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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 FFmpeg.
00009  *
00010  * FFmpeg is free software; you can redistribute it and/or
00011  * modify it under the terms of the GNU Lesser General Public
00012  * License as published by the Free Software Foundation; either
00013  * version 2.1 of the License, or (at your option) any later version.
00014  *
00015  * FFmpeg is distributed in the hope that it will be useful,
00016  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00017  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
00018  * Lesser General Public License for more details.
00019  *
00020  * You should have received a copy of the GNU Lesser General Public
00021  * License along with FFmpeg; if not, write to the Free Software
00022  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
00023  */
00024 
00033 #include <math.h>
00034 #include <stddef.h>
00035 #include <stdio.h>
00036 
00037 #define ALT_BITSTREAM_READER_LE
00038 #include "avcodec.h"
00039 #include "get_bits.h"
00040 #include "dsputil.h"
00041 #include "fft.h"
00042 #include "mpegaudio.h"
00043 
00044 #include "qdm2data.h"
00045 #include "qdm2_tablegen.h"
00046 
00047 #undef NDEBUG
00048 #include <assert.h>
00049 
00050 
00051 #define QDM2_LIST_ADD(list, size, packet) \
00052 do { \
00053       if (size > 0) { \
00054     list[size - 1].next = &list[size]; \
00055       } \
00056       list[size].packet = packet; \
00057       list[size].next = NULL; \
00058       size++; \
00059 } while(0)
00060 
00061 // Result is 8, 16 or 30
00062 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
00063 
00064 #define FIX_NOISE_IDX(noise_idx) \
00065   if ((noise_idx) >= 3840) \
00066     (noise_idx) -= 3840; \
00067 
00068 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
00069 
00070 #define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
00071 
00072 #define SAMPLES_NEEDED \
00073      av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
00074 
00075 #define SAMPLES_NEEDED_2(why) \
00076      av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
00077 
00078 #define QDM2_MAX_FRAME_SIZE 512
00079 
00080 typedef int8_t sb_int8_array[2][30][64];
00081 
00085 typedef struct {
00086     int type;            
00087     unsigned int size;   
00088     const uint8_t *data; 
00089 } QDM2SubPacket;
00090 
00094 typedef struct QDM2SubPNode {
00095     QDM2SubPacket *packet;      
00096     struct QDM2SubPNode *next; 
00097 } QDM2SubPNode;
00098 
00099 typedef struct {
00100     float re;
00101     float im;
00102 } QDM2Complex;
00103 
00104 typedef struct {
00105     float level;
00106     QDM2Complex *complex;
00107     const float *table;
00108     int   phase;
00109     int   phase_shift;
00110     int   duration;
00111     short time_index;
00112     short cutoff;
00113 } FFTTone;
00114 
00115 typedef struct {
00116     int16_t sub_packet;
00117     uint8_t channel;
00118     int16_t offset;
00119     int16_t exp;
00120     uint8_t phase;
00121 } FFTCoefficient;
00122 
00123 typedef struct {
00124     DECLARE_ALIGNED(16, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
00125 } QDM2FFT;
00126 
00130 typedef struct {
00132     int nb_channels;         
00133     int channels;            
00134     int group_size;          
00135     int fft_size;            
00136     int checksum_size;       
00137 
00139     int group_order;         
00140     int fft_order;           
00141     int fft_frame_size;      
00142     int frame_size;          
00143     int frequency_range;
00144     int sub_sampling;        
00145     int coeff_per_sb_select; 
00146     int cm_table_select;     
00147 
00149     QDM2SubPacket sub_packets[16];      
00150     QDM2SubPNode sub_packet_list_A[16]; 
00151     QDM2SubPNode sub_packet_list_B[16]; 
00152     int sub_packets_B;                  
00153     QDM2SubPNode sub_packet_list_C[16]; 
00154     QDM2SubPNode sub_packet_list_D[16]; 
00155 
00157     FFTTone fft_tones[1000];
00158     int fft_tone_start;
00159     int fft_tone_end;
00160     FFTCoefficient fft_coefs[1000];
00161     int fft_coefs_index;
00162     int fft_coefs_min_index[5];
00163     int fft_coefs_max_index[5];
00164     int fft_level_exp[6];
00165     RDFTContext rdft_ctx;
00166     QDM2FFT fft;
00167 
00169     const uint8_t *compressed_data;
00170     int compressed_size;
00171     float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
00172 
00174     DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512*2];
00175     int synth_buf_offset[MPA_MAX_CHANNELS];
00176     DECLARE_ALIGNED(16, int32_t, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
00177 
00179     float tone_level[MPA_MAX_CHANNELS][30][64];
00180     int8_t coding_method[MPA_MAX_CHANNELS][30][64];
00181     int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
00182     int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
00183     int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
00184     int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
00185     int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
00186     int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
00187     int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
00188 
00189     // Flags
00190     int has_errors;         
00191     int superblocktype_2_3; 
00192     int do_synth_filter;    
00193 
00194     int sub_packet;
00195     int noise_idx; 
00196 } QDM2Context;
00197 
00198 
00199 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
00200 
00201 static VLC vlc_tab_level;
00202 static VLC vlc_tab_diff;
00203 static VLC vlc_tab_run;
00204 static VLC fft_level_exp_alt_vlc;
00205 static VLC fft_level_exp_vlc;
00206 static VLC fft_stereo_exp_vlc;
00207 static VLC fft_stereo_phase_vlc;
00208 static VLC vlc_tab_tone_level_idx_hi1;
00209 static VLC vlc_tab_tone_level_idx_mid;
00210 static VLC vlc_tab_tone_level_idx_hi2;
00211 static VLC vlc_tab_type30;
00212 static VLC vlc_tab_type34;
00213 static VLC vlc_tab_fft_tone_offset[5];
00214 
00215 static const uint16_t qdm2_vlc_offs[] = {
00216     0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
00217 };
00218 
00219 static av_cold void qdm2_init_vlc(void)
00220 {
00221     static int vlcs_initialized = 0;
00222     static VLC_TYPE qdm2_table[3838][2];
00223 
00224     if (!vlcs_initialized) {
00225 
00226         vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]];
00227         vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
00228         init_vlc (&vlc_tab_level, 8, 24,
00229             vlc_tab_level_huffbits, 1, 1,
00230             vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00231 
00232         vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
00233         vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
00234         init_vlc (&vlc_tab_diff, 8, 37,
00235             vlc_tab_diff_huffbits, 1, 1,
00236             vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00237 
00238         vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]];
00239         vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
00240         init_vlc (&vlc_tab_run, 5, 6,
00241             vlc_tab_run_huffbits, 1, 1,
00242             vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00243 
00244         fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
00245         fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
00246         init_vlc (&fft_level_exp_alt_vlc, 8, 28,
00247             fft_level_exp_alt_huffbits, 1, 1,
00248             fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00249 
00250 
00251         fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
00252         fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
00253         init_vlc (&fft_level_exp_vlc, 8, 20,
00254             fft_level_exp_huffbits, 1, 1,
00255             fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00256 
00257         fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
00258         fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
00259         init_vlc (&fft_stereo_exp_vlc, 6, 7,
00260             fft_stereo_exp_huffbits, 1, 1,
00261             fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00262 
00263         fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
00264         fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
00265         init_vlc (&fft_stereo_phase_vlc, 6, 9,
00266             fft_stereo_phase_huffbits, 1, 1,
00267             fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00268 
00269         vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]];
00270         vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
00271         init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
00272             vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
00273             vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00274 
00275         vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]];
00276         vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
00277         init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
00278             vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
00279             vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00280 
00281         vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]];
00282         vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
00283         init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
00284             vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
00285             vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00286 
00287         vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
00288         vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
00289         init_vlc (&vlc_tab_type30, 6, 9,
00290             vlc_tab_type30_huffbits, 1, 1,
00291             vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00292 
00293         vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
00294         vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
00295         init_vlc (&vlc_tab_type34, 5, 10,
00296             vlc_tab_type34_huffbits, 1, 1,
00297             vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00298 
00299         vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
00300         vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
00301         init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
00302             vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
00303             vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00304 
00305         vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
00306         vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
00307         init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
00308             vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
00309             vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00310 
00311         vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
00312         vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
00313         init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
00314             vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
00315             vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00316 
00317         vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
00318         vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
00319         init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
00320             vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
00321             vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00322 
00323         vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
00324         vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
00325         init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
00326             vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
00327             vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
00328 
00329         vlcs_initialized=1;
00330     }
00331 }
00332 
00333 
00334 /* for floating point to fixed point conversion */
00335 static const float f2i_scale = (float) (1 << (FRAC_BITS - 15));
00336 
00337 
00338 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
00339 {
00340     int value;
00341 
00342     value = get_vlc2(gb, vlc->table, vlc->bits, depth);
00343 
00344     /* stage-2, 3 bits exponent escape sequence */
00345     if (value-- == 0)
00346         value = get_bits (gb, get_bits (gb, 3) + 1);
00347 
00348     /* stage-3, optional */
00349     if (flag) {
00350         int tmp = vlc_stage3_values[value];
00351 
00352         if ((value & ~3) > 0)
00353             tmp += get_bits (gb, (value >> 2));
00354         value = tmp;
00355     }
00356 
00357     return value;
00358 }
00359 
00360 
00361 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
00362 {
00363     int value = qdm2_get_vlc (gb, vlc, 0, depth);
00364 
00365     return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
00366 }
00367 
00368 
00378 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
00379     int i;
00380 
00381     for (i=0; i < length; i++)
00382         value -= data[i];
00383 
00384     return (uint16_t)(value & 0xffff);
00385 }
00386 
00387 
00394 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
00395 {
00396     sub_packet->type = get_bits (gb, 8);
00397 
00398     if (sub_packet->type == 0) {
00399         sub_packet->size = 0;
00400         sub_packet->data = NULL;
00401     } else {
00402         sub_packet->size = get_bits (gb, 8);
00403 
00404       if (sub_packet->type & 0x80) {
00405           sub_packet->size <<= 8;
00406           sub_packet->size  |= get_bits (gb, 8);
00407           sub_packet->type  &= 0x7f;
00408       }
00409 
00410       if (sub_packet->type == 0x7f)
00411           sub_packet->type |= (get_bits (gb, 8) << 8);
00412 
00413       sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
00414     }
00415 
00416     av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
00417         sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
00418 }
00419 
00420 
00428 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
00429 {
00430     while (list != NULL && list->packet != NULL) {
00431         if (list->packet->type == type)
00432             return list;
00433         list = list->next;
00434     }
00435     return NULL;
00436 }
00437 
00438 
00445 static void average_quantized_coeffs (QDM2Context *q)
00446 {
00447     int i, j, n, ch, sum;
00448 
00449     n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
00450 
00451     for (ch = 0; ch < q->nb_channels; ch++)
00452         for (i = 0; i < n; i++) {
00453             sum = 0;
00454 
00455             for (j = 0; j < 8; j++)
00456                 sum += q->quantized_coeffs[ch][i][j];
00457 
00458             sum /= 8;
00459             if (sum > 0)
00460                 sum--;
00461 
00462             for (j=0; j < 8; j++)
00463                 q->quantized_coeffs[ch][i][j] = sum;
00464         }
00465 }
00466 
00467 
00475 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
00476 {
00477     int ch, j;
00478 
00479     FIX_NOISE_IDX(q->noise_idx);
00480 
00481     if (!q->nb_channels)
00482         return;
00483 
00484     for (ch = 0; ch < q->nb_channels; ch++)
00485         for (j = 0; j < 64; j++) {
00486             q->sb_samples[ch][j * 2][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
00487             q->sb_samples[ch][j * 2 + 1][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
00488         }
00489 }
00490 
00491 
00500 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
00501 {
00502     int j,k;
00503     int ch;
00504     int run, case_val;
00505     int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4};
00506 
00507     for (ch = 0; ch < channels; ch++) {
00508         for (j = 0; j < 64; ) {
00509             if((coding_method[ch][sb][j] - 8) > 22) {
00510                 run = 1;
00511                 case_val = 8;
00512             } else {
00513                 switch (switchtable[coding_method[ch][sb][j]-8]) {
00514                     case 0: run = 10; case_val = 10; break;
00515                     case 1: run = 1; case_val = 16; break;
00516                     case 2: run = 5; case_val = 24; break;
00517                     case 3: run = 3; case_val = 30; break;
00518                     case 4: run = 1; case_val = 30; break;
00519                     case 5: run = 1; case_val = 8; break;
00520                     default: run = 1; case_val = 8; break;
00521                 }
00522             }
00523             for (k = 0; k < run; k++)
00524                 if (j + k < 128)
00525                     if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
00526                         if (k > 0) {
00527                            SAMPLES_NEEDED
00528                             //not debugged, almost never used
00529                             memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
00530                             memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
00531                         }
00532             j += run;
00533         }
00534     }
00535 }
00536 
00537 
00545 static void fill_tone_level_array (QDM2Context *q, int flag)
00546 {
00547     int i, sb, ch, sb_used;
00548     int tmp, tab;
00549 
00550     // This should never happen
00551     if (q->nb_channels <= 0)
00552         return;
00553 
00554     for (ch = 0; ch < q->nb_channels; ch++)
00555         for (sb = 0; sb < 30; sb++)
00556             for (i = 0; i < 8; i++) {
00557                 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
00558                     tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
00559                           q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
00560                 else
00561                     tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
00562                 if(tmp < 0)
00563                     tmp += 0xff;
00564                 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
00565             }
00566 
00567     sb_used = QDM2_SB_USED(q->sub_sampling);
00568 
00569     if ((q->superblocktype_2_3 != 0) && !flag) {
00570         for (sb = 0; sb < sb_used; sb++)
00571             for (ch = 0; ch < q->nb_channels; ch++)
00572                 for (i = 0; i < 64; i++) {
00573                     q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
00574                     if (q->tone_level_idx[ch][sb][i] < 0)
00575                         q->tone_level[ch][sb][i] = 0;
00576                     else
00577                         q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
00578                 }
00579     } else {
00580         tab = q->superblocktype_2_3 ? 0 : 1;
00581         for (sb = 0; sb < sb_used; sb++) {
00582             if ((sb >= 4) && (sb <= 23)) {
00583                 for (ch = 0; ch < q->nb_channels; ch++)
00584                     for (i = 0; i < 64; i++) {
00585                         tmp = q->tone_level_idx_base[ch][sb][i / 8] -
00586                               q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
00587                               q->tone_level_idx_mid[ch][sb - 4][i / 8] -
00588                               q->tone_level_idx_hi2[ch][sb - 4];
00589                         q->tone_level_idx[ch][sb][i] = tmp & 0xff;
00590                         if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
00591                             q->tone_level[ch][sb][i] = 0;
00592                         else
00593                             q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
00594                 }
00595             } else {
00596                 if (sb > 4) {
00597                     for (ch = 0; ch < q->nb_channels; ch++)
00598                         for (i = 0; i < 64; i++) {
00599                             tmp = q->tone_level_idx_base[ch][sb][i / 8] -
00600                                   q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
00601                                   q->tone_level_idx_hi2[ch][sb - 4];
00602                             q->tone_level_idx[ch][sb][i] = tmp & 0xff;
00603                             if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
00604                                 q->tone_level[ch][sb][i] = 0;
00605                             else
00606                                 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
00607                     }
00608                 } else {
00609                     for (ch = 0; ch < q->nb_channels; ch++)
00610                         for (i = 0; i < 64; i++) {
00611                             tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
00612                             if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
00613                                 q->tone_level[ch][sb][i] = 0;
00614                             else
00615                                 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
00616                         }
00617                 }
00618             }
00619         }
00620     }
00621 
00622     return;
00623 }
00624 
00625 
00640 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
00641                 sb_int8_array coding_method, int nb_channels,
00642                 int c, int superblocktype_2_3, int cm_table_select)
00643 {
00644     int ch, sb, j;
00645     int tmp, acc, esp_40, comp;
00646     int add1, add2, add3, add4;
00647     int64_t multres;
00648 
00649     // This should never happen
00650     if (nb_channels <= 0)
00651         return;
00652 
00653     if (!superblocktype_2_3) {
00654         /* This case is untested, no samples available */
00655         SAMPLES_NEEDED
00656         for (ch = 0; ch < nb_channels; ch++)
00657             for (sb = 0; sb < 30; sb++) {
00658                 for (j = 1; j < 63; j++) {  // The loop only iterates to 63 so the code doesn't overflow the buffer
00659                     add1 = tone_level_idx[ch][sb][j] - 10;
00660                     if (add1 < 0)
00661                         add1 = 0;
00662                     add2 = add3 = add4 = 0;
00663                     if (sb > 1) {
00664                         add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
00665                         if (add2 < 0)
00666                             add2 = 0;
00667                     }
00668                     if (sb > 0) {
00669                         add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
00670                         if (add3 < 0)
00671                             add3 = 0;
00672                     }
00673                     if (sb < 29) {
00674                         add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
00675                         if (add4 < 0)
00676                             add4 = 0;
00677                     }
00678                     tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
00679                     if (tmp < 0)
00680                         tmp = 0;
00681                     tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
00682                 }
00683                 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
00684             }
00685             acc = 0;
00686             for (ch = 0; ch < nb_channels; ch++)
00687                 for (sb = 0; sb < 30; sb++)
00688                     for (j = 0; j < 64; j++)
00689                         acc += tone_level_idx_temp[ch][sb][j];
00690 
00691             multres = 0x66666667 * (acc * 10);
00692             esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
00693             for (ch = 0;  ch < nb_channels; ch++)
00694                 for (sb = 0; sb < 30; sb++)
00695                     for (j = 0; j < 64; j++) {
00696                         comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
00697                         if (comp < 0)
00698                             comp += 0xff;
00699                         comp /= 256; // signed shift
00700                         switch(sb) {
00701                             case 0:
00702                                 if (comp < 30)
00703                                     comp = 30;
00704                                 comp += 15;
00705                                 break;
00706                             case 1:
00707                                 if (comp < 24)
00708                                     comp = 24;
00709                                 comp += 10;
00710                                 break;
00711                             case 2:
00712                             case 3:
00713                             case 4:
00714                                 if (comp < 16)
00715                                     comp = 16;
00716                         }
00717                         if (comp <= 5)
00718                             tmp = 0;
00719                         else if (comp <= 10)
00720                             tmp = 10;
00721                         else if (comp <= 16)
00722                             tmp = 16;
00723                         else if (comp <= 24)
00724                             tmp = -1;
00725                         else
00726                             tmp = 0;
00727                         coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
00728                     }
00729             for (sb = 0; sb < 30; sb++)
00730                 fix_coding_method_array(sb, nb_channels, coding_method);
00731             for (ch = 0; ch < nb_channels; ch++)
00732                 for (sb = 0; sb < 30; sb++)
00733                     for (j = 0; j < 64; j++)
00734                         if (sb >= 10) {
00735                             if (coding_method[ch][sb][j] < 10)
00736                                 coding_method[ch][sb][j] = 10;
00737                         } else {
00738                             if (sb >= 2) {
00739                                 if (coding_method[ch][sb][j] < 16)
00740                                     coding_method[ch][sb][j] = 16;
00741                             } else {
00742                                 if (coding_method[ch][sb][j] < 30)
00743                                     coding_method[ch][sb][j] = 30;
00744                             }
00745                         }
00746     } else { // superblocktype_2_3 != 0
00747         for (ch = 0; ch < nb_channels; ch++)
00748             for (sb = 0; sb < 30; sb++)
00749                 for (j = 0; j < 64; j++)
00750                     coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
00751     }
00752 
00753     return;
00754 }
00755 
00756 
00768 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
00769 {
00770     int sb, j, k, n, ch, run, channels;
00771     int joined_stereo, zero_encoding, chs;
00772     int type34_first;
00773     float type34_div = 0;
00774     float type34_predictor;
00775     float samples[10], sign_bits[16];
00776 
00777     if (length == 0) {
00778         // If no data use noise
00779         for (sb=sb_min; sb < sb_max; sb++)
00780             build_sb_samples_from_noise (q, sb);
00781 
00782         return;
00783     }
00784 
00785     for (sb = sb_min; sb < sb_max; sb++) {
00786         FIX_NOISE_IDX(q->noise_idx);
00787 
00788         channels = q->nb_channels;
00789 
00790         if (q->nb_channels <= 1 || sb < 12)
00791             joined_stereo = 0;
00792         else if (sb >= 24)
00793             joined_stereo = 1;
00794         else
00795             joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
00796 
00797         if (joined_stereo) {
00798             if (BITS_LEFT(length,gb) >= 16)
00799                 for (j = 0; j < 16; j++)
00800                     sign_bits[j] = get_bits1 (gb);
00801 
00802             for (j = 0; j < 64; j++)
00803                 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
00804                     q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
00805 
00806             fix_coding_method_array(sb, q->nb_channels, q->coding_method);
00807             channels = 1;
00808         }
00809 
00810         for (ch = 0; ch < channels; ch++) {
00811             zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
00812             type34_predictor = 0.0;
00813             type34_first = 1;
00814 
00815             for (j = 0; j < 128; ) {
00816                 switch (q->coding_method[ch][sb][j / 2]) {
00817                     case 8:
00818                         if (BITS_LEFT(length,gb) >= 10) {
00819                             if (zero_encoding) {
00820                                 for (k = 0; k < 5; k++) {
00821                                     if ((j + 2 * k) >= 128)
00822                                         break;
00823                                     samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
00824                                 }
00825                             } else {
00826                                 n = get_bits(gb, 8);
00827                                 for (k = 0; k < 5; k++)
00828                                     samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
00829                             }
00830                             for (k = 0; k < 5; k++)
00831                                 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
00832                         } else {
00833                             for (k = 0; k < 10; k++)
00834                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
00835                         }
00836                         run = 10;
00837                         break;
00838 
00839                     case 10:
00840                         if (BITS_LEFT(length,gb) >= 1) {
00841                             float f = 0.81;
00842 
00843                             if (get_bits1(gb))
00844                                 f = -f;
00845                             f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
00846                             samples[0] = f;
00847                         } else {
00848                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00849                         }
00850                         run = 1;
00851                         break;
00852 
00853                     case 16:
00854                         if (BITS_LEFT(length,gb) >= 10) {
00855                             if (zero_encoding) {
00856                                 for (k = 0; k < 5; k++) {
00857                                     if ((j + k) >= 128)
00858                                         break;
00859                                     samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
00860                                 }
00861                             } else {
00862                                 n = get_bits (gb, 8);
00863                                 for (k = 0; k < 5; k++)
00864                                     samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
00865                             }
00866                         } else {
00867                             for (k = 0; k < 5; k++)
00868                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
00869                         }
00870                         run = 5;
00871                         break;
00872 
00873                     case 24:
00874                         if (BITS_LEFT(length,gb) >= 7) {
00875                             n = get_bits(gb, 7);
00876                             for (k = 0; k < 3; k++)
00877                                 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
00878                         } else {
00879                             for (k = 0; k < 3; k++)
00880                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
00881                         }
00882                         run = 3;
00883                         break;
00884 
00885                     case 30:
00886                         if (BITS_LEFT(length,gb) >= 4)
00887                             samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
00888                         else
00889                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00890 
00891                         run = 1;
00892                         break;
00893 
00894                     case 34:
00895                         if (BITS_LEFT(length,gb) >= 7) {
00896                             if (type34_first) {
00897                                 type34_div = (float)(1 << get_bits(gb, 2));
00898                                 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
00899                                 type34_predictor = samples[0];
00900                                 type34_first = 0;
00901                             } else {
00902                                 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
00903                                 type34_predictor = samples[0];
00904                             }
00905                         } else {
00906                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00907                         }
00908                         run = 1;
00909                         break;
00910 
00911                     default:
00912                         samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
00913                         run = 1;
00914                         break;
00915                 }
00916 
00917                 if (joined_stereo) {
00918                     float tmp[10][MPA_MAX_CHANNELS];
00919 
00920                     for (k = 0; k < run; k++) {
00921                         tmp[k][0] = samples[k];
00922                         tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
00923                     }
00924                     for (chs = 0; chs < q->nb_channels; chs++)
00925                         for (k = 0; k < run; k++)
00926                             if ((j + k) < 128)
00927                                 q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
00928                 } else {
00929                     for (k = 0; k < run; k++)
00930                         if ((j + k) < 128)
00931                             q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
00932                 }
00933 
00934                 j += run;
00935             } // j loop
00936         } // channel loop
00937     } // subband loop
00938 }
00939 
00940 
00951 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
00952 {
00953     int i, k, run, level, diff;
00954 
00955     if (BITS_LEFT(length,gb) < 16)
00956         return;
00957     level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
00958 
00959     quantized_coeffs[0] = level;
00960 
00961     for (i = 0; i < 7; ) {
00962         if (BITS_LEFT(length,gb) < 16)
00963             break;
00964         run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
00965 
00966         if (BITS_LEFT(length,gb) < 16)
00967             break;
00968         diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
00969 
00970         for (k = 1; k <= run; k++)
00971             quantized_coeffs[i + k] = (level + ((k * diff) / run));
00972 
00973         level += diff;
00974         i += run;
00975     }
00976 }
00977 
00978 
00988 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
00989 {
00990     int sb, j, k, n, ch;
00991 
00992     for (ch = 0; ch < q->nb_channels; ch++) {
00993         init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
00994 
00995         if (BITS_LEFT(length,gb) < 16) {
00996             memset(q->quantized_coeffs[ch][0], 0, 8);
00997             break;
00998         }
00999     }
01000 
01001     n = q->sub_sampling + 1;
01002 
01003     for (sb = 0; sb < n; sb++)
01004         for (ch = 0; ch < q->nb_channels; ch++)
01005             for (j = 0; j < 8; j++) {
01006                 if (BITS_LEFT(length,gb) < 1)
01007                     break;
01008                 if (get_bits1(gb)) {
01009                     for (k=0; k < 8; k++) {
01010                         if (BITS_LEFT(length,gb) < 16)
01011                             break;
01012                         q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
01013                     }
01014                 } else {
01015                     for (k=0; k < 8; k++)
01016                         q->tone_level_idx_hi1[ch][sb][j][k] = 0;
01017                 }
01018             }
01019 
01020     n = QDM2_SB_USED(q->sub_sampling) - 4;
01021 
01022     for (sb = 0; sb < n; sb++)
01023         for (ch = 0; ch < q->nb_channels; ch++) {
01024             if (BITS_LEFT(length,gb) < 16)
01025                 break;
01026             q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
01027             if (sb > 19)
01028                 q->tone_level_idx_hi2[ch][sb] -= 16;
01029             else
01030                 for (j = 0; j < 8; j++)
01031                     q->tone_level_idx_mid[ch][sb][j] = -16;
01032         }
01033 
01034     n = QDM2_SB_USED(q->sub_sampling) - 5;
01035 
01036     for (sb = 0; sb < n; sb++)
01037         for (ch = 0; ch < q->nb_channels; ch++)
01038             for (j = 0; j < 8; j++) {
01039                 if (BITS_LEFT(length,gb) < 16)
01040                     break;
01041                 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
01042             }
01043 }
01044 
01051 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
01052 {
01053     GetBitContext gb;
01054     int i, j, k, n, ch, run, level, diff;
01055 
01056     init_get_bits(&gb, node->packet->data, node->packet->size*8);
01057 
01058     n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
01059 
01060     for (i = 1; i < n; i++)
01061         for (ch=0; ch < q->nb_channels; ch++) {
01062             level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
01063             q->quantized_coeffs[ch][i][0] = level;
01064 
01065             for (j = 0; j < (8 - 1); ) {
01066                 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
01067                 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
01068 
01069                 for (k = 1; k <= run; k++)
01070                     q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
01071 
01072                 level += diff;
01073                 j += run;
01074             }
01075         }
01076 
01077     for (ch = 0; ch < q->nb_channels; ch++)
01078         for (i = 0; i < 8; i++)
01079             q->quantized_coeffs[ch][0][i] = 0;
01080 }
01081 
01082 
01090 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
01091 {
01092     GetBitContext gb;
01093 
01094     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
01095 
01096     if (length != 0) {
01097         init_tone_level_dequantization(q, &gb, length);
01098         fill_tone_level_array(q, 1);
01099     } else {
01100         fill_tone_level_array(q, 0);
01101     }
01102 }
01103 
01104 
01112 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
01113 {
01114     GetBitContext gb;
01115 
01116     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
01117     if (length >= 32) {
01118         int c = get_bits (&gb, 13);
01119 
01120         if (c > 3)
01121             fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
01122                                       q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
01123     }
01124 
01125     synthfilt_build_sb_samples(q, &gb, length, 0, 8);
01126 }
01127 
01128 
01136 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
01137 {
01138     GetBitContext gb;
01139 
01140     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
01141     synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
01142 }
01143 
01144 /*
01145  * Process new subpackets for synthesis filter
01146  *
01147  * @param q       context
01148  * @param list    list with synthesis filter packets (list D)
01149  */
01150 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
01151 {
01152     QDM2SubPNode *nodes[4];
01153 
01154     nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
01155     if (nodes[0] != NULL)
01156         process_subpacket_9(q, nodes[0]);
01157 
01158     nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
01159     if (nodes[1] != NULL)
01160         process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
01161     else
01162         process_subpacket_10(q, NULL, 0);
01163 
01164     nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
01165     if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
01166         process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
01167     else
01168         process_subpacket_11(q, NULL, 0);
01169 
01170     nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
01171     if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
01172         process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
01173     else
01174         process_subpacket_12(q, NULL, 0);
01175 }
01176 
01177 
01178 /*
01179  * Decode superblock, fill packet lists.
01180  *
01181  * @param q    context
01182  */
01183 static void qdm2_decode_super_block (QDM2Context *q)
01184 {
01185     GetBitContext gb;
01186     QDM2SubPacket header, *packet;
01187     int i, packet_bytes, sub_packet_size, sub_packets_D;
01188     unsigned int next_index = 0;
01189 
01190     memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
01191     memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
01192     memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
01193 
01194     q->sub_packets_B = 0;
01195     sub_packets_D = 0;
01196 
01197     average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
01198 
01199     init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
01200     qdm2_decode_sub_packet_header(&gb, &header);
01201 
01202     if (header.type < 2 || header.type >= 8) {
01203         q->has_errors = 1;
01204         av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
01205         return;
01206     }
01207 
01208     q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
01209     packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
01210 
01211     init_get_bits(&gb, header.data, header.size*8);
01212 
01213     if (header.type == 2 || header.type == 4 || header.type == 5) {
01214         int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8);
01215 
01216         csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
01217 
01218         if (csum != 0) {
01219             q->has_errors = 1;
01220             av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
01221             return;
01222         }
01223     }
01224 
01225     q->sub_packet_list_B[0].packet = NULL;
01226     q->sub_packet_list_D[0].packet = NULL;
01227 
01228     for (i = 0; i < 6; i++)
01229         if (--q->fft_level_exp[i] < 0)
01230             q->fft_level_exp[i] = 0;
01231 
01232     for (i = 0; packet_bytes > 0; i++) {
01233         int j;
01234 
01235         q->sub_packet_list_A[i].next = NULL;
01236 
01237         if (i > 0) {
01238             q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
01239 
01240             /* seek to next block */
01241             init_get_bits(&gb, header.data, header.size*8);
01242             skip_bits(&gb, next_index*8);
01243 
01244             if (next_index >= header.size)
01245                 break;
01246         }
01247 
01248         /* decode subpacket */
01249         packet = &q->sub_packets[i];
01250         qdm2_decode_sub_packet_header(&gb, packet);
01251         next_index = packet->size + get_bits_count(&gb) / 8;
01252         sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
01253 
01254         if (packet->type == 0)
01255             break;
01256 
01257         if (sub_packet_size > packet_bytes) {
01258             if (packet->type != 10 && packet->type != 11 && packet->type != 12)
01259                 break;
01260             packet->size += packet_bytes - sub_packet_size;
01261         }
01262 
01263         packet_bytes -= sub_packet_size;
01264 
01265         /* add subpacket to 'all subpackets' list */
01266         q->sub_packet_list_A[i].packet = packet;
01267 
01268         /* add subpacket to related list */
01269         if (packet->type == 8) {
01270             SAMPLES_NEEDED_2("packet type 8");
01271             return;
01272         } else if (packet->type >= 9 && packet->type <= 12) {
01273             /* packets for MPEG Audio like Synthesis Filter */
01274             QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
01275         } else if (packet->type == 13) {
01276             for (j = 0; j < 6; j++)
01277                 q->fft_level_exp[j] = get_bits(&gb, 6);
01278         } else if (packet->type == 14) {
01279             for (j = 0; j < 6; j++)
01280                 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
01281         } else if (packet->type == 15) {
01282             SAMPLES_NEEDED_2("packet type 15")
01283             return;
01284         } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
01285             /* packets for FFT */
01286             QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
01287         }
01288     } // Packet bytes loop
01289 
01290 /* **************************************************************** */
01291     if (q->sub_packet_list_D[0].packet != NULL) {
01292         process_synthesis_subpackets(q, q->sub_packet_list_D);
01293         q->do_synth_filter = 1;
01294     } else if (q->do_synth_filter) {
01295         process_subpacket_10(q, NULL, 0);
01296         process_subpacket_11(q, NULL, 0);
01297         process_subpacket_12(q, NULL, 0);
01298     }
01299 /* **************************************************************** */
01300 }
01301 
01302 
01303 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
01304                        int offset, int duration, int channel,
01305                        int exp, int phase)
01306 {
01307     if (q->fft_coefs_min_index[duration] < 0)
01308         q->fft_coefs_min_index[duration] = q->fft_coefs_index;
01309 
01310     q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
01311     q->fft_coefs[q->fft_coefs_index].channel = channel;
01312     q->fft_coefs[q->fft_coefs_index].offset = offset;
01313     q->fft_coefs[q->fft_coefs_index].exp = exp;
01314     q->fft_coefs[q->fft_coefs_index].phase = phase;
01315     q->fft_coefs_index++;
01316 }
01317 
01318 
01319 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
01320 {
01321     int channel, stereo, phase, exp;
01322     int local_int_4,  local_int_8,  stereo_phase,  local_int_10;
01323     int local_int_14, stereo_exp, local_int_20, local_int_28;
01324     int n, offset;
01325 
01326     local_int_4 = 0;
01327     local_int_28 = 0;
01328     local_int_20 = 2;
01329     local_int_8 = (4 - duration);
01330     local_int_10 = 1 << (q->group_order - duration - 1);
01331     offset = 1;
01332 
01333     while (get_bits_left(gb)>0) {
01334         if (q->superblocktype_2_3) {
01335             while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
01336                 offset = 1;
01337                 if (n == 0) {
01338                     local_int_4 += local_int_10;
01339                     local_int_28 += (1 << local_int_8);
01340                 } else {
01341                     local_int_4 += 8*local_int_10;
01342                     local_int_28 += (8 << local_int_8);
01343                 }
01344             }
01345             offset += (n - 2);
01346         } else {
01347             offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
01348             while (offset >= (local_int_10 - 1)) {
01349                 offset += (1 - (local_int_10 - 1));
01350                 local_int_4  += local_int_10;
01351                 local_int_28 += (1 << local_int_8);
01352             }
01353         }
01354 
01355         if (local_int_4 >= q->group_size)
01356             return;
01357 
01358         local_int_14 = (offset >> local_int_8);
01359         if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
01360             return;
01361 
01362         if (q->nb_channels > 1) {
01363             channel = get_bits1(gb);
01364             stereo = get_bits1(gb);
01365         } else {
01366             channel = 0;
01367             stereo = 0;
01368         }
01369 
01370         exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
01371         exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
01372         exp = (exp < 0) ? 0 : exp;
01373 
01374         phase = get_bits(gb, 3);
01375         stereo_exp = 0;
01376         stereo_phase = 0;
01377 
01378         if (stereo) {
01379             stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
01380             stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
01381             if (stereo_phase < 0)
01382                 stereo_phase += 8;
01383         }
01384 
01385         if (q->frequency_range > (local_int_14 + 1)) {
01386             int sub_packet = (local_int_20 + local_int_28);
01387 
01388             qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
01389             if (stereo)
01390                 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
01391         }
01392 
01393         offset++;
01394     }
01395 }
01396 
01397 
01398 static void qdm2_decode_fft_packets (QDM2Context *q)
01399 {
01400     int i, j, min, max, value, type, unknown_flag;
01401     GetBitContext gb;
01402 
01403     if (q->sub_packet_list_B[0].packet == NULL)
01404         return;
01405 
01406     /* reset minimum indexes for FFT coefficients */
01407     q->fft_coefs_index = 0;
01408     for (i=0; i < 5; i++)
01409         q->fft_coefs_min_index[i] = -1;
01410 
01411     /* process subpackets ordered by type, largest type first */
01412     for (i = 0, max = 256; i < q->sub_packets_B; i++) {
01413         QDM2SubPacket *packet= NULL;
01414 
01415         /* find subpacket with largest type less than max */
01416         for (j = 0, min = 0; j < q->sub_packets_B; j++) {
01417             value = q->sub_packet_list_B[j].packet->type;
01418             if (value > min && value < max) {
01419                 min = value;
01420                 packet = q->sub_packet_list_B[j].packet;
01421             }
01422         }
01423 
01424         max = min;
01425 
01426         /* check for errors (?) */
01427         if (!packet)
01428             return;
01429 
01430         if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
01431             return;
01432 
01433         /* decode FFT tones */
01434         init_get_bits (&gb, packet->data, packet->size*8);
01435 
01436         if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
01437             unknown_flag = 1;
01438         else
01439             unknown_flag = 0;
01440 
01441         type = packet->type;
01442 
01443         if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
01444             int duration = q->sub_sampling + 5 - (type & 15);
01445 
01446             if (duration >= 0 && duration < 4)
01447                 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
01448         } else if (type == 31) {
01449             for (j=0; j < 4; j++)
01450                 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
01451         } else if (type == 46) {
01452             for (j=0; j < 6; j++)
01453                 q->fft_level_exp[j] = get_bits(&gb, 6);
01454             for (j=0; j < 4; j++)
01455             qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
01456         }
01457     } // Loop on B packets
01458 
01459     /* calculate maximum indexes for FFT coefficients */
01460     for (i = 0, j = -1; i < 5; i++)
01461         if (q->fft_coefs_min_index[i] >= 0) {
01462             if (j >= 0)
01463                 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
01464             j = i;
01465         }
01466     if (j >= 0)
01467         q->fft_coefs_max_index[j] = q->fft_coefs_index;
01468 }
01469 
01470 
01471 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
01472 {
01473    float level, f[6];
01474    int i;
01475    QDM2Complex c;
01476    const double iscale = 2.0*M_PI / 512.0;
01477 
01478     tone->phase += tone->phase_shift;
01479 
01480     /* calculate current level (maximum amplitude) of tone */
01481     level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
01482     c.im = level * sin(tone->phase*iscale);
01483     c.re = level * cos(tone->phase*iscale);
01484 
01485     /* generate FFT coefficients for tone */
01486     if (tone->duration >= 3 || tone->cutoff >= 3) {
01487         tone->complex[0].im += c.im;
01488         tone->complex[0].re += c.re;
01489         tone->complex[1].im -= c.im;
01490         tone->complex[1].re -= c.re;
01491     } else {
01492         f[1] = -tone->table[4];
01493         f[0] =  tone->table[3] - tone->table[0];
01494         f[2] =  1.0 - tone->table[2] - tone->table[3];
01495         f[3] =  tone->table[1] + tone->table[4] - 1.0;
01496         f[4] =  tone->table[0] - tone->table[1];
01497         f[5] =  tone->table[2];
01498         for (i = 0; i < 2; i++) {
01499             tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
01500             tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
01501         }
01502         for (i = 0; i < 4; i++) {
01503             tone->complex[i].re += c.re * f[i+2];
01504             tone->complex[i].im += c.im * f[i+2];
01505         }
01506     }
01507 
01508     /* copy the tone if it has not yet died out */
01509     if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
01510       memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
01511       q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
01512     }
01513 }
01514 
01515 
01516 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
01517 {
01518     int i, j, ch;
01519     const double iscale = 0.25 * M_PI;
01520 
01521     for (ch = 0; ch < q->channels; ch++) {
01522         memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
01523     }
01524 
01525 
01526     /* apply FFT tones with duration 4 (1 FFT period) */
01527     if (q->fft_coefs_min_index[4] >= 0)
01528         for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
01529             float level;
01530             QDM2Complex c;
01531 
01532             if (q->fft_coefs[i].sub_packet != sub_packet)
01533                 break;
01534 
01535             ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
01536             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];
01537 
01538             c.re = level * cos(q->fft_coefs[i].phase * iscale);
01539             c.im = level * sin(q->fft_coefs[i].phase * iscale);
01540             q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
01541             q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
01542             q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
01543             q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
01544         }
01545 
01546     /* generate existing FFT tones */
01547     for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
01548         qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
01549         q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
01550     }
01551 
01552     /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
01553     for (i = 0; i < 4; i++)
01554         if (q->fft_coefs_min_index[i] >= 0) {
01555             for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
01556                 int offset, four_i;
01557                 FFTTone tone;
01558 
01559                 if (q->fft_coefs[j].sub_packet != sub_packet)
01560                     break;
01561 
01562                 four_i = (4 - i);
01563                 offset = q->fft_coefs[j].offset >> four_i;
01564                 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
01565 
01566                 if (offset < q->frequency_range) {
01567                     if (offset < 2)
01568                         tone.cutoff = offset;
01569                     else
01570                         tone.cutoff = (offset >= 60) ? 3 : 2;
01571 
01572                     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];
01573                     tone.complex = &q->fft.complex[ch][offset];
01574                     tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
01575                     tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
01576                     tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
01577                     tone.duration = i;
01578                     tone.time_index = 0;
01579 
01580                     qdm2_fft_generate_tone(q, &tone);
01581                 }
01582             }
01583             q->fft_coefs_min_index[i] = j;
01584         }
01585 }
01586 
01587 
01588 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
01589 {
01590     const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
01591     int i;
01592     q->fft.complex[channel][0].re *= 2.0f;
01593     q->fft.complex[channel][0].im = 0.0f;
01594     ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
01595     /* add samples to output buffer */
01596     for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
01597         q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
01598 }
01599 
01600 
01605 static void qdm2_synthesis_filter (QDM2Context *q, int index)
01606 {
01607     OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
01608     int i, k, ch, sb_used, sub_sampling, dither_state = 0;
01609 
01610     /* copy sb_samples */
01611     sb_used = QDM2_SB_USED(q->sub_sampling);
01612 
01613     for (ch = 0; ch < q->channels; ch++)
01614         for (i = 0; i < 8; i++)
01615             for (k=sb_used; k < SBLIMIT; k++)
01616                 q->sb_samples[ch][(8 * index) + i][k] = 0;
01617 
01618     for (ch = 0; ch < q->nb_channels; ch++) {
01619         OUT_INT *samples_ptr = samples + ch;
01620 
01621         for (i = 0; i < 8; i++) {
01622             ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
01623                 ff_mpa_synth_window, &dither_state,
01624                 samples_ptr, q->nb_channels,
01625                 q->sb_samples[ch][(8 * index) + i]);
01626             samples_ptr += 32 * q->nb_channels;
01627         }
01628     }
01629 
01630     /* add samples to output buffer */
01631     sub_sampling = (4 >> q->sub_sampling);
01632 
01633     for (ch = 0; ch < q->channels; ch++)
01634         for (i = 0; i < q->frame_size; i++)
01635             q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
01636 }
01637 
01638 
01644 static av_cold void qdm2_init(QDM2Context *q) {
01645     static int initialized = 0;
01646 
01647     if (initialized != 0)
01648         return;
01649     initialized = 1;
01650 
01651     qdm2_init_vlc();
01652     ff_mpa_synth_init(ff_mpa_synth_window);
01653     softclip_table_init();
01654     rnd_table_init();
01655     init_noise_samples();
01656 
01657     av_log(NULL, AV_LOG_DEBUG, "init done\n");
01658 }
01659 
01660 
01661 #if 0
01662 static void dump_context(QDM2Context *q)
01663 {
01664     int i;
01665 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
01666     PRINT("compressed_data",q->compressed_data);
01667     PRINT("compressed_size",q->compressed_size);
01668     PRINT("frame_size",q->frame_size);
01669     PRINT("checksum_size",q->checksum_size);
01670     PRINT("channels",q->channels);
01671     PRINT("nb_channels",q->nb_channels);
01672     PRINT("fft_frame_size",q->fft_frame_size);
01673     PRINT("fft_size",q->fft_size);
01674     PRINT("sub_sampling",q->sub_sampling);
01675     PRINT("fft_order",q->fft_order);
01676     PRINT("group_order",q->group_order);
01677     PRINT("group_size",q->group_size);
01678     PRINT("sub_packet",q->sub_packet);
01679     PRINT("frequency_range",q->frequency_range);
01680     PRINT("has_errors",q->has_errors);
01681     PRINT("fft_tone_end",q->fft_tone_end);
01682     PRINT("fft_tone_start",q->fft_tone_start);
01683     PRINT("fft_coefs_index",q->fft_coefs_index);
01684     PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
01685     PRINT("cm_table_select",q->cm_table_select);
01686     PRINT("noise_idx",q->noise_idx);
01687 
01688     for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
01689     {
01690     FFTTone *t = &q->fft_tones[i];
01691 
01692     av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
01693     av_log(NULL,AV_LOG_DEBUG,"  level = %f\n", t->level);
01694 //  PRINT(" level", t->level);
01695     PRINT(" phase", t->phase);
01696     PRINT(" phase_shift", t->phase_shift);
01697     PRINT(" duration", t->duration);
01698     PRINT(" samples_im", t->samples_im);
01699     PRINT(" samples_re", t->samples_re);
01700     PRINT(" table", t->table);
01701     }
01702 
01703 }
01704 #endif
01705 
01706 
01710 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
01711 {
01712     QDM2Context *s = avctx->priv_data;
01713     uint8_t *extradata;
01714     int extradata_size;
01715     int tmp_val, tmp, size;
01716 
01717     /* extradata parsing
01718 
01719     Structure:
01720     wave {
01721         frma (QDM2)
01722         QDCA
01723         QDCP
01724     }
01725 
01726     32  size (including this field)
01727     32  tag (=frma)
01728     32  type (=QDM2 or QDMC)
01729 
01730     32  size (including this field, in bytes)
01731     32  tag (=QDCA) // maybe mandatory parameters
01732     32  unknown (=1)
01733     32  channels (=2)
01734     32  samplerate (=44100)
01735     32  bitrate (=96000)
01736     32  block size (=4096)
01737     32  frame size (=256) (for one channel)
01738     32  packet size (=1300)
01739 
01740     32  size (including this field, in bytes)
01741     32  tag (=QDCP) // maybe some tuneable parameters
01742     32  float1 (=1.0)
01743     32  zero ?
01744     32  float2 (=1.0)
01745     32  float3 (=1.0)
01746     32  unknown (27)
01747     32  unknown (8)
01748     32  zero ?
01749     */
01750 
01751     if (!avctx->extradata || (avctx->extradata_size < 48)) {
01752         av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
01753         return -1;
01754     }
01755 
01756     extradata = avctx->extradata;
01757     extradata_size = avctx->extradata_size;
01758 
01759     while (extradata_size > 7) {
01760         if (!memcmp(extradata, "frmaQDM", 7))
01761             break;
01762         extradata++;
01763         extradata_size--;
01764     }
01765 
01766     if (extradata_size < 12) {
01767         av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
01768                extradata_size);
01769         return -1;
01770     }
01771 
01772     if (memcmp(extradata, "frmaQDM", 7)) {
01773         av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
01774         return -1;
01775     }
01776 
01777     if (extradata[7] == 'C') {
01778 //        s->is_qdmc = 1;
01779         av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
01780         return -1;
01781     }
01782 
01783     extradata += 8;
01784     extradata_size -= 8;
01785 
01786     size = AV_RB32(extradata);
01787 
01788     if(size > extradata_size){
01789         av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
01790                extradata_size, size);
01791         return -1;
01792     }
01793 
01794     extradata += 4;
01795     av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
01796     if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
01797         av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
01798         return -1;
01799     }
01800 
01801     extradata += 8;
01802 
01803     avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
01804     extradata += 4;
01805     if (s->channels > MPA_MAX_CHANNELS)
01806         return AVERROR_INVALIDDATA;
01807 
01808     avctx->sample_rate = AV_RB32(extradata);
01809     extradata += 4;
01810 
01811     avctx->bit_rate = AV_RB32(extradata);
01812     extradata += 4;
01813 
01814     s->group_size = AV_RB32(extradata);
01815     extradata += 4;
01816 
01817     s->fft_size = AV_RB32(extradata);
01818     extradata += 4;
01819 
01820     s->checksum_size = AV_RB32(extradata);
01821 
01822     s->fft_order = av_log2(s->fft_size) + 1;
01823     s->fft_frame_size = 2 * s->fft_size; // complex has two floats
01824 
01825     // something like max decodable tones
01826     s->group_order = av_log2(s->group_size) + 1;
01827     s->frame_size = s->group_size / 16; // 16 iterations per super block
01828     if (s->frame_size > QDM2_MAX_FRAME_SIZE)
01829         return AVERROR_INVALIDDATA;
01830 
01831     s->sub_sampling = s->fft_order - 7;
01832     s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
01833 
01834     switch ((s->sub_sampling * 2 + s->channels - 1)) {
01835         case 0: tmp = 40; break;
01836         case 1: tmp = 48; break;
01837         case 2: tmp = 56; break;
01838         case 3: tmp = 72; break;
01839         case 4: tmp = 80; break;
01840         case 5: tmp = 100;break;
01841         default: tmp=s->sub_sampling; break;
01842     }
01843     tmp_val = 0;
01844     if ((tmp * 1000) < avctx->bit_rate)  tmp_val = 1;
01845     if ((tmp * 1440) < avctx->bit_rate)  tmp_val = 2;
01846     if ((tmp * 1760) < avctx->bit_rate)  tmp_val = 3;
01847     if ((tmp * 2240) < avctx->bit_rate)  tmp_val = 4;
01848     s->cm_table_select = tmp_val;
01849 
01850     if (s->sub_sampling == 0)
01851         tmp = 7999;
01852     else
01853         tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
01854     /*
01855     0: 7999 -> 0
01856     1: 20000 -> 2
01857     2: 28000 -> 2
01858     */
01859     if (tmp < 8000)
01860         s->coeff_per_sb_select = 0;
01861     else if (tmp <= 16000)
01862         s->coeff_per_sb_select = 1;
01863     else
01864         s->coeff_per_sb_select = 2;
01865 
01866     // Fail on unknown fft order
01867     if ((s->fft_order < 7) || (s->fft_order > 9)) {
01868         av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
01869         return -1;
01870     }
01871 
01872     ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
01873 
01874     qdm2_init(s);
01875 
01876     avctx->sample_fmt = SAMPLE_FMT_S16;
01877 
01878 //    dump_context(s);
01879     return 0;
01880 }
01881 
01882 
01883 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
01884 {
01885     QDM2Context *s = avctx->priv_data;
01886 
01887     ff_rdft_end(&s->rdft_ctx);
01888 
01889     return 0;
01890 }
01891 
01892 
01893 static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
01894 {
01895     int ch, i;
01896     const int frame_size = (q->frame_size * q->channels);
01897 
01898     if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
01899         return;
01900 
01901     /* select input buffer */
01902     q->compressed_data = in;
01903     q->compressed_size = q->checksum_size;
01904 
01905 //  dump_context(q);
01906 
01907     /* copy old block, clear new block of output samples */
01908     memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
01909     memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
01910 
01911     /* decode block of QDM2 compressed data */
01912     if (q->sub_packet == 0) {
01913         q->has_errors = 0; // zero it for a new super block
01914         av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
01915         qdm2_decode_super_block(q);
01916     }
01917 
01918     /* parse subpackets */
01919     if (!q->has_errors) {
01920         if (q->sub_packet == 2)
01921             qdm2_decode_fft_packets(q);
01922 
01923         qdm2_fft_tone_synthesizer(q, q->sub_packet);
01924     }
01925 
01926     /* sound synthesis stage 1 (FFT) */
01927     for (ch = 0; ch < q->channels; ch++) {
01928         qdm2_calculate_fft(q, ch, q->sub_packet);
01929 
01930         if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
01931             SAMPLES_NEEDED_2("has errors, and C list is not empty")
01932             return;
01933         }
01934     }
01935 
01936     /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
01937     if (!q->has_errors && q->do_synth_filter)
01938         qdm2_synthesis_filter(q, q->sub_packet);
01939 
01940     q->sub_packet = (q->sub_packet + 1) % 16;
01941 
01942     /* clip and convert output float[] to 16bit signed samples */
01943     for (i = 0; i < frame_size; i++) {
01944         int value = (int)q->output_buffer[i];
01945 
01946         if (value > SOFTCLIP_THRESHOLD)
01947             value = (value >  HARDCLIP_THRESHOLD) ?  32767 :  softclip_table[ value - SOFTCLIP_THRESHOLD];
01948         else if (value < -SOFTCLIP_THRESHOLD)
01949             value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
01950 
01951         out[i] = value;
01952     }
01953 }
01954 
01955 
01956 static int qdm2_decode_frame(AVCodecContext *avctx,
01957             void *data, int *data_size,
01958             AVPacket *avpkt)
01959 {
01960     const uint8_t *buf = avpkt->data;
01961     int buf_size = avpkt->size;
01962     QDM2Context *s = avctx->priv_data;
01963 
01964     if(!buf)
01965         return 0;
01966     if(buf_size < s->checksum_size)
01967         return -1;
01968 
01969     *data_size = s->channels * s->frame_size * sizeof(int16_t);
01970 
01971     av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
01972        buf_size, buf, s->checksum_size, data, *data_size);
01973 
01974     qdm2_decode(s, buf, data);
01975 
01976     // reading only when next superblock found
01977     if (s->sub_packet == 0) {
01978         return s->checksum_size;
01979     }
01980 
01981     return 0;
01982 }
01983 
01984 AVCodec qdm2_decoder =
01985 {
01986     .name = "qdm2",
01987     .type = AVMEDIA_TYPE_AUDIO,
01988     .id = CODEC_ID_QDM2,
01989     .priv_data_size = sizeof(QDM2Context),
01990     .init = qdm2_decode_init,
01991     .close = qdm2_decode_close,
01992     .decode = qdm2_decode_frame,
01993     .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
01994 };

Generated on Wed Dec 21 2011 18:44:07 for FFmpeg by  doxygen 1.7.1