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libavcodec/qdm2.c

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00001 /*
00002  * QDM2 compatible decoder
00003  * Copyright (c) 2003 Ewald Snel
00004  * Copyright (c) 2005 Benjamin Larsson
00005  * Copyright (c) 2005 Alex Beregszaszi
00006  * Copyright (c) 2005 Roberto Togni
00007  *
00008  * This file is part of Libav.
00009  *
00010  * Libav is free software; you can redistribute it and/or
00011  * modify it under the terms of the GNU Lesser General Public
00012  * License as published by the Free Software Foundation; either
00013  * version 2.1 of the License, or (at your option) any later version.
00014  *
00015  * Libav is distributed in the hope that it will be useful,
00016  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00017  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
00018  * Lesser General Public License for more details.
00019  *
00020  * You should have received a copy of the GNU Lesser General Public
00021  * License along with Libav; if not, write to the Free Software
00022  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
00023  */
00024 
00034 #include <math.h>
00035 #include <stddef.h>
00036 #include <stdio.h>
00037 
00038 #define BITSTREAM_READER_LE
00039 #include "avcodec.h"
00040 #include "get_bits.h"
00041 #include "dsputil.h"
00042 #include "rdft.h"
00043 #include "mpegaudiodsp.h"
00044 #include "mpegaudio.h"
00045 
00046 #include "qdm2data.h"
00047 #include "qdm2_tablegen.h"
00048 
00049 #undef NDEBUG
00050 #include <assert.h>
00051 
00052 
00053 #define QDM2_LIST_ADD(list, size, packet) \
00054 do { \
00055       if (size > 0) { \
00056     list[size - 1].next = &list[size]; \
00057       } \
00058       list[size].packet = packet; \
00059       list[size].next = NULL; \
00060       size++; \
00061 } while(0)
00062 
00063 // Result is 8, 16 or 30
00064 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
00065 
00066 #define FIX_NOISE_IDX(noise_idx) \
00067   if ((noise_idx) >= 3840) \
00068     (noise_idx) -= 3840; \
00069 
00070 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
00071 
00072 #define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
00073 
00074 #define SAMPLES_NEEDED \
00075      av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
00076 
00077 #define SAMPLES_NEEDED_2(why) \
00078      av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
00079 
00080 #define QDM2_MAX_FRAME_SIZE 512
00081 
00082 typedef int8_t sb_int8_array[2][30][64];
00083 
00087 typedef struct {
00088     int type;            
00089     unsigned int size;   
00090     const uint8_t *data; 
00091 } QDM2SubPacket;
00092 
00096 typedef struct QDM2SubPNode {
00097     QDM2SubPacket *packet;      
00098     struct QDM2SubPNode *next; 
00099 } QDM2SubPNode;
00100 
00101 typedef struct {
00102     float re;
00103     float im;
00104 } QDM2Complex;
00105 
00106 typedef struct {
00107     float level;
00108     QDM2Complex *complex;
00109     const float *table;
00110     int   phase;
00111     int   phase_shift;
00112     int   duration;
00113     short time_index;
00114     short cutoff;
00115 } FFTTone;
00116 
00117 typedef struct {
00118     int16_t sub_packet;
00119     uint8_t channel;
00120     int16_t offset;
00121     int16_t exp;
00122     uint8_t phase;
00123 } FFTCoefficient;
00124 
00125 typedef struct {
00126     DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
00127 } QDM2FFT;
00128 
00132 typedef struct {
00133     AVFrame frame;
00134 
00136     int nb_channels;         
00137     int channels;            
00138     int group_size;          
00139     int fft_size;            
00140     int checksum_size;       
00141 
00143     int group_order;         
00144     int fft_order;           
00145     int fft_frame_size;      
00146     int frame_size;          
00147     int frequency_range;
00148     int sub_sampling;        
00149     int coeff_per_sb_select; 
00150     int cm_table_select;     
00151 
00153     QDM2SubPacket sub_packets[16];      
00154     QDM2SubPNode sub_packet_list_A[16]; 
00155     QDM2SubPNode sub_packet_list_B[16]; 
00156     int sub_packets_B;                  
00157     QDM2SubPNode sub_packet_list_C[16]; 
00158     QDM2SubPNode sub_packet_list_D[16]; 
00159 
00161     FFTTone fft_tones[1000];
00162     int fft_tone_start;
00163     int fft_tone_end;
00164     FFTCoefficient fft_coefs[1000];
00165     int fft_coefs_index;
00166     int fft_coefs_min_index[5];
00167     int fft_coefs_max_index[5];
00168     int fft_level_exp[6];
00169     RDFTContext rdft_ctx;
00170     QDM2FFT fft;
00171 
00173     const uint8_t *compressed_data;
00174     int compressed_size;
00175     float output_buffer[QDM2_MAX_FRAME_SIZE * 2];
00176 
00178     MPADSPContext mpadsp;
00179     DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
00180     int synth_buf_offset[MPA_MAX_CHANNELS];
00181     DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
00182     DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
00183 
00185     float tone_level[MPA_MAX_CHANNELS][30][64];
00186     int8_t coding_method[MPA_MAX_CHANNELS][30][64];
00187     int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
00188     int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
00189     int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
00190     int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
00191     int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
00192     int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
00193     int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
00194 
00195     // Flags
00196     int has_errors;         
00197     int superblocktype_2_3; 
00198     int do_synth_filter;    
00199 
00200     int sub_packet;
00201     int noise_idx; 
00202 } QDM2Context;
00203 
00204 
00205 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
00206 
00207 static VLC vlc_tab_level;
00208 static VLC vlc_tab_diff;
00209 static VLC vlc_tab_run;
00210 static VLC fft_level_exp_alt_vlc;
00211 static VLC fft_level_exp_vlc;
00212 static VLC fft_stereo_exp_vlc;
00213 static VLC fft_stereo_phase_vlc;
00214 static VLC vlc_tab_tone_level_idx_hi1;
00215 static VLC vlc_tab_tone_level_idx_mid;
00216 static VLC vlc_tab_tone_level_idx_hi2;
00217 static VLC vlc_tab_type30;
00218 static VLC vlc_tab_type34;
00219 static VLC vlc_tab_fft_tone_offset[5];
00220 
00221 static const uint16_t qdm2_vlc_offs[] = {
00222     0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
00223 };
00224 
00225 static av_cold void qdm2_init_vlc(void)
00226 {
00227     static int vlcs_initialized = 0;
00228     static VLC_TYPE qdm2_table[3838][2];
00229 
00230     if (!vlcs_initialized) {
00231 
00232         vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]];
00233         vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
00234         init_vlc (&vlc_tab_level, 8, 24,
00235             vlc_tab_level_huffbits, 1, 1,
00236             vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00237 
00238         vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
00239         vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
00240         init_vlc (&vlc_tab_diff, 8, 37,
00241             vlc_tab_diff_huffbits, 1, 1,
00242             vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00243 
00244         vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]];
00245         vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
00246         init_vlc (&vlc_tab_run, 5, 6,
00247             vlc_tab_run_huffbits, 1, 1,
00248             vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00249 
00250         fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
00251         fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
00252         init_vlc (&fft_level_exp_alt_vlc, 8, 28,
00253             fft_level_exp_alt_huffbits, 1, 1,
00254             fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00255 
00256 
00257         fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
00258         fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
00259         init_vlc (&fft_level_exp_vlc, 8, 20,
00260             fft_level_exp_huffbits, 1, 1,
00261             fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00262 
00263         fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
00264         fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
00265         init_vlc (&fft_stereo_exp_vlc, 6, 7,
00266             fft_stereo_exp_huffbits, 1, 1,
00267             fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00268 
00269         fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
00270         fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
00271         init_vlc (&fft_stereo_phase_vlc, 6, 9,
00272             fft_stereo_phase_huffbits, 1, 1,
00273             fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00274 
00275         vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]];
00276         vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
00277         init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
00278             vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
00279             vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00280 
00281         vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]];
00282         vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
00283         init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
00284             vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
00285             vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00286 
00287         vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]];
00288         vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
00289         init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
00290             vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
00291             vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00292 
00293         vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
00294         vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
00295         init_vlc (&vlc_tab_type30, 6, 9,
00296             vlc_tab_type30_huffbits, 1, 1,
00297             vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00298 
00299         vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
00300         vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
00301         init_vlc (&vlc_tab_type34, 5, 10,
00302             vlc_tab_type34_huffbits, 1, 1,
00303             vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00304 
00305         vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
00306         vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
00307         init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
00308             vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
00309             vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00310 
00311         vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
00312         vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
00313         init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
00314             vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
00315             vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00316 
00317         vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
00318         vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
00319         init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
00320             vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
00321             vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00322 
00323         vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
00324         vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
00325         init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
00326             vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
00327             vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00328 
00329         vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
00330         vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
00331         init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
00332             vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
00333             vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00334 
00335         vlcs_initialized=1;
00336     }
00337 }
00338 
00339 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
00340 {
00341     int value;
00342 
00343     value = get_vlc2(gb, vlc->table, vlc->bits, depth);
00344 
00345     /* stage-2, 3 bits exponent escape sequence */
00346     if (value-- == 0)
00347         value = get_bits (gb, get_bits (gb, 3) + 1);
00348 
00349     /* stage-3, optional */
00350     if (flag) {
00351         int tmp = vlc_stage3_values[value];
00352 
00353         if ((value & ~3) > 0)
00354             tmp += get_bits (gb, (value >> 2));
00355         value = tmp;
00356     }
00357 
00358     return value;
00359 }
00360 
00361 
00362 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
00363 {
00364     int value = qdm2_get_vlc (gb, vlc, 0, depth);
00365 
00366     return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
00367 }
00368 
00369 
00379 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
00380     int i;
00381 
00382     for (i=0; i < length; i++)
00383         value -= data[i];
00384 
00385     return (uint16_t)(value & 0xffff);
00386 }
00387 
00388 
00395 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
00396 {
00397     sub_packet->type = get_bits (gb, 8);
00398 
00399     if (sub_packet->type == 0) {
00400         sub_packet->size = 0;
00401         sub_packet->data = NULL;
00402     } else {
00403         sub_packet->size = get_bits (gb, 8);
00404 
00405       if (sub_packet->type & 0x80) {
00406           sub_packet->size <<= 8;
00407           sub_packet->size  |= get_bits (gb, 8);
00408           sub_packet->type  &= 0x7f;
00409       }
00410 
00411       if (sub_packet->type == 0x7f)
00412           sub_packet->type |= (get_bits (gb, 8) << 8);
00413 
00414       sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
00415     }
00416 
00417     av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
00418         sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
00419 }
00420 
00421 
00429 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
00430 {
00431     while (list != NULL && list->packet != NULL) {
00432         if (list->packet->type == type)
00433             return list;
00434         list = list->next;
00435     }
00436     return NULL;
00437 }
00438 
00439 
00446 static void average_quantized_coeffs (QDM2Context *q)
00447 {
00448     int i, j, n, ch, sum;
00449 
00450     n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
00451 
00452     for (ch = 0; ch < q->nb_channels; ch++)
00453         for (i = 0; i < n; i++) {
00454             sum = 0;
00455 
00456             for (j = 0; j < 8; j++)
00457                 sum += q->quantized_coeffs[ch][i][j];
00458 
00459             sum /= 8;
00460             if (sum > 0)
00461                 sum--;
00462 
00463             for (j=0; j < 8; j++)
00464                 q->quantized_coeffs[ch][i][j] = sum;
00465         }
00466 }
00467 
00468 
00476 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
00477 {
00478     int ch, j;
00479 
00480     FIX_NOISE_IDX(q->noise_idx);
00481 
00482     if (!q->nb_channels)
00483         return;
00484 
00485     for (ch = 0; ch < q->nb_channels; ch++)
00486         for (j = 0; j < 64; j++) {
00487             q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
00488             q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
00489         }
00490 }
00491 
00492 
00501 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
00502 {
00503     int j,k;
00504     int ch;
00505     int run, case_val;
00506     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};
00507 
00508     for (ch = 0; ch < channels; ch++) {
00509         for (j = 0; j < 64; ) {
00510             if((coding_method[ch][sb][j] - 8) > 22) {
00511                 run = 1;
00512                 case_val = 8;
00513             } else {
00514                 switch (switchtable[coding_method[ch][sb][j]-8]) {
00515                     case 0: run = 10; case_val = 10; break;
00516                     case 1: run = 1; case_val = 16; break;
00517                     case 2: run = 5; case_val = 24; break;
00518                     case 3: run = 3; case_val = 30; break;
00519                     case 4: run = 1; case_val = 30; break;
00520                     case 5: run = 1; case_val = 8; break;
00521                     default: run = 1; case_val = 8; break;
00522                 }
00523             }
00524             for (k = 0; k < run; k++)
00525                 if (j + k < 128)
00526                     if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
00527                         if (k > 0) {
00528                            SAMPLES_NEEDED
00529                             //not debugged, almost never used
00530                             memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
00531                             memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
00532                         }
00533             j += run;
00534         }
00535     }
00536 }
00537 
00538 
00546 static void fill_tone_level_array (QDM2Context *q, int flag)
00547 {
00548     int i, sb, ch, sb_used;
00549     int tmp, tab;
00550 
00551     // This should never happen
00552     if (q->nb_channels <= 0)
00553         return;
00554 
00555     for (ch = 0; ch < q->nb_channels; ch++)
00556         for (sb = 0; sb < 30; sb++)
00557             for (i = 0; i < 8; i++) {
00558                 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
00559                     tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
00560                           q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
00561                 else
00562                     tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
00563                 if(tmp < 0)
00564                     tmp += 0xff;
00565                 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
00566             }
00567 
00568     sb_used = QDM2_SB_USED(q->sub_sampling);
00569 
00570     if ((q->superblocktype_2_3 != 0) && !flag) {
00571         for (sb = 0; sb < sb_used; sb++)
00572             for (ch = 0; ch < q->nb_channels; ch++)
00573                 for (i = 0; i < 64; i++) {
00574                     q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
00575                     if (q->tone_level_idx[ch][sb][i] < 0)
00576                         q->tone_level[ch][sb][i] = 0;
00577                     else
00578                         q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
00579                 }
00580     } else {
00581         tab = q->superblocktype_2_3 ? 0 : 1;
00582         for (sb = 0; sb < sb_used; sb++) {
00583             if ((sb >= 4) && (sb <= 23)) {
00584                 for (ch = 0; ch < q->nb_channels; ch++)
00585                     for (i = 0; i < 64; i++) {
00586                         tmp = q->tone_level_idx_base[ch][sb][i / 8] -
00587                               q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
00588                               q->tone_level_idx_mid[ch][sb - 4][i / 8] -
00589                               q->tone_level_idx_hi2[ch][sb - 4];
00590                         q->tone_level_idx[ch][sb][i] = tmp & 0xff;
00591                         if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
00592                             q->tone_level[ch][sb][i] = 0;
00593                         else
00594                             q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
00595                 }
00596             } else {
00597                 if (sb > 4) {
00598                     for (ch = 0; ch < q->nb_channels; ch++)
00599                         for (i = 0; i < 64; i++) {
00600                             tmp = q->tone_level_idx_base[ch][sb][i / 8] -
00601                                   q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
00602                                   q->tone_level_idx_hi2[ch][sb - 4];
00603                             q->tone_level_idx[ch][sb][i] = tmp & 0xff;
00604                             if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
00605                                 q->tone_level[ch][sb][i] = 0;
00606                             else
00607                                 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
00608                     }
00609                 } else {
00610                     for (ch = 0; ch < q->nb_channels; ch++)
00611                         for (i = 0; i < 64; i++) {
00612                             tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
00613                             if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
00614                                 q->tone_level[ch][sb][i] = 0;
00615                             else
00616                                 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
00617                         }
00618                 }
00619             }
00620         }
00621     }
00622 
00623     return;
00624 }
00625 
00626 
00641 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
00642                 sb_int8_array coding_method, int nb_channels,
00643                 int c, int superblocktype_2_3, int cm_table_select)
00644 {
00645     int ch, sb, j;
00646     int tmp, acc, esp_40, comp;
00647     int add1, add2, add3, add4;
00648     int64_t multres;
00649 
00650     // This should never happen
00651     if (nb_channels <= 0)
00652         return;
00653 
00654     if (!superblocktype_2_3) {
00655         /* This case is untested, no samples available */
00656         SAMPLES_NEEDED
00657         for (ch = 0; ch < nb_channels; ch++)
00658             for (sb = 0; sb < 30; sb++) {
00659                 for (j = 1; j < 63; j++) {  // The loop only iterates to 63 so the code doesn't overflow the buffer
00660                     add1 = tone_level_idx[ch][sb][j] - 10;
00661                     if (add1 < 0)
00662                         add1 = 0;
00663                     add2 = add3 = add4 = 0;
00664                     if (sb > 1) {
00665                         add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
00666                         if (add2 < 0)
00667                             add2 = 0;
00668                     }
00669                     if (sb > 0) {
00670                         add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
00671                         if (add3 < 0)
00672                             add3 = 0;
00673                     }
00674                     if (sb < 29) {
00675                         add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
00676                         if (add4 < 0)
00677                             add4 = 0;
00678                     }
00679                     tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
00680                     if (tmp < 0)
00681                         tmp = 0;
00682                     tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
00683                 }
00684                 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
00685             }
00686             acc = 0;
00687             for (ch = 0; ch < nb_channels; ch++)
00688                 for (sb = 0; sb < 30; sb++)
00689                     for (j = 0; j < 64; j++)
00690                         acc += tone_level_idx_temp[ch][sb][j];
00691 
00692             multres = 0x66666667 * (acc * 10);
00693             esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
00694             for (ch = 0;  ch < nb_channels; ch++)
00695                 for (sb = 0; sb < 30; sb++)
00696                     for (j = 0; j < 64; j++) {
00697                         comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
00698                         if (comp < 0)
00699                             comp += 0xff;
00700                         comp /= 256; // signed shift
00701                         switch(sb) {
00702                             case 0:
00703                                 if (comp < 30)
00704                                     comp = 30;
00705                                 comp += 15;
00706                                 break;
00707                             case 1:
00708                                 if (comp < 24)
00709                                     comp = 24;
00710                                 comp += 10;
00711                                 break;
00712                             case 2:
00713                             case 3:
00714                             case 4:
00715                                 if (comp < 16)
00716                                     comp = 16;
00717                         }
00718                         if (comp <= 5)
00719                             tmp = 0;
00720                         else if (comp <= 10)
00721                             tmp = 10;
00722                         else if (comp <= 16)
00723                             tmp = 16;
00724                         else if (comp <= 24)
00725                             tmp = -1;
00726                         else
00727                             tmp = 0;
00728                         coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
00729                     }
00730             for (sb = 0; sb < 30; sb++)
00731                 fix_coding_method_array(sb, nb_channels, coding_method);
00732             for (ch = 0; ch < nb_channels; ch++)
00733                 for (sb = 0; sb < 30; sb++)
00734                     for (j = 0; j < 64; j++)
00735                         if (sb >= 10) {
00736                             if (coding_method[ch][sb][j] < 10)
00737                                 coding_method[ch][sb][j] = 10;
00738                         } else {
00739                             if (sb >= 2) {
00740                                 if (coding_method[ch][sb][j] < 16)
00741                                     coding_method[ch][sb][j] = 16;
00742                             } else {
00743                                 if (coding_method[ch][sb][j] < 30)
00744                                     coding_method[ch][sb][j] = 30;
00745                             }
00746                         }
00747     } else { // superblocktype_2_3 != 0
00748         for (ch = 0; ch < nb_channels; ch++)
00749             for (sb = 0; sb < 30; sb++)
00750                 for (j = 0; j < 64; j++)
00751                     coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
00752     }
00753 
00754     return;
00755 }
00756 
00757 
00769 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
00770 {
00771     int sb, j, k, n, ch, run, channels;
00772     int joined_stereo, zero_encoding, chs;
00773     int type34_first;
00774     float type34_div = 0;
00775     float type34_predictor;
00776     float samples[10], sign_bits[16];
00777 
00778     if (length == 0) {
00779         // If no data use noise
00780         for (sb=sb_min; sb < sb_max; sb++)
00781             build_sb_samples_from_noise (q, sb);
00782 
00783         return;
00784     }
00785 
00786     for (sb = sb_min; sb < sb_max; sb++) {
00787         FIX_NOISE_IDX(q->noise_idx);
00788 
00789         channels = q->nb_channels;
00790 
00791         if (q->nb_channels <= 1 || sb < 12)
00792             joined_stereo = 0;
00793         else if (sb >= 24)
00794             joined_stereo = 1;
00795         else
00796             joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
00797 
00798         if (joined_stereo) {
00799             if (BITS_LEFT(length,gb) >= 16)
00800                 for (j = 0; j < 16; j++)
00801                     sign_bits[j] = get_bits1 (gb);
00802 
00803             for (j = 0; j < 64; j++)
00804                 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
00805                     q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
00806 
00807             fix_coding_method_array(sb, q->nb_channels, q->coding_method);
00808             channels = 1;
00809         }
00810 
00811         for (ch = 0; ch < channels; ch++) {
00812             zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
00813             type34_predictor = 0.0;
00814             type34_first = 1;
00815 
00816             for (j = 0; j < 128; ) {
00817                 switch (q->coding_method[ch][sb][j / 2]) {
00818                     case 8:
00819                         if (BITS_LEFT(length,gb) >= 10) {
00820                             if (zero_encoding) {
00821                                 for (k = 0; k < 5; k++) {
00822                                     if ((j + 2 * k) >= 128)
00823                                         break;
00824                                     samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
00825                                 }
00826                             } else {
00827                                 n = get_bits(gb, 8);
00828                                 for (k = 0; k < 5; k++)
00829                                     samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
00830                             }
00831                             for (k = 0; k < 5; k++)
00832                                 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
00833                         } else {
00834                             for (k = 0; k < 10; k++)
00835                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
00836                         }
00837                         run = 10;
00838                         break;
00839 
00840                     case 10:
00841                         if (BITS_LEFT(length,gb) >= 1) {
00842                             float f = 0.81;
00843 
00844                             if (get_bits1(gb))
00845                                 f = -f;
00846                             f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
00847                             samples[0] = f;
00848                         } else {
00849                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00850                         }
00851                         run = 1;
00852                         break;
00853 
00854                     case 16:
00855                         if (BITS_LEFT(length,gb) >= 10) {
00856                             if (zero_encoding) {
00857                                 for (k = 0; k < 5; k++) {
00858                                     if ((j + k) >= 128)
00859                                         break;
00860                                     samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
00861                                 }
00862                             } else {
00863                                 n = get_bits (gb, 8);
00864                                 for (k = 0; k < 5; k++)
00865                                     samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
00866                             }
00867                         } else {
00868                             for (k = 0; k < 5; k++)
00869                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
00870                         }
00871                         run = 5;
00872                         break;
00873 
00874                     case 24:
00875                         if (BITS_LEFT(length,gb) >= 7) {
00876                             n = get_bits(gb, 7);
00877                             for (k = 0; k < 3; k++)
00878                                 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
00879                         } else {
00880                             for (k = 0; k < 3; k++)
00881                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
00882                         }
00883                         run = 3;
00884                         break;
00885 
00886                     case 30:
00887                         if (BITS_LEFT(length,gb) >= 4) {
00888                             unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
00889                             if (index < FF_ARRAY_ELEMS(type30_dequant)) {
00890                                 samples[0] = type30_dequant[index];
00891                             } else
00892                                 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00893                         } else
00894                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00895 
00896                         run = 1;
00897                         break;
00898 
00899                     case 34:
00900                         if (BITS_LEFT(length,gb) >= 7) {
00901                             if (type34_first) {
00902                                 type34_div = (float)(1 << get_bits(gb, 2));
00903                                 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
00904                                 type34_predictor = samples[0];
00905                                 type34_first = 0;
00906                             } else {
00907                                 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
00908                                 if (index < FF_ARRAY_ELEMS(type34_delta)) {
00909                                     samples[0] = type34_delta[index] / type34_div + type34_predictor;
00910                                     type34_predictor = samples[0];
00911                                 } else
00912                                     samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00913                             }
00914                         } else {
00915                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00916                         }
00917                         run = 1;
00918                         break;
00919 
00920                     default:
00921                         samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00922                         run = 1;
00923                         break;
00924                 }
00925 
00926                 if (joined_stereo) {
00927                     float tmp[10][MPA_MAX_CHANNELS];
00928 
00929                     for (k = 0; k < run; k++) {
00930                         tmp[k][0] = samples[k];
00931                         tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
00932                     }
00933                     for (chs = 0; chs < q->nb_channels; chs++)
00934                         for (k = 0; k < run; k++)
00935                             if ((j + k) < 128)
00936                                 q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs];
00937                 } else {
00938                     for (k = 0; k < run; k++)
00939                         if ((j + k) < 128)
00940                             q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
00941                 }
00942 
00943                 j += run;
00944             } // j loop
00945         } // channel loop
00946     } // subband loop
00947 }
00948 
00949 
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);
01223             csum +=   2 * get_bits(&gb, 8);
01224 
01225         csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
01226 
01227         if (csum != 0) {
01228             q->has_errors = 1;
01229             av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
01230             return;
01231         }
01232     }
01233 
01234     q->sub_packet_list_B[0].packet = NULL;
01235     q->sub_packet_list_D[0].packet = NULL;
01236 
01237     for (i = 0; i < 6; i++)
01238         if (--q->fft_level_exp[i] < 0)
01239             q->fft_level_exp[i] = 0;
01240 
01241     for (i = 0; packet_bytes > 0; i++) {
01242         int j;
01243 
01244         q->sub_packet_list_A[i].next = NULL;
01245 
01246         if (i > 0) {
01247             q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
01248 
01249             /* seek to next block */
01250             init_get_bits(&gb, header.data, header.size*8);
01251             skip_bits(&gb, next_index*8);
01252 
01253             if (next_index >= header.size)
01254                 break;
01255         }
01256 
01257         /* decode subpacket */
01258         packet = &q->sub_packets[i];
01259         qdm2_decode_sub_packet_header(&gb, packet);
01260         next_index = packet->size + get_bits_count(&gb) / 8;
01261         sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
01262 
01263         if (packet->type == 0)
01264             break;
01265 
01266         if (sub_packet_size > packet_bytes) {
01267             if (packet->type != 10 && packet->type != 11 && packet->type != 12)
01268                 break;
01269             packet->size += packet_bytes - sub_packet_size;
01270         }
01271 
01272         packet_bytes -= sub_packet_size;
01273 
01274         /* add subpacket to 'all subpackets' list */
01275         q->sub_packet_list_A[i].packet = packet;
01276 
01277         /* add subpacket to related list */
01278         if (packet->type == 8) {
01279             SAMPLES_NEEDED_2("packet type 8");
01280             return;
01281         } else if (packet->type >= 9 && packet->type <= 12) {
01282             /* packets for MPEG Audio like Synthesis Filter */
01283             QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
01284         } else if (packet->type == 13) {
01285             for (j = 0; j < 6; j++)
01286                 q->fft_level_exp[j] = get_bits(&gb, 6);
01287         } else if (packet->type == 14) {
01288             for (j = 0; j < 6; j++)
01289                 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
01290         } else if (packet->type == 15) {
01291             SAMPLES_NEEDED_2("packet type 15")
01292             return;
01293         } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
01294             /* packets for FFT */
01295             QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
01296         }
01297     } // Packet bytes loop
01298 
01299 /* **************************************************************** */
01300     if (q->sub_packet_list_D[0].packet != NULL) {
01301         process_synthesis_subpackets(q, q->sub_packet_list_D);
01302         q->do_synth_filter = 1;
01303     } else if (q->do_synth_filter) {
01304         process_subpacket_10(q, NULL, 0);
01305         process_subpacket_11(q, NULL, 0);
01306         process_subpacket_12(q, NULL, 0);
01307     }
01308 /* **************************************************************** */
01309 }
01310 
01311 
01312 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
01313                        int offset, int duration, int channel,
01314                        int exp, int phase)
01315 {
01316     if (q->fft_coefs_min_index[duration] < 0)
01317         q->fft_coefs_min_index[duration] = q->fft_coefs_index;
01318 
01319     q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
01320     q->fft_coefs[q->fft_coefs_index].channel = channel;
01321     q->fft_coefs[q->fft_coefs_index].offset = offset;
01322     q->fft_coefs[q->fft_coefs_index].exp = exp;
01323     q->fft_coefs[q->fft_coefs_index].phase = phase;
01324     q->fft_coefs_index++;
01325 }
01326 
01327 
01328 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
01329 {
01330     int channel, stereo, phase, exp;
01331     int local_int_4,  local_int_8,  stereo_phase,  local_int_10;
01332     int local_int_14, stereo_exp, local_int_20, local_int_28;
01333     int n, offset;
01334 
01335     local_int_4 = 0;
01336     local_int_28 = 0;
01337     local_int_20 = 2;
01338     local_int_8 = (4 - duration);
01339     local_int_10 = 1 << (q->group_order - duration - 1);
01340     offset = 1;
01341 
01342     while (1) {
01343         if (q->superblocktype_2_3) {
01344             while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
01345                 offset = 1;
01346                 if (n == 0) {
01347                     local_int_4 += local_int_10;
01348                     local_int_28 += (1 << local_int_8);
01349                 } else {
01350                     local_int_4 += 8*local_int_10;
01351                     local_int_28 += (8 << local_int_8);
01352                 }
01353             }
01354             offset += (n - 2);
01355         } else {
01356             offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
01357             while (offset >= (local_int_10 - 1)) {
01358                 offset += (1 - (local_int_10 - 1));
01359                 local_int_4  += local_int_10;
01360                 local_int_28 += (1 << local_int_8);
01361             }
01362         }
01363 
01364         if (local_int_4 >= q->group_size)
01365             return;
01366 
01367         local_int_14 = (offset >> local_int_8);
01368         if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
01369             return;
01370 
01371         if (q->nb_channels > 1) {
01372             channel = get_bits1(gb);
01373             stereo = get_bits1(gb);
01374         } else {
01375             channel = 0;
01376             stereo = 0;
01377         }
01378 
01379         exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
01380         exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
01381         exp = (exp < 0) ? 0 : exp;
01382 
01383         phase = get_bits(gb, 3);
01384         stereo_exp = 0;
01385         stereo_phase = 0;
01386 
01387         if (stereo) {
01388             stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
01389             stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
01390             if (stereo_phase < 0)
01391                 stereo_phase += 8;
01392         }
01393 
01394         if (q->frequency_range > (local_int_14 + 1)) {
01395             int sub_packet = (local_int_20 + local_int_28);
01396 
01397             qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
01398             if (stereo)
01399                 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
01400         }
01401 
01402         offset++;
01403     }
01404 }
01405 
01406 
01407 static void qdm2_decode_fft_packets (QDM2Context *q)
01408 {
01409     int i, j, min, max, value, type, unknown_flag;
01410     GetBitContext gb;
01411 
01412     if (q->sub_packet_list_B[0].packet == NULL)
01413         return;
01414 
01415     /* reset minimum indexes for FFT coefficients */
01416     q->fft_coefs_index = 0;
01417     for (i=0; i < 5; i++)
01418         q->fft_coefs_min_index[i] = -1;
01419 
01420     /* process subpackets ordered by type, largest type first */
01421     for (i = 0, max = 256; i < q->sub_packets_B; i++) {
01422         QDM2SubPacket *packet= NULL;
01423 
01424         /* find subpacket with largest type less than max */
01425         for (j = 0, min = 0; j < q->sub_packets_B; j++) {
01426             value = q->sub_packet_list_B[j].packet->type;
01427             if (value > min && value < max) {
01428                 min = value;
01429                 packet = q->sub_packet_list_B[j].packet;
01430             }
01431         }
01432 
01433         max = min;
01434 
01435         /* check for errors (?) */
01436         if (!packet)
01437             return;
01438 
01439         if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
01440             return;
01441 
01442         /* decode FFT tones */
01443         init_get_bits (&gb, packet->data, packet->size*8);
01444 
01445         if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
01446             unknown_flag = 1;
01447         else
01448             unknown_flag = 0;
01449 
01450         type = packet->type;
01451 
01452         if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
01453             int duration = q->sub_sampling + 5 - (type & 15);
01454 
01455             if (duration >= 0 && duration < 4)
01456                 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
01457         } else if (type == 31) {
01458             for (j=0; j < 4; j++)
01459                 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
01460         } else if (type == 46) {
01461             for (j=0; j < 6; j++)
01462                 q->fft_level_exp[j] = get_bits(&gb, 6);
01463             for (j=0; j < 4; j++)
01464             qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
01465         }
01466     } // Loop on B packets
01467 
01468     /* calculate maximum indexes for FFT coefficients */
01469     for (i = 0, j = -1; i < 5; i++)
01470         if (q->fft_coefs_min_index[i] >= 0) {
01471             if (j >= 0)
01472                 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
01473             j = i;
01474         }
01475     if (j >= 0)
01476         q->fft_coefs_max_index[j] = q->fft_coefs_index;
01477 }
01478 
01479 
01480 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
01481 {
01482    float level, f[6];
01483    int i;
01484    QDM2Complex c;
01485    const double iscale = 2.0*M_PI / 512.0;
01486 
01487     tone->phase += tone->phase_shift;
01488 
01489     /* calculate current level (maximum amplitude) of tone */
01490     level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
01491     c.im = level * sin(tone->phase*iscale);
01492     c.re = level * cos(tone->phase*iscale);
01493 
01494     /* generate FFT coefficients for tone */
01495     if (tone->duration >= 3 || tone->cutoff >= 3) {
01496         tone->complex[0].im += c.im;
01497         tone->complex[0].re += c.re;
01498         tone->complex[1].im -= c.im;
01499         tone->complex[1].re -= c.re;
01500     } else {
01501         f[1] = -tone->table[4];
01502         f[0] =  tone->table[3] - tone->table[0];
01503         f[2] =  1.0 - tone->table[2] - tone->table[3];
01504         f[3] =  tone->table[1] + tone->table[4] - 1.0;
01505         f[4] =  tone->table[0] - tone->table[1];
01506         f[5] =  tone->table[2];
01507         for (i = 0; i < 2; i++) {
01508             tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
01509             tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
01510         }
01511         for (i = 0; i < 4; i++) {
01512             tone->complex[i].re += c.re * f[i+2];
01513             tone->complex[i].im += c.im * f[i+2];
01514         }
01515     }
01516 
01517     /* copy the tone if it has not yet died out */
01518     if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
01519       memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
01520       q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
01521     }
01522 }
01523 
01524 
01525 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
01526 {
01527     int i, j, ch;
01528     const double iscale = 0.25 * M_PI;
01529 
01530     for (ch = 0; ch < q->channels; ch++) {
01531         memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
01532     }
01533 
01534 
01535     /* apply FFT tones with duration 4 (1 FFT period) */
01536     if (q->fft_coefs_min_index[4] >= 0)
01537         for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
01538             float level;
01539             QDM2Complex c;
01540 
01541             if (q->fft_coefs[i].sub_packet != sub_packet)
01542                 break;
01543 
01544             ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
01545             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];
01546 
01547             c.re = level * cos(q->fft_coefs[i].phase * iscale);
01548             c.im = level * sin(q->fft_coefs[i].phase * iscale);
01549             q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
01550             q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
01551             q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
01552             q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
01553         }
01554 
01555     /* generate existing FFT tones */
01556     for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
01557         qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
01558         q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
01559     }
01560 
01561     /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
01562     for (i = 0; i < 4; i++)
01563         if (q->fft_coefs_min_index[i] >= 0) {
01564             for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
01565                 int offset, four_i;
01566                 FFTTone tone;
01567 
01568                 if (q->fft_coefs[j].sub_packet != sub_packet)
01569                     break;
01570 
01571                 four_i = (4 - i);
01572                 offset = q->fft_coefs[j].offset >> four_i;
01573                 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
01574 
01575                 if (offset < q->frequency_range) {
01576                     if (offset < 2)
01577                         tone.cutoff = offset;
01578                     else
01579                         tone.cutoff = (offset >= 60) ? 3 : 2;
01580 
01581                     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];
01582                     tone.complex = &q->fft.complex[ch][offset];
01583                     tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
01584                     tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
01585                     tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
01586                     tone.duration = i;
01587                     tone.time_index = 0;
01588 
01589                     qdm2_fft_generate_tone(q, &tone);
01590                 }
01591             }
01592             q->fft_coefs_min_index[i] = j;
01593         }
01594 }
01595 
01596 
01597 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
01598 {
01599     const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
01600     int i;
01601     q->fft.complex[channel][0].re *= 2.0f;
01602     q->fft.complex[channel][0].im = 0.0f;
01603     q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
01604     /* add samples to output buffer */
01605     for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
01606         q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
01607 }
01608 
01609 
01614 static void qdm2_synthesis_filter (QDM2Context *q, int index)
01615 {
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         float *samples_ptr = q->samples + ch;
01628 
01629         for (i = 0; i < 8; i++) {
01630             ff_mpa_synth_filter_float(&q->mpadsp,
01631                 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
01632                 ff_mpa_synth_window_float, &dither_state,
01633                 samples_ptr, q->nb_channels,
01634                 q->sb_samples[ch][(8 * index) + i]);
01635             samples_ptr += 32 * q->nb_channels;
01636         }
01637     }
01638 
01639     /* add samples to output buffer */
01640     sub_sampling = (4 >> q->sub_sampling);
01641 
01642     for (ch = 0; ch < q->channels; ch++)
01643         for (i = 0; i < q->frame_size; i++)
01644             q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
01645 }
01646 
01647 
01653 static av_cold void qdm2_init(QDM2Context *q) {
01654     static int initialized = 0;
01655 
01656     if (initialized != 0)
01657         return;
01658     initialized = 1;
01659 
01660     qdm2_init_vlc();
01661     ff_mpa_synth_init_float(ff_mpa_synth_window_float);
01662     softclip_table_init();
01663     rnd_table_init();
01664     init_noise_samples();
01665 
01666     av_log(NULL, AV_LOG_DEBUG, "init done\n");
01667 }
01668 
01669 
01670 #if 0
01671 static void dump_context(QDM2Context *q)
01672 {
01673     int i;
01674 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
01675     PRINT("compressed_data",q->compressed_data);
01676     PRINT("compressed_size",q->compressed_size);
01677     PRINT("frame_size",q->frame_size);
01678     PRINT("checksum_size",q->checksum_size);
01679     PRINT("channels",q->channels);
01680     PRINT("nb_channels",q->nb_channels);
01681     PRINT("fft_frame_size",q->fft_frame_size);
01682     PRINT("fft_size",q->fft_size);
01683     PRINT("sub_sampling",q->sub_sampling);
01684     PRINT("fft_order",q->fft_order);
01685     PRINT("group_order",q->group_order);
01686     PRINT("group_size",q->group_size);
01687     PRINT("sub_packet",q->sub_packet);
01688     PRINT("frequency_range",q->frequency_range);
01689     PRINT("has_errors",q->has_errors);
01690     PRINT("fft_tone_end",q->fft_tone_end);
01691     PRINT("fft_tone_start",q->fft_tone_start);
01692     PRINT("fft_coefs_index",q->fft_coefs_index);
01693     PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
01694     PRINT("cm_table_select",q->cm_table_select);
01695     PRINT("noise_idx",q->noise_idx);
01696 
01697     for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
01698     {
01699     FFTTone *t = &q->fft_tones[i];
01700 
01701     av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
01702     av_log(NULL,AV_LOG_DEBUG,"  level = %f\n", t->level);
01703 //  PRINT(" level", t->level);
01704     PRINT(" phase", t->phase);
01705     PRINT(" phase_shift", t->phase_shift);
01706     PRINT(" duration", t->duration);
01707     PRINT(" samples_im", t->samples_im);
01708     PRINT(" samples_re", t->samples_re);
01709     PRINT(" table", t->table);
01710     }
01711 
01712 }
01713 #endif
01714 
01715 
01719 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
01720 {
01721     QDM2Context *s = avctx->priv_data;
01722     uint8_t *extradata;
01723     int extradata_size;
01724     int tmp_val, tmp, size;
01725 
01726     /* extradata parsing
01727 
01728     Structure:
01729     wave {
01730         frma (QDM2)
01731         QDCA
01732         QDCP
01733     }
01734 
01735     32  size (including this field)
01736     32  tag (=frma)
01737     32  type (=QDM2 or QDMC)
01738 
01739     32  size (including this field, in bytes)
01740     32  tag (=QDCA) // maybe mandatory parameters
01741     32  unknown (=1)
01742     32  channels (=2)
01743     32  samplerate (=44100)
01744     32  bitrate (=96000)
01745     32  block size (=4096)
01746     32  frame size (=256) (for one channel)
01747     32  packet size (=1300)
01748 
01749     32  size (including this field, in bytes)
01750     32  tag (=QDCP) // maybe some tuneable parameters
01751     32  float1 (=1.0)
01752     32  zero ?
01753     32  float2 (=1.0)
01754     32  float3 (=1.0)
01755     32  unknown (27)
01756     32  unknown (8)
01757     32  zero ?
01758     */
01759 
01760     if (!avctx->extradata || (avctx->extradata_size < 48)) {
01761         av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
01762         return -1;
01763     }
01764 
01765     extradata = avctx->extradata;
01766     extradata_size = avctx->extradata_size;
01767 
01768     while (extradata_size > 7) {
01769         if (!memcmp(extradata, "frmaQDM", 7))
01770             break;
01771         extradata++;
01772         extradata_size--;
01773     }
01774 
01775     if (extradata_size < 12) {
01776         av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
01777                extradata_size);
01778         return -1;
01779     }
01780 
01781     if (memcmp(extradata, "frmaQDM", 7)) {
01782         av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
01783         return -1;
01784     }
01785 
01786     if (extradata[7] == 'C') {
01787 //        s->is_qdmc = 1;
01788         av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
01789         return -1;
01790     }
01791 
01792     extradata += 8;
01793     extradata_size -= 8;
01794 
01795     size = AV_RB32(extradata);
01796 
01797     if(size > extradata_size){
01798         av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
01799                extradata_size, size);
01800         return -1;
01801     }
01802 
01803     extradata += 4;
01804     av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
01805     if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
01806         av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
01807         return -1;
01808     }
01809 
01810     extradata += 8;
01811 
01812     avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
01813     extradata += 4;
01814     if (s->channels > MPA_MAX_CHANNELS)
01815         return AVERROR_INVALIDDATA;
01816 
01817     avctx->sample_rate = AV_RB32(extradata);
01818     extradata += 4;
01819 
01820     avctx->bit_rate = AV_RB32(extradata);
01821     extradata += 4;
01822 
01823     s->group_size = AV_RB32(extradata);
01824     extradata += 4;
01825 
01826     s->fft_size = AV_RB32(extradata);
01827     extradata += 4;
01828 
01829     s->checksum_size = AV_RB32(extradata);
01830     if (s->checksum_size >= 1U << 28) {
01831         av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
01832         return AVERROR_INVALIDDATA;
01833     }
01834 
01835     s->fft_order = av_log2(s->fft_size) + 1;
01836     s->fft_frame_size = 2 * s->fft_size; // complex has two floats
01837 
01838     // something like max decodable tones
01839     s->group_order = av_log2(s->group_size) + 1;
01840     s->frame_size = s->group_size / 16; // 16 iterations per super block
01841     if (s->frame_size > QDM2_MAX_FRAME_SIZE)
01842         return AVERROR_INVALIDDATA;
01843 
01844     s->sub_sampling = s->fft_order - 7;
01845     s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
01846 
01847     switch ((s->sub_sampling * 2 + s->channels - 1)) {
01848         case 0: tmp = 40; break;
01849         case 1: tmp = 48; break;
01850         case 2: tmp = 56; break;
01851         case 3: tmp = 72; break;
01852         case 4: tmp = 80; break;
01853         case 5: tmp = 100;break;
01854         default: tmp=s->sub_sampling; break;
01855     }
01856     tmp_val = 0;
01857     if ((tmp * 1000) < avctx->bit_rate)  tmp_val = 1;
01858     if ((tmp * 1440) < avctx->bit_rate)  tmp_val = 2;
01859     if ((tmp * 1760) < avctx->bit_rate)  tmp_val = 3;
01860     if ((tmp * 2240) < avctx->bit_rate)  tmp_val = 4;
01861     s->cm_table_select = tmp_val;
01862 
01863     if (s->sub_sampling == 0)
01864         tmp = 7999;
01865     else
01866         tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
01867     /*
01868     0: 7999 -> 0
01869     1: 20000 -> 2
01870     2: 28000 -> 2
01871     */
01872     if (tmp < 8000)
01873         s->coeff_per_sb_select = 0;
01874     else if (tmp <= 16000)
01875         s->coeff_per_sb_select = 1;
01876     else
01877         s->coeff_per_sb_select = 2;
01878 
01879     // Fail on unknown fft order
01880     if ((s->fft_order < 7) || (s->fft_order > 9)) {
01881         av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
01882         return -1;
01883     }
01884 
01885     ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
01886     ff_mpadsp_init(&s->mpadsp);
01887 
01888     qdm2_init(s);
01889 
01890     avctx->sample_fmt = AV_SAMPLE_FMT_S16;
01891 
01892     avcodec_get_frame_defaults(&s->frame);
01893     avctx->coded_frame = &s->frame;
01894 
01895 //    dump_context(s);
01896     return 0;
01897 }
01898 
01899 
01900 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
01901 {
01902     QDM2Context *s = avctx->priv_data;
01903 
01904     ff_rdft_end(&s->rdft_ctx);
01905 
01906     return 0;
01907 }
01908 
01909 
01910 static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
01911 {
01912     int ch, i;
01913     const int frame_size = (q->frame_size * q->channels);
01914 
01915     /* select input buffer */
01916     q->compressed_data = in;
01917     q->compressed_size = q->checksum_size;
01918 
01919 //  dump_context(q);
01920 
01921     /* copy old block, clear new block of output samples */
01922     memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
01923     memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
01924 
01925     /* decode block of QDM2 compressed data */
01926     if (q->sub_packet == 0) {
01927         q->has_errors = 0; // zero it for a new super block
01928         av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
01929         qdm2_decode_super_block(q);
01930     }
01931 
01932     /* parse subpackets */
01933     if (!q->has_errors) {
01934         if (q->sub_packet == 2)
01935             qdm2_decode_fft_packets(q);
01936 
01937         qdm2_fft_tone_synthesizer(q, q->sub_packet);
01938     }
01939 
01940     /* sound synthesis stage 1 (FFT) */
01941     for (ch = 0; ch < q->channels; ch++) {
01942         qdm2_calculate_fft(q, ch, q->sub_packet);
01943 
01944         if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
01945             SAMPLES_NEEDED_2("has errors, and C list is not empty")
01946             return -1;
01947         }
01948     }
01949 
01950     /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
01951     if (!q->has_errors && q->do_synth_filter)
01952         qdm2_synthesis_filter(q, q->sub_packet);
01953 
01954     q->sub_packet = (q->sub_packet + 1) % 16;
01955 
01956     /* clip and convert output float[] to 16bit signed samples */
01957     for (i = 0; i < frame_size; i++) {
01958         int value = (int)q->output_buffer[i];
01959 
01960         if (value > SOFTCLIP_THRESHOLD)
01961             value = (value >  HARDCLIP_THRESHOLD) ?  32767 :  softclip_table[ value - SOFTCLIP_THRESHOLD];
01962         else if (value < -SOFTCLIP_THRESHOLD)
01963             value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
01964 
01965         out[i] = value;
01966     }
01967 
01968     return 0;
01969 }
01970 
01971 
01972 static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
01973                              int *got_frame_ptr, AVPacket *avpkt)
01974 {
01975     const uint8_t *buf = avpkt->data;
01976     int buf_size = avpkt->size;
01977     QDM2Context *s = avctx->priv_data;
01978     int16_t *out;
01979     int i, ret;
01980 
01981     if(!buf)
01982         return 0;
01983     if(buf_size < s->checksum_size)
01984         return -1;
01985 
01986     /* get output buffer */
01987     s->frame.nb_samples = 16 * s->frame_size;
01988     if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
01989         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
01990         return ret;
01991     }
01992     out = (int16_t *)s->frame.data[0];
01993 
01994     for (i = 0; i < 16; i++) {
01995         if (qdm2_decode(s, buf, out) < 0)
01996             return -1;
01997         out += s->channels * s->frame_size;
01998     }
01999 
02000     *got_frame_ptr   = 1;
02001     *(AVFrame *)data = s->frame;
02002 
02003     return s->checksum_size;
02004 }
02005 
02006 AVCodec ff_qdm2_decoder =
02007 {
02008     .name = "qdm2",
02009     .type = AVMEDIA_TYPE_AUDIO,
02010     .id = CODEC_ID_QDM2,
02011     .priv_data_size = sizeof(QDM2Context),
02012     .init = qdm2_decode_init,
02013     .close = qdm2_decode_close,
02014     .decode = qdm2_decode_frame,
02015     .capabilities = CODEC_CAP_DR1,
02016     .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
02017 };
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