm2_table[3838][2];
00223 
00224     if (!vlcs_initialized) {
00225 
00226         vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]];
00227         vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
00228         init_vlc (&vlc_tab_level, 8, 24,
00229             vlc_tab_level_huffbits, 1, 1,
00230             vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00231 
00232         vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
00233         vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
00234         init_vlc (&vlc_tab_diff, 8, 37,
00235             vlc_tab_diff_huffbits, 1, 1,
00236             vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00237 
00238         vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]];
00239         vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
00240         init_vlc (&vlc_tab_run, 5, 6,
00241             vlc_tab_run_huffbits, 1, 1,
00242             vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00243 
00244         fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
00245         fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
00246         init_vlc (&fft_level_exp_alt_vlc, 8, 28,
00247             fft_level_exp_alt_huffbits, 1, 1,
00248             fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00249 
00250 
00251         fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
00252         fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
00253         init_vlc (&fft_level_exp_vlc, 8, 20,
00254             fft_level_exp_huffbits, 1, 1,
00255             fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00256 
00257         fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
00258         fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
00259         init_vlc (&fft_stereo_exp_vlc, 6, 7,
00260             fft_stereo_exp_huffbits, 1, 1,
00261             fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00262 
00263         fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
00264         fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
00265         init_vlc (&fft_stereo_phase_vlc, 6, 9,
00266             fft_stereo_phase_huffbits, 1, 1,
00267             fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00268 
00269         vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]];
00270         vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
00271         init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
00272             vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
00273             vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00274 
00275         vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]];
00276         vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
00277         init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
00278             vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
00279             vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00280 
00281         vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]];
00282         vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
00283         init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
00284             vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
00285             vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00286 
00287         vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
00288         vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
00289         init_vlc (&vlc_tab_type30, 6, 9,
00290             vlc_tab_type30_huffbits, 1, 1,
00291             vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00292 
00293         vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
00294         vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
00295         init_vlc (&vlc_tab_type34, 5, 10,
00296             vlc_tab_type34_huffbits, 1, 1,
00297             vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00298 
00299         vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
00300         vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
00301         init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
00302             vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
00303             vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00304 
00305         vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
00306         vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
00307         init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
00308             vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
00309             vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00310 
00311         vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
00312         vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
00313         init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
00314             vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
00315             vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00316 
00317         vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
00318         vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
00319         init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
00320             vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
00321             vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00322 
00323         vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
00324         vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
00325         init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
00326             vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
00327             vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00328 
00329         vlcs_initialized=1;
00330     }
00331 }
00332 
00333 
00334 /* for floating point to fixed point conversion */
00335 static const float f2i_scale = (float) (1 << (FRAC_BITS - 15));
00336 
00337 
00338 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
00339 {
00340     int value;
00341 
00342     value = get_vlc2(gb, vlc->table, vlc->bits, depth);
00343 
00344     /* stage-2, 3 bits exponent escape sequence */
00345     if (value-- == 0)
00346         value = get_bits (gb, get_bits (gb, 3) + 1);
00347 
00348     /* stage-3, optional */
00349     if (flag) {
00350         int tmp = vlc_stage3_values[value];
00351 
00352         if ((value & ~3) > 0)
00353             tmp += get_bits (gb, (value >> 2));
00354         value = tmp;
00355     }
00356 
00357     return value;
00358 }
00359 
00360 
00361 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
00362 {
00363     int value = qdm2_get_vlc (gb, vlc, 0, depth);
00364 
00365     return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
00366 }
00367 
00368 
00378 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
00379     int i;
00380 
00381     for (i=0; i < length; i++)
00382         value -= data[i];
00383 
00384     return (uint16_t)(value & 0xffff);
00385 }
00386 
00387 
00394 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
00395 {
00396     sub_packet->type = get_bits (gb, 8);
00397 
00398     if (sub_packet->type == 0) {
00399         sub_packet->size = 0;
00400         sub_packet->data = NULL;
00401     } else {
00402         sub_packet->size = get_bits (gb, 8);
00403 
00404       if (sub_packet->type & 0x80) {
00405           sub_packet->size <<= 8;
00406           sub_packet->size  |= get_bits (gb, 8);
00407           sub_packet->type  &= 0x7f;
00408       }
00409 
00410       if (sub_packet->type == 0x7f)
00411           sub_packet->type |= (get_bits (gb, 8) << 8);
00412 
00413       sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
00414     }
00415 
00416     av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
00417         sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
00418 }
00419 
00420 
00428 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
00429 {
00430     while (list != NULL && list->packet != NULL) {
00431         if (list->packet->type == type)
00432             return list;
00433         list = list->next;
00434     }
00435     return NULL;
00436 }
00437 
00438 
00445 static void average_quantized_coeffs (QDM2Context *q)
00446 {
00447     int i, j, n, ch, sum;
00448 
00449     n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
00450 
00451     for (ch = 0; ch < q->nb_channels; ch++)
00452         for (i = 0; i < n; i++) {
00453             sum = 0;
00454 
00455             for (j = 0; j < 8; j++)
00456                 sum += q->quantized_coeffs[ch][i][j];
00457 
00458             sum /= 8;
00459             if (sum > 0)
00460                 sum--;
00461 
00462             for (j=0; j < 8; j++)
00463                 q->quantized_coeffs[ch][i][j] = sum;
00464         }
00465 }
00466 
00467 
00475 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
00476 {
00477     int ch, j;
00478 
00479     FIX_NOISE_IDX(q->noise_idx);
00480 
00481     if (!q->nb_channels)
00482         return;
00483 
00484     for (ch = 0; ch < q->nb_channels; ch++)
00485         for (j = 0; j < 64; j++) {
00486             q->sb_samples[ch][j * 2][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
00487             q->sb_samples[ch][j * 2 + 1][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
00488         }
00489 }
00490 
00491 
00500 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
00501 {
00502     int j,k;
00503     int ch;
00504     int run, case_val;
00505     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};
00506 
00507     for (ch = 0; ch < channels; ch++) {
00508         for (j = 0; j < 64; ) {
00509             if((coding_method[ch][sb][j] - 8) > 22) {
00510                 run = 1;
00511                 case_val = 8;
00512             } else {
00513                 switch (switchtable[coding_method[ch][sb][j]-8]) {
00514                     case 0: run = 10; case_val = 10; break;
00515                     case 1: run = 1; case_val = 16; break;
00516                     case 2: run = 5; case_val = 24; break;
00517                     case 3: run = 3; case_val = 30; break;
00518                     case 4: run = 1; case_val = 30; break;
00519                     case 5: run = 1; case_val = 8; break;
00520                     default: run = 1; case_val = 8; break;
00521                 }
00522             }
00523             for (k = 0; k < run; k++)
00524                 if (j + k < 128)
00525                     if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
00526                         if (k > 0) {
00527                            SAMPLES_NEEDED
00528                             //not debugged, almost never used
00529                             memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
00530                             memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
00531                         }
00532             j += run;
00533         }
00534     }
00535 }
00536 
00537 
00545 static void fill_tone_level_array (QDM2Context *q, int flag)
00546 {
00547     int i, sb, ch, sb_used;
00548     int tmp, tab;
00549 
00550     // This should never happen
00551     if (q->nb_channels <= 0)
00552         return;
00553 
00554     for (ch = 0; ch < q->nb_channels; ch++)
00555         for (sb = 0; sb < 30; sb++)
00556             for (i = 0; i < 8; i++) {
00557                 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
00558                     tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
00559                           q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
00560                 else
00561                     tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
00562                 if(tmp < 0)
00563                     tmp += 0xff;
00564                 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
00565             }
00566 
00567     sb_used = QDM2_SB_USED(q->sub_sampling);
00568 
00569     if ((q->superblocktype_2_3 != 0) && !flag) {
00570         for (sb = 0; sb < sb_used; sb++)
00571             for (ch = 0; ch < q->nb_channels; ch++)
00572                 for (i = 0; i < 64; i++) {
00573                     q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
00574                     if (q->tone_level_idx[ch][sb][i] < 0)
00575                         q->tone_level[ch][sb][i] = 0;
00576                     else
00577                         q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
00578                 }
00579     } else {
00580         tab = q->superblocktype_2_3 ? 0 : 1;
00581         for (sb = 0; sb < sb_used; sb++) {
00582             if ((sb >= 4) && (sb <= 23)) {
00583                 for (ch = 0; ch < q->nb_channels; ch++)
00584                     for (i = 0; i < 64; i++) {
00585                         tmp = q->tone_level_idx_base[ch][sb][i / 8] -
00586                               q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
00587                               q->tone_level_idx_mid[ch][sb - 4][i / 8] -
00588                               q->tone_level_idx_hi2[ch][sb - 4];
00589                         q->tone_level_idx[ch][sb][i] = tmp & 0xff;
00590                         if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
00591                             q->tone_level[ch][sb][i] = 0;
00592                         else
00593                             q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
00594                 }
00595             } else {
00596                 if (sb > 4) {
00597                     for (ch = 0; ch < q->nb_channels; ch++)
00598                         for (i = 0; i < 64; i++) {
00599                             tmp = q->tone_level_idx_base[ch][sb][i / 8] -
00600                                   q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
00601                                   q->tone_level_idx_hi2[ch][sb - 4];
00602                             q->tone_level_idx[ch][sb][i] = tmp & 0xff;
00603                             if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
00604                                 q->tone_level[ch][sb][i] = 0;
00605                             else
00606                                 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
00607                     }
00608                 } else {
00609                     for (ch = 0; ch < q->nb_channels; ch++)
00610                         for (i = 0; i < 64; i++) {
00611                             tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
00612                             if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
00613                                 q->tone_level[ch][sb][i] = 0;
00614                             else
00615                                 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
00616                         }
00617                 }
00618             }
00619         }
00620     }
00621 
00622     return;
00623 }
00624 
00625 
00640 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
00641                 sb_int8_array coding_method, int nb_channels,
00642                 int c, int superblocktype_2_3, int cm_table_select)
00643 {
00644     int ch, sb, j;
00645     int tmp, acc, esp_40, comp;
00646     int add1, add2, add3, add4;
00647     int64_t multres;
00648 
00649     // This should never happen
00650     if (nb_channels <= 0)
00651         return;
00652 
00653     if (!superblocktype_2_3) {
00654         /* This case is untested, no samples available */
00655         SAMPLES_NEEDED
00656         for (ch = 0; ch < nb_channels; ch++)
00657             for (sb = 0; sb < 30; sb++) {
00658                 for (j = 1; j < 63; j++) {  // The loop only iterates to 63 so the code doesn't overflow the buffer
00659                     add1 = tone_level_idx[ch][sb][j] - 10;
00660                     if (add1 < 0)
00661                         add1 = 0;
00662                     add2 = add3 = add4 = 0;
00663                     if (sb > 1) {
00664                         add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
00665                         if (add2 < 0)
00666                             add2 = 0;
00667                     }
00668                     if (sb > 0) {
00669                         add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
00670                         if (add3 < 0)
00671                             add3 = 0;
00672                     }
00673                     if (sb < 29) {
00674                         add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
00675                         if (add4 < 0)
00676                             add4 = 0;
00677                     }
00678                     tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
00679                     if (tmp < 0)
00680                         tmp = 0;
00681                     tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
00682                 }
00683                 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
00684             }
00685             acc = 0;
00686             for (ch = 0; ch < nb_channels; ch++)
00687                 for (sb = 0; sb < 30; sb++)
00688                     for (j = 0; j < 64; j++)
00689                         acc += tone_level_idx_temp[ch][sb][j];
00690 
00691             multres = 0x66666667 * (acc * 10);
00692             esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
00693             for (ch = 0;  ch < nb_channels; ch++)
00694                 for (sb = 0; sb < 30; sb++)
00695                     for (j = 0; j < 64; j++) {
00696                         comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
00697                         if (comp < 0)
00698                             comp += 0xff;
00699                         comp /= 256; // signed shift
00700                         switch(sb) {
00701                             case 0:
00702                                 if (comp < 30)
00703                                     comp = 30;
00704                                 comp += 15;
00705                                 break;
00706                             case 1:
00707                                 if (comp < 24)
00708                                     comp = 24;
00709                                 comp += 10;
00710                                 break;
00711                             case 2:
00712                             case 3:
00713                             case 4:
00714                                 if (comp < 16)
00715                                     comp = 16;
00716                         }
00717                         if (comp <= 5)
00718                             tmp = 0;
00719                         else if (comp <= 10)
00720                             tmp = 10;
00721                         else if (comp <= 16)
00722                             tmp = 16;
00723                         else if (comp <= 24)
00724                             tmp = -1;
00725                         else
00726                             tmp = 0;
00727                         coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
00728                     }
00729             for (sb = 0; sb < 30; sb++)
00730                 fix_coding_method_array(sb, nb_channels, coding_method);
00731             for (ch = 0; ch < nb_channels; ch++)
00732                 for (sb = 0; sb < 30; sb++)
00733                     for (j = 0; j < 64; j++)
00734                         if (sb >= 10) {
00735                             if (coding_method[ch][sb][j] < 10)
00736                                 coding_method[ch][sb][j] = 10;
00737                         } else {
00738                             if (sb >= 2) {
00739                                 if (coding_method[ch][sb][j] < 16)
00740                                     coding_method[ch][sb][j] = 16;
00741                             } else {
00742                                 if (coding_method[ch][sb][j] < 30)
00743                                     coding_method[ch][sb][j] = 30;
00744                             }
00745                         }
00746     } else { // superblocktype_2_3 != 0
00747         for (ch = 0; ch < nb_channels; ch++)
00748             for (sb = 0; sb < 30; sb++)
00749                 for (j = 0; j < 64; j++)
00750                     coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
00751     }
00752 
00753     return;
00754 }
00755 
00756 
00768 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
00769 {
00770     int sb, j, k, n, ch, run, channels;
00771     int joined_stereo, zero_encoding, chs;
00772     int type34_first;
00773     float type34_div = 0;
00774     float type34_predictor;
00775     float samples[10], sign_bits[16];
00776 
00777     if (length == 0) {
00778         // If no data use noise
00779         for (sb=sb_min; sb < sb_max; sb++)
00780             build_sb_samples_from_noise (q, sb);
00781 
00782         return;
00783     }
00784 
00785     for (sb = sb_min; sb < sb_max; sb++) {
00786         FIX_NOISE_IDX(q->noise_idx);
00787 
00788         channels = q->nb_channels;
00789 
00790         if (q->nb_channels <= 1 || sb < 12)
00791             joined_stereo = 0;
00792         else if (sb >= 24)
00793             joined_stereo = 1;
00794         else
00795             joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
00796 
00797         if (joined_stereo) {
00798             if (BITS_LEFT(length,gb) >= 16)
00799                 for (j = 0; j < 16; j++)
00800                     sign_bits[j] = get_bits1 (gb);
00801 
00802             for (j = 0; j < 64; j++)
00803                 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
00804                     q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
00805 
00806             fix_coding_method_array(sb, q->nb_channels, q->coding_method);
00807             channels = 1;
00808         }
00809 
00810         for (ch = 0; ch < channels; ch++) {
00811             zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
00812             type34_predictor = 0.0;
00813             type34_first = 1;
00814 
00815             for (j = 0; j < 128; ) {
00816                 switch (q->coding_method[ch][sb][j / 2]) {
00817                     case 8:
00818                         if (BITS_LEFT(length,gb) >= 10) {
00819                             if (zero_encoding) {
00820                                 for (k = 0; k < 5; k++) {
00821                                     if ((j + 2 * k) >= 128)
00822                                         break;
00823                                     samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
00824                                 }
00825                             } else {
00826                                 n = get_bits(gb, 8);
00827                                 for (k = 0; k < 5; k++)
00828                                     samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
00829                             }
00830                             for (k = 0; k < 5; k++)
00831                                 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
00832                         } else {
00833                             for (k = 0; k < 10; k++)
00834                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
00835                         }
00836                         run = 10;
00837                         break;
00838 
00839                     case 10:
00840                         if (BITS_LEFT(length,gb) >= 1) {
00841                             float f = 0.81;
00842 
00843                             if (get_bits1(gb))
00844                                 f = -f;
00845                             f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
00846                             samples[0] = f;
00847                         } else {
00848                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00849                         }
00850                         run = 1;
00851                         break;
00852 
00853                     case 16:
00854                         if (BITS_LEFT(length,gb) >= 10) {
00855                             if (zero_encoding) {
00856                                 for (k = 0; k < 5; k++) {
00857                                     if ((j + k) >= 128)
00858                                         break;
00859                                     samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
00860                                 }
00861                             } else {
00862                                 n = get_bits (gb, 8);
00863                                 for (k = 0; k < 5; k++)
00864                                     samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
00865                             }
00866                         } else {
00867                             for (k = 0; k < 5; k++)
00868                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
00869                         }
00870                         run = 5;
00871                         break;
00872 
00873                     case 24:
00874                         if (BITS_LEFT(length,gb) >= 7) {
00875                             n = get_bits(gb, 7);
00876                             for (k = 0; k < 3; k++)
00877                                 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
00878                         } else {
00879                             for (k = 0; k < 3; k++)
00880                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
00881                         }
00882                         run = 3;
00883                         break;
00884 
00885                     case 30:
00886                         if (BITS_LEFT(length,gb) >= 4) {
00887                             unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
00888                             if (index < FF_ARRAY_ELEMS(type30_dequant)) {
00889                                 samples[0] = type30_dequant[index];
00890                             } else
00891                                 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00892                         } else
00893                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00894 
00895                         run = 1;
00896                         break;
00897 
00898                     case 34:
00899                         if (BITS_LEFT(length,gb) >= 7) {
00900                             if (type34_first) {
00901                                 type34_div = (float)(1 << get_bits(gb, 2));
00902                                 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
00903                                 type34_predictor = samples[0];
00904                                 type34_first = 0;
00905                             } else {
00906                                 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
00907                                 if (index < FF_ARRAY_ELEMS(type34_delta)) {
00908                                     samples[0] = type34_delta[index] / type34_div + type34_predictor;
00909                                     type34_predictor = samples[0];
00910                                 } else
00911                                     samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00912                             }
00913                         } else {
00914                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00915                         }
00916                         run = 1;
00917                         break;
00918 
00919                     default:
00920                         samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00921                         run = 1;
00922                         break;
00923                 }
00924 
00925                 if (joined_stereo) {
00926                     float tmp[10][MPA_MAX_CHANNELS];
00927 
00928                     for (k = 0; k < run; k++) {
00929                         tmp[k][0] = samples[k];
00930                         tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
00931                     }
00932                     for (chs = 0; chs < q->nb_channels; chs++)
00933                         for (k = 0; k < run; k++)
00934                             if ((j + k) < 128)
00935                                 q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
00936                 } else {
00937                     for (k = 0; k < run; k++)
00938                         if ((j + k) < 128)
00939                             q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
00940                 }
00941 
00942                 j += run;
00943             } // j loop
00944         } // channel loop
00945     } // subband loop
00946 }
00947 
00948 
00959 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
00960 {
00961     int i, k, run, level, diff;
00962 
00963     if (BITS_LEFT(length,gb) < 16)
00964         return;
00965     level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
00966 
00967     quantized_coeffs[0] = level;
00968 
00969     for (i = 0; i < 7; ) {
00970         if (BITS_LEFT(length,gb) < 16)
00971             break;
00972         run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
00973 
00974         if (BITS_LEFT(length,gb) < 16)
00975             break;
00976         diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
00977 
00978         for (k = 1; k <= run; k++)
00979             quantized_coeffs[i + k] = (level + ((k * diff) / run));
00980 
00981         level += diff;
00982         i += run;
00983     }
00984 }
00985 
00986 
00996 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
00997 {
00998     int sb, j, k, n, ch;
00999 
01000     for (ch = 0; ch < q->nb_channels; ch++) {
01001         init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
01002 
01003         if (BITS_LEFT(length,gb) < 16) {
01004             memset(q->quantized_coeffs[ch][0], 0, 8);
01005             break;
01006         }
01007     }
01008 
01009     n = q->sub_sampling + 1;
01010 
01011     for (sb = 0; sb < n; sb++)
01012         for (ch = 0; ch < q->nb_channels; ch++)
01013             for (j = 0; j < 8; j++) {
01014                 if (BITS_LEFT(length,gb) < 1)
01015                     break;
01016                 if (get_bits1(gb)) {
01017                     for (k=0; k < 8; k++) {
01018                         if (BITS_LEFT(length,gb) < 16)
01019                             break;
01020                         q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
01021                     }
01022                 } else {
01023                     for (k=0; k < 8; k++)
01024                         q->tone_level_idx_hi1[ch][sb][j][k] = 0;
01025                 }
01026             }
01027 
01028     n = QDM2_SB_USED(q->sub_sampling) - 4;
01029 
01030     for (sb = 0; sb < n; sb++)
01031         for (ch = 0; ch < q->nb_channels; ch++) {
01032             if (BITS_LEFT(length,gb) < 16)
01033                 break;
01034             q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
01035             if (sb > 19)
01036                 q->tone_level_idx_hi2[ch][sb] -= 16;
01037             else
01038                 for (j = 0; j < 8; j++)
01039                     q->tone_level_idx_mid[ch][sb][j] = -16;
01040         }
01041 
01042     n = QDM2_SB_USED(q->sub_sampling) - 5;
01043 
01044     for (sb = 0; sb < n; sb++)
01045         for (ch = 0; ch < q->nb_channels; ch++)
01046             for (j = 0; j < 8; j++) {
01047                 if (BITS_LEFT(length,gb) < 16)
01048                     break;
01049                 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
01050             }
01051 }
01052 
01059 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
01060 {
01061     GetBitContext gb;
01062     int i, j, k, n, ch, run, level, diff;
01063 
01064     init_get_bits(&gb, node->packet->data, node->packet->size*8);
01065 
01066     n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
01067 
01068     for (i = 1; i < n; i++)
01069         for (ch=0; ch < q->nb_channels; ch++) {
01070             level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
01071             q->quantized_coeffs[ch][i][0] = level;
01072 
01073             for (j = 0; j < (8 - 1); ) {
01074                 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
01075                 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
01076 
01077                 for (k = 1; k <= run; k++)
01078                     q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
01079 
01080                 level += diff;
01081                 j += run;
01082             }
01083         }
01084 
01085     for (ch = 0; ch < q->nb_channels; ch++)
01086         for (i = 0; i < 8; i++)
01087             q->quantized_coeffs[ch][0][i] = 0;
01088 }
01089 
01090 
01098 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
01099 {
01100     GetBitContext gb;
01101 
01102     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
01103 
01104     if (length != 0) {
01105         init_tone_level_dequantization(q, &gb, length);
01106         fill_tone_level_array(q, 1);
01107     } else {
01108         fill_tone_level_array(q, 0);
01109     }
01110 }
01111 
01112 
01120 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
01121 {
01122     GetBitContext gb;
01123 
01124     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
01125     if (length >= 32) {
01126         int c = get_bits (&gb, 13);
01127 
01128         if (c > 3)
01129             fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
01130                                       q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
01131     }
01132 
01133     synthfilt_build_sb_samples(q, &gb, length, 0, 8);
01134 }
01135 
01136 
01144 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
01145 {
01146     GetBitContext gb;
01147 
01148     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
01149     synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
01150 }
01151 
01152 /*
01153  * Process new subpackets for synthesis filter
01154  *
01155  * @param q       context
01156  * @param list    list with synthesis filter packets (list D)
01157  */
01158 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
01159 {
01160     QDM2SubPNode *nodes[4];
01161 
01162     nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
01163     if (nodes[0] != NULL)
01164         process_subpacket_9(q, nodes[0]);
01165 
01166     nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
01167     if (nodes[1] != NULL)
01168         process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
01169     else
01170         process_subpacket_10(q, NULL, 0);
01171 
01172     nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
01173     if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
01174         process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
01175     else
01176         process_subpacket_11(q, NULL, 0);
01177 
01178     nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
01179     if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
01180         process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
01181     else
01182         process_subpacket_12(q, NULL, 0);
01183 }
01184 
01185 
01186 /*
01187  * Decode superblock, fill packet lists.
01188  *
01189  * @param q    context
01190  */
01191 static void qdm2_decode_super_block (QDM2Context *q)
01192 {
01193     GetBitContext gb;
01194     QDM2SubPacket header, *packet;
01195     int i, packet_bytes, sub_packet_size, sub_packets_D;
01196     unsigned int next_index = 0;
01197 
01198     memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
01199     memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
01200     memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
01201 
01202     q->sub_packets_B = 0;
01203     sub_packets_D = 0;
01204 
01205     average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
01206 
01207     init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
01208     qdm2_decode_sub_packet_header(&gb, &header);
01209 
01210     if (header.type < 2 || header.type >= 8) {
01211         q->has_errors = 1;
01212         av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
01213         return;
01214     }
01215 
01216     q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
01217     packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
01218 
01219     init_get_bits(&gb, header.data, header.size*8);
01220 
01221     if (header.type == 2 || header.type == 4 || header.type == 5) {
01222         int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8);
01223 
01224         csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
01225 
01226         if (csum != 0) {
01227             q->has_errors = 1;
01228             av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
01229             return;
01230         }
01231     }
01232 
01233     q->sub_packet_list_B[0].packet = NULL;
01234     q->sub_packet_list_D[0].packet = NULL;
01235 
01236     for (i = 0; i < 6; i++)
01237         if (--q->fft_level_exp[i] < 0)
01238             q->fft_level_exp[i] = 0;
01239 
01240     for (i = 0; packet_bytes > 0; i++) {
01241         int j;
01242 
01243         q->sub_packet_list_A[i].next = NULL;
01244 
01245         if (i > 0) {
01246             q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
01247 
01248             /* seek to next block */
01249             init_get_bits(&gb, header.data, header.size*8);
01250             skip_bits(&gb, next_index*8);
01251 
01252             if (next_index >= header.size)
01253                 break;
01254         }
01255 
01256         /* decode subpacket */
01257         packet = &q->sub_packets[i];
01258         qdm2_decode_sub_packet_header(&gb, packet);
01259         next_index = packet->size + get_bits_count(&gb) / 8;
01260         sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
01261 
01262         if (packet->type == 0)
01263             break;
01264 
01265         if (sub_packet_size > packet_bytes) {
01266             if (packet->type != 10 && packet->type != 11 && packet->type != 12)
01267                 break;
01268             packet->size += packet_bytes - sub_packet_size;
01269         }
01270 
01271         packet_bytes -= sub_packet_size;
01272 
01273         /* add subpacket to 'all subpackets' list */
01274         q->sub_packet_list_A[i].packet = packet;
01275 
01276         /* add subpacket to related list */
01277         if (packet->type == 8) {
01278             SAMPLES_NEEDED_2("packet type 8");
01279             return;
01280         } else if (packet->type >= 9 && packet->type <= 12) {
01281             /* packets for MPEG Audio like Synthesis Filter */
01282             QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
01283         } else if (packet->type == 13) {
01284             for (j = 0; j < 6; j++)
01285                 q->fft_level_exp[j] = get_bits(&gb, 6);
01286         } else if (packet->type == 14) {
01287             for (j = 0; j < 6; j++)
01288                 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
01289         } else if (packet->type == 15) {
01290             SAMPLES_NEEDED_2("packet type 15")
01291             return;
01292         } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
01293             /* packets for FFT */
01294             QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
01295         }
01296     } // Packet bytes loop
01297 
01298 /* **************************************************************** */
01299     if (q->sub_packet_list_D[0].packet != NULL) {
01300         process_synthesis_subpackets(q, q->sub_packet_list_D);
01301         q->do_synth_filter = 1;
01302     } else if (q->do_synth_filter) {
01303         process_subpacket_10(q, NULL, 0);
01304         process_subpacket_11(q, NULL, 0);
01305         process_subpacket_12(q, NULL, 0);
01306     }
01307 /* **************************************************************** */
01308 }
01309 
01310 
01311 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
01312                        int offset, int duration, int channel,
01313                        int exp, int phase)
01314 {
01315     if (q->fft_coefs_min_index[duration] < 0)
01316         q->fft_coefs_min_index[duration] = q->fft_coefs_index;
01317 
01318     q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
01319     q->fft_coefs[q->fft_coefs_index].channel = channel;
01320     q->fft_coefs[q->fft_coefs_index].offset = offset;
01321     q->fft_coefs[q->fft_coefs_index].exp = exp;
01322     q->fft_coefs[q->fft_coefs_index].phase = phase;
01323     q->fft_coefs_index++;
01324 }
01325 
01326 
01327 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
01328 {
01329     int channel, stereo, phase, exp;
01330     int local_int_4,  local_int_8,  stereo_phase,  local_int_10;
01331     int local_int_14, stereo_exp, local_int_20, local_int_28;
01332     int n, offset;
01333 
01334     local_int_4 = 0;
01335     local_int_28 = 0;
01336     local_int_20 = 2;
01337     local_int_8 = (4 - duration);
01338     local_int_10 = 1 << (q->group_order - duration - 1);
01339     offset = 1;
01340 
01341     while (1) {
01342         if (q->superblocktype_2_3) {
01343             while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
01344                 offset = 1;
01345                 if (n == 0) {
01346                     local_int_4 += local_int_10;
01347                     local_int_28 += (1 << local_int_8);
01348                 } else {
01349                     local_int_4 += 8*local_int_10;
01350                     local_int_28 += (8 << local_int_8);
01351                 }
01352             }
01353             offset += (n - 2);
01354         } else {
01355             offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
01356             while (offset >= (local_int_10 - 1)) {
01357                 offset += (1 - (local_int_10 - 1));
01358                 local_int_4  += local_int_10;
01359                 local_int_28 += (1 << local_int_8);
01360             }
01361         }
01362 
01363         if (local_int_4 >= q->group_size)
01364             return;
01365 
01366         local_int_14 = (offset >> local_int_8);
01367         if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
01368             return;
01369 
01370         if (q->nb_channels > 1) {
01371             channel = get_bits1(gb);
01372             stereo = get_bits1(gb);
01373         } else {
01374             channel = 0;
01375             stereo = 0;
01376         }
01377 
01378         exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
01379         exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
01380         exp = (exp < 0) ? 0 : exp;
01381 
01382         phase = get_bits(gb, 3);
01383         stereo_exp = 0;
01384         stereo_phase = 0;
01385 
01386         if (stereo) {
01387             stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
01388             stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
01389             if (stereo_phase < 0)
01390                 stereo_phase += 8;
01391         }
01392 
01393         if (q->frequency_range > (local_int_14 + 1)) {
01394             int sub_packet = (local_int_20 + local_int_28);
01395 
01396             qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
01397             if (stereo)
01398                 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
01399         }
01400 
01401         offset++;
01402     }
01403 }
01404 
01405 
01406 static void qdm2_decode_fft_packets (QDM2Context *q)
01407 {
01408     int i, j, min, max, value, type, unknown_flag;
01409     GetBitContext gb;
01410 
01411     if (q->sub_packet_list_B[0].packet == NULL)
01412         return;
01413 
01414     /* reset minimum indexes for FFT coefficients */
01415     q->fft_coefs_index = 0;
01416     for (i=0; i < 5; i++)
01417         q->fft_coefs_min_index[i] = -1;
01418 
01419     /* process subpackets ordered by type, largest type first */
01420     for (i = 0, max = 256; i < q->sub_packets_B; i++) {
01421         QDM2SubPacket *packet= NULL;
01422 
01423         /* find subpacket with largest type less than max */
01424         for (j = 0, min = 0; j < q->sub_packets_B; j++) {
01425             value = q->sub_packet_list_B[j].packet->type;
01426             if (value > min && value < max) {
01427                 min = value;
01428                 packet = q->sub_packet_list_B[j].packet;
01429             }
01430         }
01431 
01432         max = min;
01433 
01434         /* check for errors (?) */
01435         if (!packet)
01436             return;
01437 
01438         if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
01439             return;
01440 
01441         /* decode FFT tones */
01442         init_get_bits (&gb, packet->data, packet->size*8);
01443 
01444         if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
01445             unknown_flag = 1;
01446         else
01447             unknown_flag = 0;
01448 
01449         type = packet->type;
01450 
01451         if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
01452             int duration = q->sub_sampling + 5 - (type & 15);
01453 
01454             if (duration >= 0 && duration < 4)
01455                 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
01456         } else if (type == 31) {
01457             for (j=0; j < 4; j++)
01458                 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
01459         } else if (type == 46) {
01460             for (j=0; j < 6; j++)
01461                 q->fft_level_exp[j] = get_bits(&gb, 6);
01462             for (j=0; j < 4; j++)
01463             qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
01464         }
01465     } // Loop on B packets
01466 
01467     /* calculate maximum indexes for FFT coefficients */
01468     for (i = 0, j = -1; i < 5; i++)
01469         if (q->fft_coefs_min_index[i] >= 0) {
01470             if (j >= 0)
01471                 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
01472             j = i;
01473         }
01474     if (j >= 0)
01475         q->fft_coefs_max_index[j] = q->fft_coefs_index;
01476 }
01477 
01478 
01479 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
01480 {
01481    float level, f[6];
01482    int i;
01483    QDM2Complex c;
01484    const double iscale = 2.0*M_PI / 512.0;
01485 
01486     tone->phase += tone->phase_shift;
01487 
01488     /* calculate current level (maximum amplitude) of tone */
01489     level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
01490     c.im = level * sin(tone->phase*iscale);
01491     c.re = level * cos(tone->phase*iscale);
01492 
01493     /* generate FFT coefficients for tone */
01494     if (tone->duration >= 3 || tone->cutoff >= 3) {
01495         tone->complex[0].im += c.im;
01496         tone->complex[0].re += c.re;
01497         tone->complex[1].im -= c.im;
01498         tone->complex[1].re -= c.re;
01499     } else {
01500         f[1] = -tone->table[4];
01501         f[0] =  tone->table[3] - tone->table[0];
01502         f[2] =  1.0 - tone->table[2] - tone->table[3];
01503         f[3] =  tone->table[1] + tone->table[4] - 1.0;
01504         f[4] =  tone->table[0] - tone->table[1];
01505         f[5] =  tone->table[2];
01506         for (i = 0; i < 2; i++) {
01507             tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
01508             tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
01509         }
01510         for (i = 0; i < 4; i++) {
01511             tone->complex[i].re += c.re * f[i+2];
01512             tone->complex[i].im += c.im * f[i+2];
01513         }
01514     }
01515 
01516     /* copy the tone if it has not yet died out */
01517     if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
01518       memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
01519       q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
01520     }
01521 }
01522 
01523 
01524 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
01525 {
01526     int i, j, ch;
01527     const double iscale = 0.25 * M_PI;
01528 
01529     for (ch = 0; ch < q->channels; ch++) {
01530         memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
01531     }
01532 
01533 
01534     /* apply FFT tones with duration 4 (1 FFT period) */
01535     if (q->fft_coefs_min_index[4] >= 0)
01536         for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
01537             float level;
01538             QDM2Complex c;
01539 
01540             if (q->fft_coefs[i].sub_packet != sub_packet)
01541                 break;
01542 
01543             ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
01544             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];
01545 
01546             c.re = level * cos(q->fft_coefs[i].phase * iscale);
01547             c.im = level * sin(q->fft_coefs[i].phase * iscale);
01548             q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
01549             q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
01550             q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
01551             q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
01552         }
01553 
01554     /* generate existing FFT tones */
01555     for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
01556         qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
01557         q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
01558     }
01559 
01560     /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
01561     for (i = 0; i < 4; i++)
01562         if (q->fft_coefs_min_index[i] >= 0) {
01563             for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
01564                 int offset, four_i;
01565                 FFTTone tone;
01566 
01567                 if (q->fft_coefs[j].sub_packet != sub_packet)
01568                     break;
01569 
01570                 four_i = (4 - i);
01571                 offset = q->fft_coefs[j].offset >> four_i;
01572                 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
01573 
01574                 if (offset < q->frequency_range) {
01575                     if (offset < 2)
01576                         tone.cutoff = offset;
01577                     else
01578                         tone.cutoff = (offset >= 60) ? 3 : 2;
01579 
01580                     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];
01581                     tone.complex = &q->fft.complex[ch][offset];
01582                     tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
01583                     tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
01584                     tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
01585                     tone.duration = i;
01586                     tone.time_index = 0;
01587 
01588                     qdm2_fft_generate_tone(q, &tone);
01589                 }
01590             }
01591             q->fft_coefs_min_index[i] = j;
01592         }
01593 }
01594 
01595 
01596 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
01597 {
01598     const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
01599     int i;
01600     q->fft.complex[channel][0].re *= 2.0f;
01601     q->fft.complex[channel][0].im = 0.0f;
01602     ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
01603     /* add samples to output buffer */
01604     for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
01605         q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
01606 }
01607 
01608 
01613 static void qdm2_synthesis_filter (QDM2Context *q, int index)
01614 {
01615     OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
01616     int i, k, ch, sb_used, sub_sampling, dither_state = 0;
01617 
01618     /* copy sb_samples */
01619     sb_used = QDM2_SB_USED(q->sub_sampling);
01620 
01621     for (ch = 0; ch < q->channels; ch++)
01622         for (i = 0; i < 8; i++)
01623             for (k=sb_used; k < SBLIMIT; k++)
01624                 q->sb_samples[ch][(8 * index) + i][k] = 0;
01625 
01626     for (ch = 0; ch < q->nb_channels; ch++) {
01627         OUT_INT *samples_ptr = samples + ch;
01628 
01629         for (i = 0; i < 8; i++) {
01630             ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
01631                 ff_mpa_synth_window, &dither_state,
01632                 samples_ptr, q->nb_channels,
01633                 q->sb_samples[ch][(8 * index) + i]);
01634             samples_ptr += 32 * q->nb_channels;
01635         }
01636     }
01637 
01638     /* add samples to output buffer */
01639     sub_sampling = (4 >> q->sub_sampling);
01640 
01641     for (ch = 0; ch < q->channels; ch++)
01642         for (i = 0; i < q->frame_size; i++)
01643             q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
01644 }
01645 
01646 
01652 static av_cold void qdm2_init(QDM2Context *q) {
01653     static int initialized = 0;
01654 
01655     if (initialized != 0)
01656         return;
01657     initialized = 1;
01658 
01659     qdm2_init_vlc();
01660     ff_mpa_synth_init(ff_mpa_synth_window);
01661     softclip_table_init();
01662     rnd_table_init();
01663     init_noise_samples();
01664 
01665     av_log(NULL, AV_LOG_DEBUG, "init done\n");
01666 }
01667 
01668 
01669 #if 0
01670 static void dump_context(QDM2Context *q)
01671 {
01672     int i;
01673 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
01674     PRINT("compressed_data",q->compressed_data);
01675     PRINT("compressed_size",q->compressed_size);
01676     PRINT("frame_size",q->frame_size);
01677     PRINT("checksum_size",q->checksum_size);
01678     PRINT("channels",q->channels);
01679     PRINT("nb_channels",q->nb_channels);
01680     PRINT("fft_frame_size",q->fft_frame_size);
01681     PRINT("fft_size",q->fft_size);
01682     PRINT("sub_sampling",q->sub_sampling);
01683     PRINT("fft_order",q->fft_order);
01684     PRINT("group_order",q->group_order);
01685     PRINT("group_size",q->group_size);
01686     PRINT("sub_packet",q->sub_packet);
01687     PRINT("frequency_range",q->frequency_range);
01688     PRINT("has_errors",q->has_errors);
01689     PRINT("fft_tone_end",q->fft_tone_end);
01690     PRINT("fft_tone_start",q->fft_tone_start);
01691     PRINT("fft_coefs_index",q->fft_coefs_index);
01692     PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
01693     PRINT("cm_table_select",q->cm_table_select);
01694     PRINT("noise_idx",q->noise_idx);
01695 
01696     for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
01697     {
01698     FFTTone *t = &q->fft_tones[i];
01699 
01700     av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
01701     av_log(NULL,AV_LOG_DEBUG,"  level = %f\n", t->level);
01702 //  PRINT(" level", t->level);
01703     PRINT(" phase", t->phase);
01704     PRINT(" phase_shift", t->phase_shift);
01705     PRINT(" duration", t->duration);
01706     PRINT(" samples_im", t->samples_im);
01707     PRINT(" samples_re", t->samples_re);
01708     PRINT(" table", t->table);
01709     }
01710 
01711 }
01712 #endif
01713 
01714 
01718 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
01719 {
01720     QDM2Context *s = avctx->priv_data;
01721     uint8_t *extradata;
01722     int extradata_size;
01723     int tmp_val, tmp, size;
01724 
01725     /* extradata parsing
01726 
01727     Structure:
01728     wave {
01729         frma (QDM2)
01730         QDCA
01731         QDCP
01732     }
01733 
01734     32  size (including this field)
01735     32  tag (=frma)
01736     32  type (=QDM2 or QDMC)
01737 
01738     32  size (including this field, in bytes)
01739     32  tag (=QDCA) // maybe mandatory parameters
01740     32  unknown (=1)
01741     32  channels (=2)
01742     32  samplerate (=44100)
01743     32  bitrate (=96000)
01744     32  block size (=4096)
01745     32  frame size (=256) (for one channel)
01746     32  packet size (=1300)
01747 
01748     32  size (including this field, in bytes)
01749     32  tag (=QDCP) // maybe some tuneable parameters
01750     32  float1 (=1.0)
01751     32  zero ?
01752     32  float2 (=1.0)
01753     32  float3 (=1.0)
01754     32  unknown (27)
01755     32  unknown (8)
01756     32  zero ?
01757     */
01758 
01759     if (!avctx->extradata || (avctx->extradata_size < 48)) {
01760         av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
01761         return -1;
01762     }
01763 
01764     extradata = avctx->extradata;
01765     extradata_size = avctx->extradata_size;
01766 
01767     while (extradata_size > 7) {
01768         if (!memcmp(extradata, "frmaQDM", 7))
01769             break;
01770         extradata++;
01771         extradata_size--;
01772     }
01773 
01774     if (extradata_size < 12) {
01775         av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
01776                extradata_size);
01777         return -1;
01778     }
01779 
01780     if (memcmp(extradata, "frmaQDM", 7)) {
01781         av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
01782         return -1;
01783     }
01784 
01785     if (extradata[7] == 'C') {
01786 //        s->is_qdmc = 1;
01787         av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
01788         return -1;
01789     }
01790 
01791     extradata += 8;
01792     extradata_size -= 8;
01793 
01794     size = AV_RB32(extradata);
01795 
01796     if(size > extradata_size){
01797         av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
01798                extradata_size, size);
01799         return -1;
01800     }
01801 
01802     extradata += 4;
01803     av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
01804     if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
01805         av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
01806         return -1;
01807     }
01808 
01809     extradata += 8;
01810 
01811     avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
01812     extradata += 4;
01813     if (s->channels > MPA_MAX_CHANNELS)
01814         return AVERROR_INVALIDDATA;
01815 
01816     avctx->sample_rate = AV_RB32(extradata);
01817     extradata += 4;
01818 
01819     avctx->bit_rate = AV_RB32(extradata);
01820     extradata += 4;
01821 
01822     s->group_size = AV_RB32(extradata);
01823     extradata += 4;
01824 
01825     s->fft_size = AV_RB32(extradata);
01826     extradata += 4;
01827 
01828     s->checksum_size = AV_RB32(extradata);
01829 
01830     s->fft_order = av_log2(s->fft_size) + 1;
01831     s->fft_frame_size = 2 * s->fft_size; // complex has two floats
01832 
01833     // something like max decodable tones
01834     s->group_order = av_log2(s->group_size) + 1;
01835     s->frame_size = s->group_size / 16; // 16 iterations per super block
01836     if (s->frame_size > QDM2_MAX_FRAME_SIZE)
01837         return AVERROR_INVALIDDATA;
01838 
01839     s->sub_sampling = s->fft_order - 7;
01840     s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
01841 
01842     switch ((s->sub_sampling * 2 + s->channels - 1)) {
01843         case 0: tmp = 40; break;
01844         case 1: tmp = 48; break;
01845         case 2: tmp = 56; break;
01846         case 3: tmp = 72; break;
01847         case 4: tmp = 80; break;
01848         case 5: tmp = 100;break;
01849         default: tmp=s->sub_sampling; break;
01850     }
01851     tmp_val = 0;
01852     if ((tmp * 1000) < avctx->bit_rate)  tmp_val = 1;
01853     if ((tmp * 1440) < avctx->bit_rate)  tmp_val = 2;
01854     if ((tmp * 1760) < avctx->bit_rate)  tmp_val = 3;
01855     if ((tmp * 2240) < avctx->bit_rate)  tmp_val = 4;
01856     s->cm_table_select = tmp_val;
01857 
01858     if (s->sub_sampling == 0)
01859         tmp = 7999;
01860     else
01861         tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
01862     /*
01863     0: 7999 -> 0
01864     1: 20000 -> 2
01865     2: 28000 -> 2
01866     */
01867     if (tmp < 8000)
01868         s->coeff_per_sb_select = 0;
01869     else if (tmp <= 16000)
01870         s->coeff_per_sb_select = 1;
01871     else
01872         s->coeff_per_sb_select = 2;
01873 
01874     // Fail on unknown fft order
01875     if ((s->fft_order < 7) || (s->fft_order > 9)) {
01876         av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
01877         return -1;
01878     }
01879 
01880     ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
01881 
01882     qdm2_init(s);
01883 
01884     avctx->sample_fmt = SAMPLE_FMT_S16;
01885 
01886 //    dump_context(s);
01887     return 0;
01888 }
01889 
01890 
01891 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
01892 {
01893     QDM2Context *s = avctx->priv_data;
01894 
01895     ff_rdft_end(&s->rdft_ctx);
01896 
01897     return 0;
01898 }
01899 
01900 
01901 static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
01902 {
01903     int ch, i;
01904     const int frame_size = (q->frame_size * q->channels);
01905 
01906     /* select input buffer */
01907     q->compressed_data = in;
01908     q->compressed_size = q->checksum_size;
01909 
01910 //  dump_context(q);
01911 
01912     /* copy old block, clear new block of output samples */
01913     memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
01914     memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
01915 
01916     /* decode block of QDM2 compressed data */
01917     if (q->sub_packet == 0) {
01918         q->has_errors = 0; // zero it for a new super block
01919         av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
01920         qdm2_decode_super_block(q);
01921     }
01922 
01923     /* parse subpackets */
01924     if (!q->has_errors) {
01925         if (q->sub_packet == 2)
01926             qdm2_decode_fft_packets(q);
01927 
01928         qdm2_fft_tone_synthesizer(q, q->sub_packet);
01929     }
01930 
01931     /* sound synthesis stage 1 (FFT) */
01932     for (ch = 0; ch < q->channels; ch++) {
01933         qdm2_calculate_fft(q, ch, q->sub_packet);
01934 
01935         if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
01936             SAMPLES_NEEDED_2("has errors, and C list is not empty")
01937             return -1;
01938         }
01939     }
01940 
01941     /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
01942     if (!q->has_errors && q->do_synth_filter)
01943         qdm2_synthesis_filter(q, q->sub_packet);
01944 
01945     q->sub_packet = (q->sub_packet + 1) % 16;
01946 
01947     /* clip and convert output float[] to 16bit signed samples */
01948     for (i = 0; i < frame_size; i++) {
01949         int value = (int)q->output_buffer[i];
01950 
01951         if (value > SOFTCLIP_THRESHOLD)
01952             value = (value >  HARDCLIP_THRESHOLD) ?  32767 :  softclip_table[ value - SOFTCLIP_THRESHOLD];
01953         else if (value < -SOFTCLIP_THRESHOLD)
01954             value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
01955 
01956         out[i] = value;
01957     }
01958 
01959     return 0;
01960 }
01961 
01962 
01963 static int qdm2_decode_frame(AVCodecContext *avctx,
01964             void *data, int *data_size,
01965             AVPacket *avpkt)
01966 {
01967     const uint8_t *buf = avpkt->data;
01968     int buf_size = avpkt->size;
01969     QDM2Context *s = avctx->priv_data;
01970     int16_t *out = data;
01971     int i, out_size;
01972 
01973     if(!buf)
01974         return 0;
01975     if(buf_size < s->checksum_size)
01976         return -1;
01977 
01978     out_size = 16 * s->channels * s->frame_size *
01979                av_get_bits_per_sample_format(avctx->sample_fmt)/8;
01980     if (*data_size < out_size) {
01981         av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
01982         return AVERROR(EINVAL);
01983     }
01984 
01985     av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
01986        buf_size, buf, s->checksum_size, data, *data_size);
01987 
01988     for (i = 0; i < 16; i++) {
01989         if (qdm2_decode(s, buf, out) < 0)
01990             return -1;
01991         out += s->channels * s->frame_size;
01992     }
01993 
01994     *data_size = out_size;
01995 
01996     return buf_size;
01997 }
01998 
01999 AVCodec qdm2_decoder =
02000 {
02001     .name = "qdm2",
02002     .type = AVMEDIA_TYPE_AUDIO,
02003     .id = CODEC_ID_QDM2,
02004     .priv_data_size = sizeof(QDM2Context),
02005     .init = qdm2_decode_init,
02006     .close = qdm2_decode_close,
02007     .decode = qdm2_decode_frame,
02008     .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
02009 };