libavcodec/qdm2.c
Go to the documentation of this file.
00001 /*
00002  * QDM2 compatible decoder
00003  * Copyright (c) 2003 Ewald Snel
00004  * Copyright (c) 2005 Benjamin Larsson
00005  * Copyright (c) 2005 Alex Beregszaszi
00006  * Copyright (c) 2005 Roberto Togni
00007  *
00008  * This file is part of Libav.
00009  *
00010  * Libav is free software; you can redistribute it and/or
00011  * modify it under the terms of the GNU Lesser General Public
00012  * License as published by the Free Software Foundation; either
00013  * version 2.1 of the License, or (at your option) any later version.
00014  *
00015  * Libav is distributed in the hope that it will be useful,
00016  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00017  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
00018  * Lesser General Public License for more details.
00019  *
00020  * You should have received a copy of the GNU Lesser General Public
00021  * License along with Libav; if not, write to the Free Software
00022  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
00023  */
00024 
00034 #include <math.h>
00035 #include <stddef.h>
00036 #include <stdio.h>
00037 
00038 #define BITSTREAM_READER_LE
00039 #include "avcodec.h"
00040 #include "internal.h"
00041 #include "get_bits.h"
00042 #include "dsputil.h"
00043 #include "rdft.h"
00044 #include "mpegaudiodsp.h"
00045 #include "mpegaudio.h"
00046 
00047 #include "qdm2data.h"
00048 #include "qdm2_tablegen.h"
00049 
00050 #undef NDEBUG
00051 #include <assert.h>
00052 
00053 
00054 #define QDM2_LIST_ADD(list, size, packet) \
00055 do { \
00056       if (size > 0) { \
00057     list[size - 1].next = &list[size]; \
00058       } \
00059       list[size].packet = packet; \
00060       list[size].next = NULL; \
00061       size++; \
00062 } while(0)
00063 
00064 // Result is 8, 16 or 30
00065 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
00066 
00067 #define FIX_NOISE_IDX(noise_idx) \
00068   if ((noise_idx) >= 3840) \
00069     (noise_idx) -= 3840; \
00070 
00071 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
00072 
00073 #define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
00074 
00075 #define SAMPLES_NEEDED \
00076      av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
00077 
00078 #define SAMPLES_NEEDED_2(why) \
00079      av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
00080 
00081 #define QDM2_MAX_FRAME_SIZE 512
00082 
00083 typedef int8_t sb_int8_array[2][30][64];
00084 
00088 typedef struct {
00089     int type;            
00090     unsigned int size;   
00091     const uint8_t *data; 
00092 } QDM2SubPacket;
00093 
00097 typedef struct QDM2SubPNode {
00098     QDM2SubPacket *packet;      
00099     struct QDM2SubPNode *next; 
00100 } QDM2SubPNode;
00101 
00102 typedef struct {
00103     float re;
00104     float im;
00105 } QDM2Complex;
00106 
00107 typedef struct {
00108     float level;
00109     QDM2Complex *complex;
00110     const float *table;
00111     int   phase;
00112     int   phase_shift;
00113     int   duration;
00114     short time_index;
00115     short cutoff;
00116 } FFTTone;
00117 
00118 typedef struct {
00119     int16_t sub_packet;
00120     uint8_t channel;
00121     int16_t offset;
00122     int16_t exp;
00123     uint8_t phase;
00124 } FFTCoefficient;
00125 
00126 typedef struct {
00127     DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
00128 } QDM2FFT;
00129 
00133 typedef struct {
00134     AVFrame frame;
00135 
00137     int nb_channels;         
00138     int channels;            
00139     int group_size;          
00140     int fft_size;            
00141     int checksum_size;       
00142 
00144     int group_order;         
00145     int fft_order;           
00146     int fft_frame_size;      
00147     int frame_size;          
00148     int frequency_range;
00149     int sub_sampling;        
00150     int coeff_per_sb_select; 
00151     int cm_table_select;     
00152 
00154     QDM2SubPacket sub_packets[16];      
00155     QDM2SubPNode sub_packet_list_A[16]; 
00156     QDM2SubPNode sub_packet_list_B[16]; 
00157     int sub_packets_B;                  
00158     QDM2SubPNode sub_packet_list_C[16]; 
00159     QDM2SubPNode sub_packet_list_D[16]; 
00160 
00162     FFTTone fft_tones[1000];
00163     int fft_tone_start;
00164     int fft_tone_end;
00165     FFTCoefficient fft_coefs[1000];
00166     int fft_coefs_index;
00167     int fft_coefs_min_index[5];
00168     int fft_coefs_max_index[5];
00169     int fft_level_exp[6];
00170     RDFTContext rdft_ctx;
00171     QDM2FFT fft;
00172 
00174     const uint8_t *compressed_data;
00175     int compressed_size;
00176     float output_buffer[QDM2_MAX_FRAME_SIZE * 2];
00177 
00179     MPADSPContext mpadsp;
00180     DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
00181     int synth_buf_offset[MPA_MAX_CHANNELS];
00182     DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
00183     DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
00184 
00186     float tone_level[MPA_MAX_CHANNELS][30][64];
00187     int8_t coding_method[MPA_MAX_CHANNELS][30][64];
00188     int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
00189     int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
00190     int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
00191     int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
00192     int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
00193     int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
00194     int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
00195 
00196     // Flags
00197     int has_errors;         
00198     int superblocktype_2_3; 
00199     int do_synth_filter;    
00200 
00201     int sub_packet;
00202     int noise_idx; 
00203 } QDM2Context;
00204 
00205 
00206 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
00207 
00208 static VLC vlc_tab_level;
00209 static VLC vlc_tab_diff;
00210 static VLC vlc_tab_run;
00211 static VLC fft_level_exp_alt_vlc;
00212 static VLC fft_level_exp_vlc;
00213 static VLC fft_stereo_exp_vlc;
00214 static VLC fft_stereo_phase_vlc;
00215 static VLC vlc_tab_tone_level_idx_hi1;
00216 static VLC vlc_tab_tone_level_idx_mid;
00217 static VLC vlc_tab_tone_level_idx_hi2;
00218 static VLC vlc_tab_type30;
00219 static VLC vlc_tab_type34;
00220 static VLC vlc_tab_fft_tone_offset[5];
00221 
00222 static const uint16_t qdm2_vlc_offs[] = {
00223     0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
00224 };
00225 
00226 static av_cold void qdm2_init_vlc(void)
00227 {
00228     static int vlcs_initialized = 0;
00229     static VLC_TYPE qdm2_table[3838][2];
00230 
00231     if (!vlcs_initialized) {
00232 
00233         vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]];
00234         vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
00235         init_vlc (&vlc_tab_level, 8, 24,
00236             vlc_tab_level_huffbits, 1, 1,
00237             vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00238 
00239         vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
00240         vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
00241         init_vlc (&vlc_tab_diff, 8, 37,
00242             vlc_tab_diff_huffbits, 1, 1,
00243             vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00244 
00245         vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]];
00246         vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
00247         init_vlc (&vlc_tab_run, 5, 6,
00248             vlc_tab_run_huffbits, 1, 1,
00249             vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00250 
00251         fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
00252         fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
00253         init_vlc (&fft_level_exp_alt_vlc, 8, 28,
00254             fft_level_exp_alt_huffbits, 1, 1,
00255             fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00256 
00257 
00258         fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
00259         fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
00260         init_vlc (&fft_level_exp_vlc, 8, 20,
00261             fft_level_exp_huffbits, 1, 1,
00262             fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00263 
00264         fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
00265         fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
00266         init_vlc (&fft_stereo_exp_vlc, 6, 7,
00267             fft_stereo_exp_huffbits, 1, 1,
00268             fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00269 
00270         fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
00271         fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
00272         init_vlc (&fft_stereo_phase_vlc, 6, 9,
00273             fft_stereo_phase_huffbits, 1, 1,
00274             fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00275 
00276         vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]];
00277         vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
00278         init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
00279             vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
00280             vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00281 
00282         vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]];
00283         vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
00284         init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
00285             vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
00286             vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00287 
00288         vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]];
00289         vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
00290         init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
00291             vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
00292             vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00293 
00294         vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
00295         vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
00296         init_vlc (&vlc_tab_type30, 6, 9,
00297             vlc_tab_type30_huffbits, 1, 1,
00298             vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00299 
00300         vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
00301         vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
00302         init_vlc (&vlc_tab_type34, 5, 10,
00303             vlc_tab_type34_huffbits, 1, 1,
00304             vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00305 
00306         vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
00307         vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
00308         init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
00309             vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
00310             vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00311 
00312         vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
00313         vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
00314         init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
00315             vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
00316             vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00317 
00318         vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
00319         vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
00320         init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
00321             vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
00322             vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00323 
00324         vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
00325         vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
00326         init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
00327             vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
00328             vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00329 
00330         vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
00331         vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
00332         init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
00333             vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
00334             vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00335 
00336         vlcs_initialized=1;
00337     }
00338 }
00339 
00340 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
00341 {
00342     int value;
00343 
00344     value = get_vlc2(gb, vlc->table, vlc->bits, depth);
00345 
00346     /* stage-2, 3 bits exponent escape sequence */
00347     if (value-- == 0)
00348         value = get_bits (gb, get_bits (gb, 3) + 1);
00349 
00350     /* stage-3, optional */
00351     if (flag) {
00352         int tmp = vlc_stage3_values[value];
00353 
00354         if ((value & ~3) > 0)
00355             tmp += get_bits (gb, (value >> 2));
00356         value = tmp;
00357     }
00358 
00359     return value;
00360 }
00361 
00362 
00363 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
00364 {
00365     int value = qdm2_get_vlc (gb, vlc, 0, depth);
00366 
00367     return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
00368 }
00369 
00370 
00380 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
00381     int i;
00382 
00383     for (i=0; i < length; i++)
00384         value -= data[i];
00385 
00386     return (uint16_t)(value & 0xffff);
00387 }
00388 
00389 
00396 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
00397 {
00398     sub_packet->type = get_bits (gb, 8);
00399 
00400     if (sub_packet->type == 0) {
00401         sub_packet->size = 0;
00402         sub_packet->data = NULL;
00403     } else {
00404         sub_packet->size = get_bits (gb, 8);
00405 
00406       if (sub_packet->type & 0x80) {
00407           sub_packet->size <<= 8;
00408           sub_packet->size  |= get_bits (gb, 8);
00409           sub_packet->type  &= 0x7f;
00410       }
00411 
00412       if (sub_packet->type == 0x7f)
00413           sub_packet->type |= (get_bits (gb, 8) << 8);
00414 
00415       sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
00416     }
00417 
00418     av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
00419         sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
00420 }
00421 
00422 
00430 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
00431 {
00432     while (list != NULL && list->packet != NULL) {
00433         if (list->packet->type == type)
00434             return list;
00435         list = list->next;
00436     }
00437     return NULL;
00438 }
00439 
00440 
00447 static void average_quantized_coeffs (QDM2Context *q)
00448 {
00449     int i, j, n, ch, sum;
00450 
00451     n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
00452 
00453     for (ch = 0; ch < q->nb_channels; ch++)
00454         for (i = 0; i < n; i++) {
00455             sum = 0;
00456 
00457             for (j = 0; j < 8; j++)
00458                 sum += q->quantized_coeffs[ch][i][j];
00459 
00460             sum /= 8;
00461             if (sum > 0)
00462                 sum--;
00463 
00464             for (j=0; j < 8; j++)
00465                 q->quantized_coeffs[ch][i][j] = sum;
00466         }
00467 }
00468 
00469 
00477 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
00478 {
00479     int ch, j;
00480 
00481     FIX_NOISE_IDX(q->noise_idx);
00482 
00483     if (!q->nb_channels)
00484         return;
00485 
00486     for (ch = 0; ch < q->nb_channels; ch++)
00487         for (j = 0; j < 64; j++) {
00488             q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
00489             q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
00490         }
00491 }
00492 
00493 
00502 static int fix_coding_method_array(int sb, int channels,
00503                                    sb_int8_array coding_method)
00504 {
00505     int j,k;
00506     int ch;
00507     int run, case_val;
00508     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};
00509 
00510     for (ch = 0; ch < channels; ch++) {
00511         for (j = 0; j < 64; ) {
00512             if (coding_method[ch][sb][j] < 8)
00513                 return -1;
00514             if ((coding_method[ch][sb][j] - 8) > 22) {
00515                 run      = 1;
00516                 case_val = 8;
00517             } else {
00518                 switch (switchtable[coding_method[ch][sb][j]-8]) {
00519                     case 0: run = 10; case_val = 10; break;
00520                     case 1: run = 1; case_val = 16; break;
00521                     case 2: run = 5; case_val = 24; break;
00522                     case 3: run = 3; case_val = 30; break;
00523                     case 4: run = 1; case_val = 30; break;
00524                     case 5: run = 1; case_val = 8; break;
00525                     default: run = 1; case_val = 8; break;
00526                 }
00527             }
00528             for (k = 0; k < run; k++)
00529                 if (j + k < 128)
00530                     if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
00531                         if (k > 0) {
00532                            SAMPLES_NEEDED
00533                             //not debugged, almost never used
00534                             memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
00535                             memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
00536                         }
00537             j += run;
00538         }
00539     }
00540     return 0;
00541 }
00542 
00543 
00551 static void fill_tone_level_array (QDM2Context *q, int flag)
00552 {
00553     int i, sb, ch, sb_used;
00554     int tmp, tab;
00555 
00556     // This should never happen
00557     if (q->nb_channels <= 0)
00558         return;
00559 
00560     for (ch = 0; ch < q->nb_channels; ch++)
00561         for (sb = 0; sb < 30; sb++)
00562             for (i = 0; i < 8; i++) {
00563                 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
00564                     tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
00565                           q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
00566                 else
00567                     tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
00568                 if(tmp < 0)
00569                     tmp += 0xff;
00570                 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
00571             }
00572 
00573     sb_used = QDM2_SB_USED(q->sub_sampling);
00574 
00575     if ((q->superblocktype_2_3 != 0) && !flag) {
00576         for (sb = 0; sb < sb_used; sb++)
00577             for (ch = 0; ch < q->nb_channels; ch++)
00578                 for (i = 0; i < 64; i++) {
00579                     q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
00580                     if (q->tone_level_idx[ch][sb][i] < 0)
00581                         q->tone_level[ch][sb][i] = 0;
00582                     else
00583                         q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
00584                 }
00585     } else {
00586         tab = q->superblocktype_2_3 ? 0 : 1;
00587         for (sb = 0; sb < sb_used; sb++) {
00588             if ((sb >= 4) && (sb <= 23)) {
00589                 for (ch = 0; ch < q->nb_channels; ch++)
00590                     for (i = 0; i < 64; i++) {
00591                         tmp = q->tone_level_idx_base[ch][sb][i / 8] -
00592                               q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
00593                               q->tone_level_idx_mid[ch][sb - 4][i / 8] -
00594                               q->tone_level_idx_hi2[ch][sb - 4];
00595                         q->tone_level_idx[ch][sb][i] = tmp & 0xff;
00596                         if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
00597                             q->tone_level[ch][sb][i] = 0;
00598                         else
00599                             q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
00600                 }
00601             } else {
00602                 if (sb > 4) {
00603                     for (ch = 0; ch < q->nb_channels; ch++)
00604                         for (i = 0; i < 64; i++) {
00605                             tmp = q->tone_level_idx_base[ch][sb][i / 8] -
00606                                   q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
00607                                   q->tone_level_idx_hi2[ch][sb - 4];
00608                             q->tone_level_idx[ch][sb][i] = tmp & 0xff;
00609                             if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
00610                                 q->tone_level[ch][sb][i] = 0;
00611                             else
00612                                 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
00613                     }
00614                 } else {
00615                     for (ch = 0; ch < q->nb_channels; ch++)
00616                         for (i = 0; i < 64; i++) {
00617                             tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
00618                             if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
00619                                 q->tone_level[ch][sb][i] = 0;
00620                             else
00621                                 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
00622                         }
00623                 }
00624             }
00625         }
00626     }
00627 
00628     return;
00629 }
00630 
00631 
00646 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
00647                 sb_int8_array coding_method, int nb_channels,
00648                 int c, int superblocktype_2_3, int cm_table_select)
00649 {
00650     int ch, sb, j;
00651     int tmp, acc, esp_40, comp;
00652     int add1, add2, add3, add4;
00653     int64_t multres;
00654 
00655     // This should never happen
00656     if (nb_channels <= 0)
00657         return;
00658 
00659     if (!superblocktype_2_3) {
00660         /* This case is untested, no samples available */
00661         SAMPLES_NEEDED
00662         for (ch = 0; ch < nb_channels; ch++)
00663             for (sb = 0; sb < 30; sb++) {
00664                 for (j = 1; j < 63; j++) {  // The loop only iterates to 63 so the code doesn't overflow the buffer
00665                     add1 = tone_level_idx[ch][sb][j] - 10;
00666                     if (add1 < 0)
00667                         add1 = 0;
00668                     add2 = add3 = add4 = 0;
00669                     if (sb > 1) {
00670                         add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
00671                         if (add2 < 0)
00672                             add2 = 0;
00673                     }
00674                     if (sb > 0) {
00675                         add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
00676                         if (add3 < 0)
00677                             add3 = 0;
00678                     }
00679                     if (sb < 29) {
00680                         add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
00681                         if (add4 < 0)
00682                             add4 = 0;
00683                     }
00684                     tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
00685                     if (tmp < 0)
00686                         tmp = 0;
00687                     tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
00688                 }
00689                 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
00690             }
00691             acc = 0;
00692             for (ch = 0; ch < nb_channels; ch++)
00693                 for (sb = 0; sb < 30; sb++)
00694                     for (j = 0; j < 64; j++)
00695                         acc += tone_level_idx_temp[ch][sb][j];
00696 
00697             multres = 0x66666667 * (acc * 10);
00698             esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
00699             for (ch = 0;  ch < nb_channels; ch++)
00700                 for (sb = 0; sb < 30; sb++)
00701                     for (j = 0; j < 64; j++) {
00702                         comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
00703                         if (comp < 0)
00704                             comp += 0xff;
00705                         comp /= 256; // signed shift
00706                         switch(sb) {
00707                             case 0:
00708                                 if (comp < 30)
00709                                     comp = 30;
00710                                 comp += 15;
00711                                 break;
00712                             case 1:
00713                                 if (comp < 24)
00714                                     comp = 24;
00715                                 comp += 10;
00716                                 break;
00717                             case 2:
00718                             case 3:
00719                             case 4:
00720                                 if (comp < 16)
00721                                     comp = 16;
00722                         }
00723                         if (comp <= 5)
00724                             tmp = 0;
00725                         else if (comp <= 10)
00726                             tmp = 10;
00727                         else if (comp <= 16)
00728                             tmp = 16;
00729                         else if (comp <= 24)
00730                             tmp = -1;
00731                         else
00732                             tmp = 0;
00733                         coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
00734                     }
00735             for (sb = 0; sb < 30; sb++)
00736                 fix_coding_method_array(sb, nb_channels, coding_method);
00737             for (ch = 0; ch < nb_channels; ch++)
00738                 for (sb = 0; sb < 30; sb++)
00739                     for (j = 0; j < 64; j++)
00740                         if (sb >= 10) {
00741                             if (coding_method[ch][sb][j] < 10)
00742                                 coding_method[ch][sb][j] = 10;
00743                         } else {
00744                             if (sb >= 2) {
00745                                 if (coding_method[ch][sb][j] < 16)
00746                                     coding_method[ch][sb][j] = 16;
00747                             } else {
00748                                 if (coding_method[ch][sb][j] < 30)
00749                                     coding_method[ch][sb][j] = 30;
00750                             }
00751                         }
00752     } else { // superblocktype_2_3 != 0
00753         for (ch = 0; ch < nb_channels; ch++)
00754             for (sb = 0; sb < 30; sb++)
00755                 for (j = 0; j < 64; j++)
00756                     coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
00757     }
00758 
00759     return;
00760 }
00761 
00762 
00774 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
00775 {
00776     int sb, j, k, n, ch, run, channels;
00777     int joined_stereo, zero_encoding;
00778     int type34_first;
00779     float type34_div = 0;
00780     float type34_predictor;
00781     float samples[10], sign_bits[16];
00782 
00783     if (length == 0) {
00784         // If no data use noise
00785         for (sb=sb_min; sb < sb_max; sb++)
00786             build_sb_samples_from_noise (q, sb);
00787 
00788         return;
00789     }
00790 
00791     for (sb = sb_min; sb < sb_max; sb++) {
00792         channels = q->nb_channels;
00793 
00794         if (q->nb_channels <= 1 || sb < 12)
00795             joined_stereo = 0;
00796         else if (sb >= 24)
00797             joined_stereo = 1;
00798         else
00799             joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
00800 
00801         if (joined_stereo) {
00802             if (BITS_LEFT(length,gb) >= 16)
00803                 for (j = 0; j < 16; j++)
00804                     sign_bits[j] = get_bits1 (gb);
00805 
00806             for (j = 0; j < 64; j++)
00807                 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
00808                     q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
00809 
00810             if (fix_coding_method_array(sb, q->nb_channels,
00811                                             q->coding_method)) {
00812                 build_sb_samples_from_noise(q, sb);
00813                 continue;
00814             }
00815             channels = 1;
00816         }
00817 
00818         for (ch = 0; ch < channels; ch++) {
00819             FIX_NOISE_IDX(q->noise_idx);
00820             zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
00821             type34_predictor = 0.0;
00822             type34_first = 1;
00823 
00824             for (j = 0; j < 128; ) {
00825                 switch (q->coding_method[ch][sb][j / 2]) {
00826                     case 8:
00827                         if (BITS_LEFT(length,gb) >= 10) {
00828                             if (zero_encoding) {
00829                                 for (k = 0; k < 5; k++) {
00830                                     if ((j + 2 * k) >= 128)
00831                                         break;
00832                                     samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
00833                                 }
00834                             } else {
00835                                 n = get_bits(gb, 8);
00836                                 for (k = 0; k < 5; k++)
00837                                     samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
00838                             }
00839                             for (k = 0; k < 5; k++)
00840                                 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
00841                         } else {
00842                             for (k = 0; k < 10; k++)
00843                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
00844                         }
00845                         run = 10;
00846                         break;
00847 
00848                     case 10:
00849                         if (BITS_LEFT(length,gb) >= 1) {
00850                             float f = 0.81;
00851 
00852                             if (get_bits1(gb))
00853                                 f = -f;
00854                             f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
00855                             samples[0] = f;
00856                         } else {
00857                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00858                         }
00859                         run = 1;
00860                         break;
00861 
00862                     case 16:
00863                         if (BITS_LEFT(length,gb) >= 10) {
00864                             if (zero_encoding) {
00865                                 for (k = 0; k < 5; k++) {
00866                                     if ((j + k) >= 128)
00867                                         break;
00868                                     samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
00869                                 }
00870                             } else {
00871                                 n = get_bits (gb, 8);
00872                                 for (k = 0; k < 5; k++)
00873                                     samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
00874                             }
00875                         } else {
00876                             for (k = 0; k < 5; k++)
00877                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
00878                         }
00879                         run = 5;
00880                         break;
00881 
00882                     case 24:
00883                         if (BITS_LEFT(length,gb) >= 7) {
00884                             n = get_bits(gb, 7);
00885                             for (k = 0; k < 3; k++)
00886                                 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
00887                         } else {
00888                             for (k = 0; k < 3; k++)
00889                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
00890                         }
00891                         run = 3;
00892                         break;
00893 
00894                     case 30:
00895                         if (BITS_LEFT(length,gb) >= 4) {
00896                             unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
00897                             if (index < FF_ARRAY_ELEMS(type30_dequant)) {
00898                                 samples[0] = type30_dequant[index];
00899                             } else
00900                                 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00901                         } else
00902                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00903 
00904                         run = 1;
00905                         break;
00906 
00907                     case 34:
00908                         if (BITS_LEFT(length,gb) >= 7) {
00909                             if (type34_first) {
00910                                 type34_div = (float)(1 << get_bits(gb, 2));
00911                                 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
00912                                 type34_predictor = samples[0];
00913                                 type34_first = 0;
00914                             } else {
00915                                 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
00916                                 if (index < FF_ARRAY_ELEMS(type34_delta)) {
00917                                     samples[0] = type34_delta[index] / type34_div + type34_predictor;
00918                                     type34_predictor = samples[0];
00919                                 } else
00920                                     samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00921                             }
00922                         } else {
00923                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00924                         }
00925                         run = 1;
00926                         break;
00927 
00928                     default:
00929                         samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00930                         run = 1;
00931                         break;
00932                 }
00933 
00934                 if (joined_stereo) {
00935                     for (k = 0; k < run && j + k < 128; k++) {
00936                         q->sb_samples[0][j + k][sb] =
00937                             q->tone_level[0][sb][(j + k) / 2] * samples[k];
00938                         if (q->nb_channels == 2) {
00939                             if (sign_bits[(j + k) / 8])
00940                                 q->sb_samples[1][j + k][sb] =
00941                                     q->tone_level[1][sb][(j + k) / 2] * -samples[k];
00942                             else
00943                                 q->sb_samples[1][j + k][sb] =
00944                                     q->tone_level[1][sb][(j + k) / 2] * samples[k];
00945                         }
00946                     }
00947                 } else {
00948                     for (k = 0; k < run; k++)
00949                         if ((j + k) < 128)
00950                             q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
00951                 }
00952 
00953                 j += run;
00954             } // j loop
00955         } // channel loop
00956     } // subband loop
00957 }
00958 
00959 
00969 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
00970 {
00971     int i, k, run, level, diff;
00972 
00973     if (BITS_LEFT(length,gb) < 16)
00974         return;
00975     level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
00976 
00977     quantized_coeffs[0] = level;
00978 
00979     for (i = 0; i < 7; ) {
00980         if (BITS_LEFT(length,gb) < 16)
00981             break;
00982         run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
00983 
00984         if (BITS_LEFT(length,gb) < 16)
00985             break;
00986         diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
00987 
00988         for (k = 1; k <= run; k++)
00989             quantized_coeffs[i + k] = (level + ((k * diff) / run));
00990 
00991         level += diff;
00992         i += run;
00993     }
00994 }
00995 
00996 
01006 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
01007 {
01008     int sb, j, k, n, ch;
01009 
01010     for (ch = 0; ch < q->nb_channels; ch++) {
01011         init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
01012 
01013         if (BITS_LEFT(length,gb) < 16) {
01014             memset(q->quantized_coeffs[ch][0], 0, 8);
01015             break;
01016         }
01017     }
01018 
01019     n = q->sub_sampling + 1;
01020 
01021     for (sb = 0; sb < n; sb++)
01022         for (ch = 0; ch < q->nb_channels; ch++)
01023             for (j = 0; j < 8; j++) {
01024                 if (BITS_LEFT(length,gb) < 1)
01025                     break;
01026                 if (get_bits1(gb)) {
01027                     for (k=0; k < 8; k++) {
01028                         if (BITS_LEFT(length,gb) < 16)
01029                             break;
01030                         q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
01031                     }
01032                 } else {
01033                     for (k=0; k < 8; k++)
01034                         q->tone_level_idx_hi1[ch][sb][j][k] = 0;
01035                 }
01036             }
01037 
01038     n = QDM2_SB_USED(q->sub_sampling) - 4;
01039 
01040     for (sb = 0; sb < n; sb++)
01041         for (ch = 0; ch < q->nb_channels; ch++) {
01042             if (BITS_LEFT(length,gb) < 16)
01043                 break;
01044             q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
01045             if (sb > 19)
01046                 q->tone_level_idx_hi2[ch][sb] -= 16;
01047             else
01048                 for (j = 0; j < 8; j++)
01049                     q->tone_level_idx_mid[ch][sb][j] = -16;
01050         }
01051 
01052     n = QDM2_SB_USED(q->sub_sampling) - 5;
01053 
01054     for (sb = 0; sb < n; sb++)
01055         for (ch = 0; ch < q->nb_channels; ch++)
01056             for (j = 0; j < 8; j++) {
01057                 if (BITS_LEFT(length,gb) < 16)
01058                     break;
01059                 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
01060             }
01061 }
01062 
01069 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
01070 {
01071     GetBitContext gb;
01072     int i, j, k, n, ch, run, level, diff;
01073 
01074     init_get_bits(&gb, node->packet->data, node->packet->size*8);
01075 
01076     n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
01077 
01078     for (i = 1; i < n; i++)
01079         for (ch=0; ch < q->nb_channels; ch++) {
01080             level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
01081             q->quantized_coeffs[ch][i][0] = level;
01082 
01083             for (j = 0; j < (8 - 1); ) {
01084                 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
01085                 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
01086 
01087                 for (k = 1; k <= run; k++)
01088                     q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
01089 
01090                 level += diff;
01091                 j += run;
01092             }
01093         }
01094 
01095     for (ch = 0; ch < q->nb_channels; ch++)
01096         for (i = 0; i < 8; i++)
01097             q->quantized_coeffs[ch][0][i] = 0;
01098 }
01099 
01100 
01108 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
01109 {
01110     GetBitContext gb;
01111 
01112     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
01113 
01114     if (length != 0) {
01115         init_tone_level_dequantization(q, &gb, length);
01116         fill_tone_level_array(q, 1);
01117     } else {
01118         fill_tone_level_array(q, 0);
01119     }
01120 }
01121 
01122 
01130 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
01131 {
01132     GetBitContext gb;
01133 
01134     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
01135     if (length >= 32) {
01136         int c = get_bits (&gb, 13);
01137 
01138         if (c > 3)
01139             fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
01140                                       q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
01141     }
01142 
01143     synthfilt_build_sb_samples(q, &gb, length, 0, 8);
01144 }
01145 
01146 
01154 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
01155 {
01156     GetBitContext gb;
01157 
01158     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
01159     synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
01160 }
01161 
01162 /*
01163  * Process new subpackets for synthesis filter
01164  *
01165  * @param q       context
01166  * @param list    list with synthesis filter packets (list D)
01167  */
01168 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
01169 {
01170     QDM2SubPNode *nodes[4];
01171 
01172     nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
01173     if (nodes[0] != NULL)
01174         process_subpacket_9(q, nodes[0]);
01175 
01176     nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
01177     if (nodes[1] != NULL)
01178         process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
01179     else
01180         process_subpacket_10(q, NULL, 0);
01181 
01182     nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
01183     if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
01184         process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
01185     else
01186         process_subpacket_11(q, NULL, 0);
01187 
01188     nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
01189     if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
01190         process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
01191     else
01192         process_subpacket_12(q, NULL, 0);
01193 }
01194 
01195 
01196 /*
01197  * Decode superblock, fill packet lists.
01198  *
01199  * @param q    context
01200  */
01201 static void qdm2_decode_super_block (QDM2Context *q)
01202 {
01203     GetBitContext gb;
01204     QDM2SubPacket header, *packet;
01205     int i, packet_bytes, sub_packet_size, sub_packets_D;
01206     unsigned int next_index = 0;
01207 
01208     memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
01209     memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
01210     memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
01211 
01212     q->sub_packets_B = 0;
01213     sub_packets_D = 0;
01214 
01215     average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
01216 
01217     init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
01218     qdm2_decode_sub_packet_header(&gb, &header);
01219 
01220     if (header.type < 2 || header.type >= 8) {
01221         q->has_errors = 1;
01222         av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
01223         return;
01224     }
01225 
01226     q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
01227     packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
01228 
01229     init_get_bits(&gb, header.data, header.size*8);
01230 
01231     if (header.type == 2 || header.type == 4 || header.type == 5) {
01232         int csum  = 257 * get_bits(&gb, 8);
01233             csum +=   2 * get_bits(&gb, 8);
01234 
01235         csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
01236 
01237         if (csum != 0) {
01238             q->has_errors = 1;
01239             av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
01240             return;
01241         }
01242     }
01243 
01244     q->sub_packet_list_B[0].packet = NULL;
01245     q->sub_packet_list_D[0].packet = NULL;
01246 
01247     for (i = 0; i < 6; i++)
01248         if (--q->fft_level_exp[i] < 0)
01249             q->fft_level_exp[i] = 0;
01250 
01251     for (i = 0; packet_bytes > 0; i++) {
01252         int j;
01253 
01254         if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
01255             SAMPLES_NEEDED_2("too many packet bytes");
01256             return;
01257         }
01258 
01259         q->sub_packet_list_A[i].next = NULL;
01260 
01261         if (i > 0) {
01262             q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
01263 
01264             /* seek to next block */
01265             init_get_bits(&gb, header.data, header.size*8);
01266             skip_bits(&gb, next_index*8);
01267 
01268             if (next_index >= header.size)
01269                 break;
01270         }
01271 
01272         /* decode subpacket */
01273         packet = &q->sub_packets[i];
01274         qdm2_decode_sub_packet_header(&gb, packet);
01275         next_index = packet->size + get_bits_count(&gb) / 8;
01276         sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
01277 
01278         if (packet->type == 0)
01279             break;
01280 
01281         if (sub_packet_size > packet_bytes) {
01282             if (packet->type != 10 && packet->type != 11 && packet->type != 12)
01283                 break;
01284             packet->size += packet_bytes - sub_packet_size;
01285         }
01286 
01287         packet_bytes -= sub_packet_size;
01288 
01289         /* add subpacket to 'all subpackets' list */
01290         q->sub_packet_list_A[i].packet = packet;
01291 
01292         /* add subpacket to related list */
01293         if (packet->type == 8) {
01294             SAMPLES_NEEDED_2("packet type 8");
01295             return;
01296         } else if (packet->type >= 9 && packet->type <= 12) {
01297             /* packets for MPEG Audio like Synthesis Filter */
01298             QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
01299         } else if (packet->type == 13) {
01300             for (j = 0; j < 6; j++)
01301                 q->fft_level_exp[j] = get_bits(&gb, 6);
01302         } else if (packet->type == 14) {
01303             for (j = 0; j < 6; j++)
01304                 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
01305         } else if (packet->type == 15) {
01306             SAMPLES_NEEDED_2("packet type 15")
01307             return;
01308         } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
01309             /* packets for FFT */
01310             QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
01311         }
01312     } // Packet bytes loop
01313 
01314 /* **************************************************************** */
01315     if (q->sub_packet_list_D[0].packet != NULL) {
01316         process_synthesis_subpackets(q, q->sub_packet_list_D);
01317         q->do_synth_filter = 1;
01318     } else if (q->do_synth_filter) {
01319         process_subpacket_10(q, NULL, 0);
01320         process_subpacket_11(q, NULL, 0);
01321         process_subpacket_12(q, NULL, 0);
01322     }
01323 /* **************************************************************** */
01324 }
01325 
01326 
01327 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
01328                        int offset, int duration, int channel,
01329                        int exp, int phase)
01330 {
01331     if (q->fft_coefs_min_index[duration] < 0)
01332         q->fft_coefs_min_index[duration] = q->fft_coefs_index;
01333 
01334     q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
01335     q->fft_coefs[q->fft_coefs_index].channel = channel;
01336     q->fft_coefs[q->fft_coefs_index].offset = offset;
01337     q->fft_coefs[q->fft_coefs_index].exp = exp;
01338     q->fft_coefs[q->fft_coefs_index].phase = phase;
01339     q->fft_coefs_index++;
01340 }
01341 
01342 
01343 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
01344 {
01345     int channel, stereo, phase, exp;
01346     int local_int_4,  local_int_8,  stereo_phase,  local_int_10;
01347     int local_int_14, stereo_exp, local_int_20, local_int_28;
01348     int n, offset;
01349 
01350     local_int_4 = 0;
01351     local_int_28 = 0;
01352     local_int_20 = 2;
01353     local_int_8 = (4 - duration);
01354     local_int_10 = 1 << (q->group_order - duration - 1);
01355     offset = 1;
01356 
01357     while (1) {
01358         if (q->superblocktype_2_3) {
01359             while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
01360                 offset = 1;
01361                 if (n == 0) {
01362                     local_int_4 += local_int_10;
01363                     local_int_28 += (1 << local_int_8);
01364                 } else {
01365                     local_int_4 += 8*local_int_10;
01366                     local_int_28 += (8 << local_int_8);
01367                 }
01368             }
01369             offset += (n - 2);
01370         } else {
01371             offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
01372             while (offset >= (local_int_10 - 1)) {
01373                 offset += (1 - (local_int_10 - 1));
01374                 local_int_4  += local_int_10;
01375                 local_int_28 += (1 << local_int_8);
01376             }
01377         }
01378 
01379         if (local_int_4 >= q->group_size)
01380             return;
01381 
01382         local_int_14 = (offset >> local_int_8);
01383         if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
01384             return;
01385 
01386         if (q->nb_channels > 1) {
01387             channel = get_bits1(gb);
01388             stereo = get_bits1(gb);
01389         } else {
01390             channel = 0;
01391             stereo = 0;
01392         }
01393 
01394         exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
01395         exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
01396         exp = (exp < 0) ? 0 : exp;
01397 
01398         phase = get_bits(gb, 3);
01399         stereo_exp = 0;
01400         stereo_phase = 0;
01401 
01402         if (stereo) {
01403             stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
01404             stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
01405             if (stereo_phase < 0)
01406                 stereo_phase += 8;
01407         }
01408 
01409         if (q->frequency_range > (local_int_14 + 1)) {
01410             int sub_packet = (local_int_20 + local_int_28);
01411 
01412             qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
01413             if (stereo)
01414                 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
01415         }
01416 
01417         offset++;
01418     }
01419 }
01420 
01421 
01422 static void qdm2_decode_fft_packets (QDM2Context *q)
01423 {
01424     int i, j, min, max, value, type, unknown_flag;
01425     GetBitContext gb;
01426 
01427     if (q->sub_packet_list_B[0].packet == NULL)
01428         return;
01429 
01430     /* reset minimum indexes for FFT coefficients */
01431     q->fft_coefs_index = 0;
01432     for (i=0; i < 5; i++)
01433         q->fft_coefs_min_index[i] = -1;
01434 
01435     /* process subpackets ordered by type, largest type first */
01436     for (i = 0, max = 256; i < q->sub_packets_B; i++) {
01437         QDM2SubPacket *packet= NULL;
01438 
01439         /* find subpacket with largest type less than max */
01440         for (j = 0, min = 0; j < q->sub_packets_B; j++) {
01441             value = q->sub_packet_list_B[j].packet->type;
01442             if (value > min && value < max) {
01443                 min = value;
01444                 packet = q->sub_packet_list_B[j].packet;
01445             }
01446         }
01447 
01448         max = min;
01449 
01450         /* check for errors (?) */
01451         if (!packet)
01452             return;
01453 
01454         if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
01455             return;
01456 
01457         /* decode FFT tones */
01458         init_get_bits (&gb, packet->data, packet->size*8);
01459 
01460         if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
01461             unknown_flag = 1;
01462         else
01463             unknown_flag = 0;
01464 
01465         type = packet->type;
01466 
01467         if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
01468             int duration = q->sub_sampling + 5 - (type & 15);
01469 
01470             if (duration >= 0 && duration < 4)
01471                 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
01472         } else if (type == 31) {
01473             for (j=0; j < 4; j++)
01474                 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
01475         } else if (type == 46) {
01476             for (j=0; j < 6; j++)
01477                 q->fft_level_exp[j] = get_bits(&gb, 6);
01478             for (j=0; j < 4; j++)
01479             qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
01480         }
01481     } // Loop on B packets
01482 
01483     /* calculate maximum indexes for FFT coefficients */
01484     for (i = 0, j = -1; i < 5; i++)
01485         if (q->fft_coefs_min_index[i] >= 0) {
01486             if (j >= 0)
01487                 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
01488             j = i;
01489         }
01490     if (j >= 0)
01491         q->fft_coefs_max_index[j] = q->fft_coefs_index;
01492 }
01493 
01494 
01495 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
01496 {
01497    float level, f[6];
01498    int i;
01499    QDM2Complex c;
01500    const double iscale = 2.0*M_PI / 512.0;
01501 
01502     tone->phase += tone->phase_shift;
01503 
01504     /* calculate current level (maximum amplitude) of tone */
01505     level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
01506     c.im = level * sin(tone->phase*iscale);
01507     c.re = level * cos(tone->phase*iscale);
01508 
01509     /* generate FFT coefficients for tone */
01510     if (tone->duration >= 3 || tone->cutoff >= 3) {
01511         tone->complex[0].im += c.im;
01512         tone->complex[0].re += c.re;
01513         tone->complex[1].im -= c.im;
01514         tone->complex[1].re -= c.re;
01515     } else {
01516         f[1] = -tone->table[4];
01517         f[0] =  tone->table[3] - tone->table[0];
01518         f[2] =  1.0 - tone->table[2] - tone->table[3];
01519         f[3] =  tone->table[1] + tone->table[4] - 1.0;
01520         f[4] =  tone->table[0] - tone->table[1];
01521         f[5] =  tone->table[2];
01522         for (i = 0; i < 2; i++) {
01523             tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
01524             tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
01525         }
01526         for (i = 0; i < 4; i++) {
01527             tone->complex[i].re += c.re * f[i+2];
01528             tone->complex[i].im += c.im * f[i+2];
01529         }
01530     }
01531 
01532     /* copy the tone if it has not yet died out */
01533     if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
01534       memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
01535       q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
01536     }
01537 }
01538 
01539 
01540 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
01541 {
01542     int i, j, ch;
01543     const double iscale = 0.25 * M_PI;
01544 
01545     for (ch = 0; ch < q->channels; ch++) {
01546         memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
01547     }
01548 
01549 
01550     /* apply FFT tones with duration 4 (1 FFT period) */
01551     if (q->fft_coefs_min_index[4] >= 0)
01552         for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
01553             float level;
01554             QDM2Complex c;
01555 
01556             if (q->fft_coefs[i].sub_packet != sub_packet)
01557                 break;
01558 
01559             ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
01560             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];
01561 
01562             c.re = level * cos(q->fft_coefs[i].phase * iscale);
01563             c.im = level * sin(q->fft_coefs[i].phase * iscale);
01564             q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
01565             q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
01566             q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
01567             q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
01568         }
01569 
01570     /* generate existing FFT tones */
01571     for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
01572         qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
01573         q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
01574     }
01575 
01576     /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
01577     for (i = 0; i < 4; i++)
01578         if (q->fft_coefs_min_index[i] >= 0) {
01579             for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
01580                 int offset, four_i;
01581                 FFTTone tone;
01582 
01583                 if (q->fft_coefs[j].sub_packet != sub_packet)
01584                     break;
01585 
01586                 four_i = (4 - i);
01587                 offset = q->fft_coefs[j].offset >> four_i;
01588                 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
01589 
01590                 if (offset < q->frequency_range) {
01591                     if (offset < 2)
01592                         tone.cutoff = offset;
01593                     else
01594                         tone.cutoff = (offset >= 60) ? 3 : 2;
01595 
01596                     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];
01597                     tone.complex = &q->fft.complex[ch][offset];
01598                     tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
01599                     tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
01600                     tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
01601                     tone.duration = i;
01602                     tone.time_index = 0;
01603 
01604                     qdm2_fft_generate_tone(q, &tone);
01605                 }
01606             }
01607             q->fft_coefs_min_index[i] = j;
01608         }
01609 }
01610 
01611 
01612 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
01613 {
01614     const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
01615     int i;
01616     q->fft.complex[channel][0].re *= 2.0f;
01617     q->fft.complex[channel][0].im = 0.0f;
01618     q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
01619     /* add samples to output buffer */
01620     for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
01621         q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
01622 }
01623 
01624 
01629 static void qdm2_synthesis_filter (QDM2Context *q, int index)
01630 {
01631     int i, k, ch, sb_used, sub_sampling, dither_state = 0;
01632 
01633     /* copy sb_samples */
01634     sb_used = QDM2_SB_USED(q->sub_sampling);
01635 
01636     for (ch = 0; ch < q->channels; ch++)
01637         for (i = 0; i < 8; i++)
01638             for (k=sb_used; k < SBLIMIT; k++)
01639                 q->sb_samples[ch][(8 * index) + i][k] = 0;
01640 
01641     for (ch = 0; ch < q->nb_channels; ch++) {
01642         float *samples_ptr = q->samples + ch;
01643 
01644         for (i = 0; i < 8; i++) {
01645             ff_mpa_synth_filter_float(&q->mpadsp,
01646                 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
01647                 ff_mpa_synth_window_float, &dither_state,
01648                 samples_ptr, q->nb_channels,
01649                 q->sb_samples[ch][(8 * index) + i]);
01650             samples_ptr += 32 * q->nb_channels;
01651         }
01652     }
01653 
01654     /* add samples to output buffer */
01655     sub_sampling = (4 >> q->sub_sampling);
01656 
01657     for (ch = 0; ch < q->channels; ch++)
01658         for (i = 0; i < q->frame_size; i++)
01659             q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
01660 }
01661 
01662 
01668 static av_cold void qdm2_init(QDM2Context *q) {
01669     static int initialized = 0;
01670 
01671     if (initialized != 0)
01672         return;
01673     initialized = 1;
01674 
01675     qdm2_init_vlc();
01676     ff_mpa_synth_init_float(ff_mpa_synth_window_float);
01677     softclip_table_init();
01678     rnd_table_init();
01679     init_noise_samples();
01680 
01681     av_log(NULL, AV_LOG_DEBUG, "init done\n");
01682 }
01683 
01684 
01685 #if 0
01686 static void dump_context(QDM2Context *q)
01687 {
01688     int i;
01689 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
01690     PRINT("compressed_data",q->compressed_data);
01691     PRINT("compressed_size",q->compressed_size);
01692     PRINT("frame_size",q->frame_size);
01693     PRINT("checksum_size",q->checksum_size);
01694     PRINT("channels",q->channels);
01695     PRINT("nb_channels",q->nb_channels);
01696     PRINT("fft_frame_size",q->fft_frame_size);
01697     PRINT("fft_size",q->fft_size);
01698     PRINT("sub_sampling",q->sub_sampling);
01699     PRINT("fft_order",q->fft_order);
01700     PRINT("group_order",q->group_order);
01701     PRINT("group_size",q->group_size);
01702     PRINT("sub_packet",q->sub_packet);
01703     PRINT("frequency_range",q->frequency_range);
01704     PRINT("has_errors",q->has_errors);
01705     PRINT("fft_tone_end",q->fft_tone_end);
01706     PRINT("fft_tone_start",q->fft_tone_start);
01707     PRINT("fft_coefs_index",q->fft_coefs_index);
01708     PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
01709     PRINT("cm_table_select",q->cm_table_select);
01710     PRINT("noise_idx",q->noise_idx);
01711 
01712     for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
01713     {
01714     FFTTone *t = &q->fft_tones[i];
01715 
01716     av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
01717     av_log(NULL,AV_LOG_DEBUG,"  level = %f\n", t->level);
01718 //  PRINT(" level", t->level);
01719     PRINT(" phase", t->phase);
01720     PRINT(" phase_shift", t->phase_shift);
01721     PRINT(" duration", t->duration);
01722     PRINT(" samples_im", t->samples_im);
01723     PRINT(" samples_re", t->samples_re);
01724     PRINT(" table", t->table);
01725     }
01726 
01727 }
01728 #endif
01729 
01730 
01734 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
01735 {
01736     QDM2Context *s = avctx->priv_data;
01737     uint8_t *extradata;
01738     int extradata_size;
01739     int tmp_val, tmp, size;
01740 
01741     /* extradata parsing
01742 
01743     Structure:
01744     wave {
01745         frma (QDM2)
01746         QDCA
01747         QDCP
01748     }
01749 
01750     32  size (including this field)
01751     32  tag (=frma)
01752     32  type (=QDM2 or QDMC)
01753 
01754     32  size (including this field, in bytes)
01755     32  tag (=QDCA) // maybe mandatory parameters
01756     32  unknown (=1)
01757     32  channels (=2)
01758     32  samplerate (=44100)
01759     32  bitrate (=96000)
01760     32  block size (=4096)
01761     32  frame size (=256) (for one channel)
01762     32  packet size (=1300)
01763 
01764     32  size (including this field, in bytes)
01765     32  tag (=QDCP) // maybe some tuneable parameters
01766     32  float1 (=1.0)
01767     32  zero ?
01768     32  float2 (=1.0)
01769     32  float3 (=1.0)
01770     32  unknown (27)
01771     32  unknown (8)
01772     32  zero ?
01773     */
01774 
01775     if (!avctx->extradata || (avctx->extradata_size < 48)) {
01776         av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
01777         return -1;
01778     }
01779 
01780     extradata = avctx->extradata;
01781     extradata_size = avctx->extradata_size;
01782 
01783     while (extradata_size > 7) {
01784         if (!memcmp(extradata, "frmaQDM", 7))
01785             break;
01786         extradata++;
01787         extradata_size--;
01788     }
01789 
01790     if (extradata_size < 12) {
01791         av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
01792                extradata_size);
01793         return -1;
01794     }
01795 
01796     if (memcmp(extradata, "frmaQDM", 7)) {
01797         av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
01798         return -1;
01799     }
01800 
01801     if (extradata[7] == 'C') {
01802 //        s->is_qdmc = 1;
01803         av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
01804         return -1;
01805     }
01806 
01807     extradata += 8;
01808     extradata_size -= 8;
01809 
01810     size = AV_RB32(extradata);
01811 
01812     if(size > extradata_size){
01813         av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
01814                extradata_size, size);
01815         return -1;
01816     }
01817 
01818     extradata += 4;
01819     av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
01820     if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
01821         av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
01822         return -1;
01823     }
01824 
01825     extradata += 8;
01826 
01827     avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
01828     extradata += 4;
01829     if (s->channels > MPA_MAX_CHANNELS)
01830         return AVERROR_INVALIDDATA;
01831 
01832     avctx->sample_rate = AV_RB32(extradata);
01833     extradata += 4;
01834 
01835     avctx->bit_rate = AV_RB32(extradata);
01836     extradata += 4;
01837 
01838     s->group_size = AV_RB32(extradata);
01839     extradata += 4;
01840 
01841     s->fft_size = AV_RB32(extradata);
01842     extradata += 4;
01843 
01844     s->checksum_size = AV_RB32(extradata);
01845     if (s->checksum_size >= 1U << 28) {
01846         av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
01847         return AVERROR_INVALIDDATA;
01848     }
01849 
01850     s->fft_order = av_log2(s->fft_size) + 1;
01851     s->fft_frame_size = 2 * s->fft_size; // complex has two floats
01852 
01853     // something like max decodable tones
01854     s->group_order = av_log2(s->group_size) + 1;
01855     s->frame_size = s->group_size / 16; // 16 iterations per super block
01856     if (s->frame_size > QDM2_MAX_FRAME_SIZE)
01857         return AVERROR_INVALIDDATA;
01858 
01859     s->sub_sampling = s->fft_order - 7;
01860     s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
01861 
01862     switch ((s->sub_sampling * 2 + s->channels - 1)) {
01863         case 0: tmp = 40; break;
01864         case 1: tmp = 48; break;
01865         case 2: tmp = 56; break;
01866         case 3: tmp = 72; break;
01867         case 4: tmp = 80; break;
01868         case 5: tmp = 100;break;
01869         default: tmp=s->sub_sampling; break;
01870     }
01871     tmp_val = 0;
01872     if ((tmp * 1000) < avctx->bit_rate)  tmp_val = 1;
01873     if ((tmp * 1440) < avctx->bit_rate)  tmp_val = 2;
01874     if ((tmp * 1760) < avctx->bit_rate)  tmp_val = 3;
01875     if ((tmp * 2240) < avctx->bit_rate)  tmp_val = 4;
01876     s->cm_table_select = tmp_val;
01877 
01878     if (s->sub_sampling == 0)
01879         tmp = 7999;
01880     else
01881         tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
01882     /*
01883     0: 7999 -> 0
01884     1: 20000 -> 2
01885     2: 28000 -> 2
01886     */
01887     if (tmp < 8000)
01888         s->coeff_per_sb_select = 0;
01889     else if (tmp <= 16000)
01890         s->coeff_per_sb_select = 1;
01891     else
01892         s->coeff_per_sb_select = 2;
01893 
01894     // Fail on unknown fft order
01895     if ((s->fft_order < 7) || (s->fft_order > 9)) {
01896         av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
01897         return -1;
01898     }
01899     if (s->fft_size != (1 << (s->fft_order - 1))) {
01900         av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
01901         return AVERROR_INVALIDDATA;
01902     }
01903 
01904     ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
01905     ff_mpadsp_init(&s->mpadsp);
01906 
01907     qdm2_init(s);
01908 
01909     avctx->sample_fmt = AV_SAMPLE_FMT_S16;
01910 
01911     avcodec_get_frame_defaults(&s->frame);
01912     avctx->coded_frame = &s->frame;
01913 
01914 //    dump_context(s);
01915     return 0;
01916 }
01917 
01918 
01919 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
01920 {
01921     QDM2Context *s = avctx->priv_data;
01922 
01923     ff_rdft_end(&s->rdft_ctx);
01924 
01925     return 0;
01926 }
01927 
01928 
01929 static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
01930 {
01931     int ch, i;
01932     const int frame_size = (q->frame_size * q->channels);
01933 
01934     /* select input buffer */
01935     q->compressed_data = in;
01936     q->compressed_size = q->checksum_size;
01937 
01938 //  dump_context(q);
01939 
01940     /* copy old block, clear new block of output samples */
01941     memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
01942     memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
01943 
01944     /* decode block of QDM2 compressed data */
01945     if (q->sub_packet == 0) {
01946         q->has_errors = 0; // zero it for a new super block
01947         av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
01948         qdm2_decode_super_block(q);
01949     }
01950 
01951     /* parse subpackets */
01952     if (!q->has_errors) {
01953         if (q->sub_packet == 2)
01954             qdm2_decode_fft_packets(q);
01955 
01956         qdm2_fft_tone_synthesizer(q, q->sub_packet);
01957     }
01958 
01959     /* sound synthesis stage 1 (FFT) */
01960     for (ch = 0; ch < q->channels; ch++) {
01961         qdm2_calculate_fft(q, ch, q->sub_packet);
01962 
01963         if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
01964             SAMPLES_NEEDED_2("has errors, and C list is not empty")
01965             return -1;
01966         }
01967     }
01968 
01969     /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
01970     if (!q->has_errors && q->do_synth_filter)
01971         qdm2_synthesis_filter(q, q->sub_packet);
01972 
01973     q->sub_packet = (q->sub_packet + 1) % 16;
01974 
01975     /* clip and convert output float[] to 16bit signed samples */
01976     for (i = 0; i < frame_size; i++) {
01977         int value = (int)q->output_buffer[i];
01978 
01979         if (value > SOFTCLIP_THRESHOLD)
01980             value = (value >  HARDCLIP_THRESHOLD) ?  32767 :  softclip_table[ value - SOFTCLIP_THRESHOLD];
01981         else if (value < -SOFTCLIP_THRESHOLD)
01982             value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
01983 
01984         out[i] = value;
01985     }
01986 
01987     return 0;
01988 }
01989 
01990 
01991 static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
01992                              int *got_frame_ptr, AVPacket *avpkt)
01993 {
01994     const uint8_t *buf = avpkt->data;
01995     int buf_size = avpkt->size;
01996     QDM2Context *s = avctx->priv_data;
01997     int16_t *out;
01998     int i, ret;
01999 
02000     if(!buf)
02001         return 0;
02002     if(buf_size < s->checksum_size)
02003         return -1;
02004 
02005     /* get output buffer */
02006     s->frame.nb_samples = 16 * s->frame_size;
02007     if ((ret = ff_get_buffer(avctx, &s->frame)) < 0) {
02008         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
02009         return ret;
02010     }
02011     out = (int16_t *)s->frame.data[0];
02012 
02013     for (i = 0; i < 16; i++) {
02014         if (qdm2_decode(s, buf, out) < 0)
02015             return -1;
02016         out += s->channels * s->frame_size;
02017     }
02018 
02019     *got_frame_ptr   = 1;
02020     *(AVFrame *)data = s->frame;
02021 
02022     return s->checksum_size;
02023 }
02024 
02025 AVCodec ff_qdm2_decoder =
02026 {
02027     .name = "qdm2",
02028     .type = AVMEDIA_TYPE_AUDIO,
02029     .id = CODEC_ID_QDM2,
02030     .priv_data_size = sizeof(QDM2Context),
02031     .init = qdm2_decode_init,
02032     .close = qdm2_decode_close,
02033     .decode = qdm2_decode_frame,
02034     .capabilities = CODEC_CAP_DR1,
02035     .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
02036 };