/*
  * QDM2 compatible decoder
  * Copyright (c) 2003 Ewald Snel
  * Copyright (c) 2005 Benjamin Larsson
  * Copyright (c) 2005 Alex Beregszaszi
  * Copyright (c) 2005 Roberto Togni
  *

  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or

  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either

  * version 2.1 of the License, or (at your option) any later version.

  *

  * FFmpeg is distributed in the hope that it will be useful,

  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public

  * License along with FFmpeg; if not, write to the Free Software

  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

  */
 
 /**

  * @file libavcodec/qdm2.c

  * QDM2 decoder
  * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni

  * The decoder is not perfect yet, there are still some distortions
  * especially on files encoded with 16 or 8 subbands.

  */
 
 #include <math.h>
 #include <stddef.h>
 #include <stdio.h>
 
 #define ALT_BITSTREAM_READER_LE
 #include "avcodec.h"
 #include "bitstream.h"
 #include "dsputil.h"
 #include "mpegaudio.h"
 
 #include "qdm2data.h"
 
 #undef NDEBUG
 #include <assert.h>
 
 
 #define SOFTCLIP_THRESHOLD 27600
 #define HARDCLIP_THRESHOLD 35716
 
 
 #define QDM2_LIST_ADD(list, size, packet) \
 do { \
       if (size > 0) { \
     list[size - 1].next = &list[size]; \
       } \
       list[size].packet = packet; \
       list[size].next = NULL; \
       size++; \
 } while(0)
 
 // Result is 8, 16 or 30
 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
 
 #define FIX_NOISE_IDX(noise_idx) \
   if ((noise_idx) >= 3840) \
     (noise_idx) -= 3840; \
 
 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
 
 #define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
 
 #define SAMPLES_NEEDED \
      av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
 
 #define SAMPLES_NEEDED_2(why) \
      av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
 

 #define QDM2_MAX_FRAME_SIZE 512

 
 typedef int8_t sb_int8_array[2][30][64];
 
 /**
  * Subpacket
  */
 typedef struct {
     int type;            ///< subpacket type
     unsigned int size;   ///< subpacket size
     const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
 } QDM2SubPacket;
 
 /**

  * A node in the subpacket list

  */

 typedef struct QDM2SubPNode {

     QDM2SubPacket *packet;      ///< packet

     struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node

 } QDM2SubPNode;
 
 typedef struct {

     float re;
     float im;
 } QDM2Complex;
 
 typedef struct {

     float level;

     QDM2Complex *complex;

     const float *table;

     int   phase;
     int   phase_shift;
     int   duration;
     short time_index;
     short cutoff;
 } FFTTone;
 
 typedef struct {
     int16_t sub_packet;
     uint8_t channel;
     int16_t offset;
     int16_t exp;
     uint8_t phase;
 } FFTCoefficient;
 
 typedef struct {

     DECLARE_ALIGNED_16(QDM2Complex, complex[MPA_MAX_CHANNELS][256]);

 } QDM2FFT;
 
 /**
  * QDM2 decoder context
  */
 typedef struct {
     /// Parameters from codec header, do not change during playback
     int nb_channels;         ///< number of channels
     int channels;            ///< number of channels
     int group_size;          ///< size of frame group (16 frames per group)
     int fft_size;            ///< size of FFT, in complex numbers
     int checksum_size;       ///< size of data block, used also for checksum
 
     /// Parameters built from header parameters, do not change during playback
     int group_order;         ///< order of frame group
     int fft_order;           ///< order of FFT (actually fftorder+1)
     int fft_frame_size;      ///< size of fft frame, in components (1 comples = re + im)
     int frame_size;          ///< size of data frame
     int frequency_range;
     int sub_sampling;        ///< subsampling: 0=25%, 1=50%, 2=100% */
     int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
     int cm_table_select;     ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
 
     /// Packets and packet lists
     QDM2SubPacket sub_packets[16];      ///< the packets themselves
     QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
     QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
     int sub_packets_B;                  ///< number of packets on 'B' list
     QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
     QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
 
     /// FFT and tones
     FFTTone fft_tones[1000];
     int fft_tone_start;
     int fft_tone_end;
     FFTCoefficient fft_coefs[1000];
     int fft_coefs_index;
     int fft_coefs_min_index[5];
     int fft_coefs_max_index[5];
     int fft_level_exp[6];

     RDFTContext rdft_ctx;

     QDM2FFT fft;
 
     /// I/O data

     const uint8_t *compressed_data;

     int compressed_size;

     float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];

 
     /// Synthesis filter

     DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512*2]);

     int synth_buf_offset[MPA_MAX_CHANNELS];

     DECLARE_ALIGNED_16(int32_t, sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT]);

 
     /// Mixed temporary data used in decoding
     float tone_level[MPA_MAX_CHANNELS][30][64];
     int8_t coding_method[MPA_MAX_CHANNELS][30][64];
     int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
     int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
     int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
     int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
     int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
     int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
     int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
 
     // Flags

     int has_errors;         ///< packet has errors

     int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
     int do_synth_filter;    ///< used to perform or skip synthesis filter
 
     int sub_packet;

     int noise_idx; ///< index for dithering noise table

 } QDM2Context;
 
 
 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
 
 static VLC vlc_tab_level;
 static VLC vlc_tab_diff;
 static VLC vlc_tab_run;
 static VLC fft_level_exp_alt_vlc;
 static VLC fft_level_exp_vlc;
 static VLC fft_stereo_exp_vlc;
 static VLC fft_stereo_phase_vlc;
 static VLC vlc_tab_tone_level_idx_hi1;
 static VLC vlc_tab_tone_level_idx_mid;
 static VLC vlc_tab_tone_level_idx_hi2;
 static VLC vlc_tab_type30;
 static VLC vlc_tab_type34;
 static VLC vlc_tab_fft_tone_offset[5];
 
 static uint16_t softclip_table[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1];
 static float noise_table[4096];
 static uint8_t random_dequant_index[256][5];
 static uint8_t random_dequant_type24[128][3];
 static float noise_samples[128];
 

 static DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]);

 static av_cold void softclip_table_init(void) {

     int i;
     double dfl = SOFTCLIP_THRESHOLD - 32767;
     float delta = 1.0 / -dfl;
     for (i = 0; i < HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1; i++)
         softclip_table[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF);
 }
 
 
 // random generated table

 static av_cold void rnd_table_init(void) {

     int i,j;
     uint32_t ldw,hdw;
     uint64_t tmp64_1;
     uint64_t random_seed = 0;
     float delta = 1.0 / 16384.0;
     for(i = 0; i < 4096 ;i++) {
         random_seed = random_seed * 214013 + 2531011;
         noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3;
     }
 
     for (i = 0; i < 256 ;i++) {
         random_seed = 81;
         ldw = i;
         for (j = 0; j < 5 ;j++) {
             random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
             ldw = (uint32_t)ldw % (uint32_t)random_seed;
             tmp64_1 = (random_seed * 0x55555556);
             hdw = (uint32_t)(tmp64_1 >> 32);
             random_seed = (uint64_t)(hdw + (ldw >> 31));
         }
     }
     for (i = 0; i < 128 ;i++) {
         random_seed = 25;
         ldw = i;
         for (j = 0; j < 3 ;j++) {
             random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
             ldw = (uint32_t)ldw % (uint32_t)random_seed;
             tmp64_1 = (random_seed * 0x66666667);
             hdw = (uint32_t)(tmp64_1 >> 33);
             random_seed = hdw + (ldw >> 31);
         }
     }
 }
 
 

 static av_cold void init_noise_samples(void) {

     int i;
     int random_seed = 0;
     float delta = 1.0 / 16384.0;
     for (i = 0; i < 128;i++) {
         random_seed = random_seed * 214013 + 2531011;
         noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0);
     }
 }
 
 

 static av_cold void qdm2_init_vlc(void)

 {
     init_vlc (&vlc_tab_level, 8, 24,
         vlc_tab_level_huffbits, 1, 1,
         vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
 
     init_vlc (&vlc_tab_diff, 8, 37,
         vlc_tab_diff_huffbits, 1, 1,
         vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
 
     init_vlc (&vlc_tab_run, 5, 6,
         vlc_tab_run_huffbits, 1, 1,
         vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
 
     init_vlc (&fft_level_exp_alt_vlc, 8, 28,
         fft_level_exp_alt_huffbits, 1, 1,
         fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
 
     init_vlc (&fft_level_exp_vlc, 8, 20,
         fft_level_exp_huffbits, 1, 1,
         fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
 
     init_vlc (&fft_stereo_exp_vlc, 6, 7,
         fft_stereo_exp_huffbits, 1, 1,
         fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
 
     init_vlc (&fft_stereo_phase_vlc, 6, 9,
         fft_stereo_phase_huffbits, 1, 1,
         fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
 
     init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
         vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
         vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
 
     init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
         vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
         vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
 
     init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
         vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
         vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
 
     init_vlc (&vlc_tab_type30, 6, 9,
         vlc_tab_type30_huffbits, 1, 1,
         vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
 
     init_vlc (&vlc_tab_type34, 5, 10,
         vlc_tab_type34_huffbits, 1, 1,
         vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
 
     init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
         vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
         vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
 
     init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
         vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
         vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
 
     init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
         vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
         vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
 
     init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
         vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
         vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
 
     init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
         vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
         vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
 }
 
 
 /* for floating point to fixed point conversion */

 static const float f2i_scale = (float) (1 << (FRAC_BITS - 15));

 
 
 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
 {
     int value;
 
     value = get_vlc2(gb, vlc->table, vlc->bits, depth);
 
     /* stage-2, 3 bits exponent escape sequence */
     if (value-- == 0)
         value = get_bits (gb, get_bits (gb, 3) + 1);
 
     /* stage-3, optional */
     if (flag) {
         int tmp = vlc_stage3_values[value];
 
         if ((value & ~3) > 0)
             tmp += get_bits (gb, (value >> 2));
         value = tmp;
     }
 
     return value;
 }
 
 
 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
 {
     int value = qdm2_get_vlc (gb, vlc, 0, depth);
 
     return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
 }
 
 
 /**
  * QDM2 checksum
  *
  * @param data      pointer to data to be checksum'ed
  * @param length    data length
  * @param value     checksum value
  *

  * @return          0 if checksum is OK

  */

 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {

     int i;
 
     for (i=0; i < length; i++)
         value -= data[i];
 
     return (uint16_t)(value & 0xffff);
 }
 
 
 /**

  * Fills a QDM2SubPacket structure with packet type, size, and data pointer.

  *
  * @param gb            bitreader context
  * @param sub_packet    packet under analysis
  */
 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
 {
     sub_packet->type = get_bits (gb, 8);
 
     if (sub_packet->type == 0) {
         sub_packet->size = 0;
         sub_packet->data = NULL;
     } else {
         sub_packet->size = get_bits (gb, 8);
 
       if (sub_packet->type & 0x80) {
           sub_packet->size <<= 8;
           sub_packet->size  |= get_bits (gb, 8);
           sub_packet->type  &= 0x7f;
       }
 
       if (sub_packet->type == 0x7f)
           sub_packet->type |= (get_bits (gb, 8) << 8);
 
       sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
     }
 

     av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",

         sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
 }
 
 
 /**

  * Return node pointer to first packet of requested type in list.

  *

  * @param list    list of subpackets to be scanned

  * @param type    type of searched subpacket
  * @return        node pointer for subpacket if found, else NULL
  */
 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
 {
     while (list != NULL && list->packet != NULL) {
         if (list->packet->type == type)
             return list;
         list = list->next;
     }
     return NULL;
 }
 
 
 /**

  * Replaces 8 elements with their average value.
  * Called by qdm2_decode_superblock before starting subblock decoding.

  *
  * @param q       context
  */
 static void average_quantized_coeffs (QDM2Context *q)
 {
     int i, j, n, ch, sum;
 
     n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
 
     for (ch = 0; ch < q->nb_channels; ch++)
         for (i = 0; i < n; i++) {
             sum = 0;
 
             for (j = 0; j < 8; j++)
                 sum += q->quantized_coeffs[ch][i][j];
 
             sum /= 8;
             if (sum > 0)
                 sum--;
 
             for (j=0; j < 8; j++)
                 q->quantized_coeffs[ch][i][j] = sum;
         }
 }
 
 
 /**

  * Build subband samples with noise weighted by q->tone_level.
  * Called by synthfilt_build_sb_samples.

  *
  * @param q     context
  * @param sb    subband index
  */
 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
 {
     int ch, j;
 
     FIX_NOISE_IDX(q->noise_idx);
 
     if (!q->nb_channels)
         return;
 
     for (ch = 0; ch < q->nb_channels; ch++)
         for (j = 0; j < 64; j++) {
             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);
             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);
         }
 }
 
 
 /**

  * Called while processing data from subpackets 11 and 12.
  * Used after making changes to coding_method array.

  *
  * @param sb               subband index
  * @param channels         number of channels
  * @param coding_method    q->coding_method[0][0][0]
  */

 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)

 {
     int j,k;
     int ch;
     int run, case_val;
     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};
 
     for (ch = 0; ch < channels; ch++) {
         for (j = 0; j < 64; ) {
             if((coding_method[ch][sb][j] - 8) > 22) {
                 run = 1;
                 case_val = 8;
             } else {

                 switch (switchtable[coding_method[ch][sb][j]-8]) {

                     case 0: run = 10; case_val = 10; break;
                     case 1: run = 1; case_val = 16; break;
                     case 2: run = 5; case_val = 24; break;
                     case 3: run = 3; case_val = 30; break;
                     case 4: run = 1; case_val = 30; break;
                     case 5: run = 1; case_val = 8; break;
                     default: run = 1; case_val = 8; break;
                 }
             }
             for (k = 0; k < run; k++)
                 if (j + k < 128)
                     if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
                         if (k > 0) {
                            SAMPLES_NEEDED
                             //not debugged, almost never used
                             memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
                             memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
                         }
             j += run;
         }
     }
 }
 
 
 /**
  * Related to synthesis filter
  * Called by process_subpacket_10
  *
  * @param q       context
  * @param flag    1 if called after getting data from subpacket 10, 0 if no subpacket 10
  */
 static void fill_tone_level_array (QDM2Context *q, int flag)
 {
     int i, sb, ch, sb_used;
     int tmp, tab;
 
     // This should never happen
     if (q->nb_channels <= 0)
         return;
 
     for (ch = 0; ch < q->nb_channels; ch++)
         for (sb = 0; sb < 30; sb++)
             for (i = 0; i < 8; i++) {
                 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
                     tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
                           q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
                 else
                     tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
                 if(tmp < 0)
                     tmp += 0xff;
                 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
             }
 
     sb_used = QDM2_SB_USED(q->sub_sampling);
 
     if ((q->superblocktype_2_3 != 0) && !flag) {
         for (sb = 0; sb < sb_used; sb++)
             for (ch = 0; ch < q->nb_channels; ch++)
                 for (i = 0; i < 64; i++) {
                     q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
                     if (q->tone_level_idx[ch][sb][i] < 0)
                         q->tone_level[ch][sb][i] = 0;
                     else
                         q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
                 }
     } else {
         tab = q->superblocktype_2_3 ? 0 : 1;
         for (sb = 0; sb < sb_used; sb++) {
             if ((sb >= 4) && (sb <= 23)) {
                 for (ch = 0; ch < q->nb_channels; ch++)
                     for (i = 0; i < 64; i++) {
                         tmp = q->tone_level_idx_base[ch][sb][i / 8] -
                               q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
                               q->tone_level_idx_mid[ch][sb - 4][i / 8] -
                               q->tone_level_idx_hi2[ch][sb - 4];
                         q->tone_level_idx[ch][sb][i] = tmp & 0xff;
                         if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
                             q->tone_level[ch][sb][i] = 0;
                         else
                             q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
                 }
             } else {
                 if (sb > 4) {
                     for (ch = 0; ch < q->nb_channels; ch++)
                         for (i = 0; i < 64; i++) {
                             tmp = q->tone_level_idx_base[ch][sb][i / 8] -
                                   q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
                                   q->tone_level_idx_hi2[ch][sb - 4];
                             q->tone_level_idx[ch][sb][i] = tmp & 0xff;
                             if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
                                 q->tone_level[ch][sb][i] = 0;
                             else
                                 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
                     }
                 } else {
                     for (ch = 0; ch < q->nb_channels; ch++)
                         for (i = 0; i < 64; i++) {
                             tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
                             if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
                                 q->tone_level[ch][sb][i] = 0;
                             else
                                 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
                         }
                 }
             }
         }
     }
 
     return;
 }
 
 
 /**
  * Related to synthesis filter
  * Called by process_subpacket_11
  * c is built with data from subpacket 11
  * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
  *

  * @param tone_level_idx

  * @param tone_level_idx_temp
  * @param coding_method        q->coding_method[0][0][0]
  * @param nb_channels          number of channels
  * @param c                    coming from subpacket 11, passed as 8*c
  * @param superblocktype_2_3   flag based on superblock packet type
  * @param cm_table_select      q->cm_table_select
  */
 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
                 sb_int8_array coding_method, int nb_channels,
                 int c, int superblocktype_2_3, int cm_table_select)
 {
     int ch, sb, j;
     int tmp, acc, esp_40, comp;
     int add1, add2, add3, add4;
     int64_t multres;
 
     // This should never happen
     if (nb_channels <= 0)
         return;
 
     if (!superblocktype_2_3) {
         /* This case is untested, no samples available */
         SAMPLES_NEEDED
         for (ch = 0; ch < nb_channels; ch++)
             for (sb = 0; sb < 30; sb++) {

                 for (j = 1; j < 63; j++) {  // The loop only iterates to 63 so the code doesn't overflow the buffer

                     add1 = tone_level_idx[ch][sb][j] - 10;
                     if (add1 < 0)
                         add1 = 0;
                     add2 = add3 = add4 = 0;
                     if (sb > 1) {
                         add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
                         if (add2 < 0)
                             add2 = 0;
                     }
                     if (sb > 0) {
                         add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
                         if (add3 < 0)
                             add3 = 0;
                     }
                     if (sb < 29) {
                         add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
                         if (add4 < 0)
                             add4 = 0;
                     }
                     tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
                     if (tmp < 0)
                         tmp = 0;
                     tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
                 }
                 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
             }
             acc = 0;
             for (ch = 0; ch < nb_channels; ch++)
                 for (sb = 0; sb < 30; sb++)
                     for (j = 0; j < 64; j++)
                         acc += tone_level_idx_temp[ch][sb][j];
             if (acc)
                 tmp = c * 256 / (acc & 0xffff);
             multres = 0x66666667 * (acc * 10);
             esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
             for (ch = 0;  ch < nb_channels; ch++)
                 for (sb = 0; sb < 30; sb++)
                     for (j = 0; j < 64; j++) {
                         comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
                         if (comp < 0)
                             comp += 0xff;
                         comp /= 256; // signed shift
                         switch(sb) {
                             case 0:
                                 if (comp < 30)
                                     comp = 30;
                                 comp += 15;
                                 break;
                             case 1:
                                 if (comp < 24)
                                     comp = 24;
                                 comp += 10;
                                 break;
                             case 2:
                             case 3:
                             case 4:
                                 if (comp < 16)
                                     comp = 16;
                         }
                         if (comp <= 5)
                             tmp = 0;
                         else if (comp <= 10)
                             tmp = 10;
                         else if (comp <= 16)
                             tmp = 16;
                         else if (comp <= 24)
                             tmp = -1;
                         else
                             tmp = 0;
                         coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
                     }
             for (sb = 0; sb < 30; sb++)
                 fix_coding_method_array(sb, nb_channels, coding_method);
             for (ch = 0; ch < nb_channels; ch++)
                 for (sb = 0; sb < 30; sb++)
                     for (j = 0; j < 64; j++)
                         if (sb >= 10) {
                             if (coding_method[ch][sb][j] < 10)
                                 coding_method[ch][sb][j] = 10;
                         } else {
                             if (sb >= 2) {
                                 if (coding_method[ch][sb][j] < 16)
                                     coding_method[ch][sb][j] = 16;
                             } else {
                                 if (coding_method[ch][sb][j] < 30)
                                     coding_method[ch][sb][j] = 30;
                             }
                         }
     } else { // superblocktype_2_3 != 0
         for (ch = 0; ch < nb_channels; ch++)
             for (sb = 0; sb < 30; sb++)
                 for (j = 0; j < 64; j++)
                     coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
     }
 
     return;
 }
 
 
 /**
  *
  * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
  * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
  *
  * @param q         context
  * @param gb        bitreader context

  * @param length    packet length in bits

  * @param sb_min    lower subband processed (sb_min included)
  * @param sb_max    higher subband processed (sb_max excluded)
  */
 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
 {
     int sb, j, k, n, ch, run, channels;
     int joined_stereo, zero_encoding, chs;
     int type34_first;
     float type34_div = 0;
     float type34_predictor;
     float samples[10], sign_bits[16];
 
     if (length == 0) {
         // If no data use noise
         for (sb=sb_min; sb < sb_max; sb++)
             build_sb_samples_from_noise (q, sb);
 
         return;
     }
 
     for (sb = sb_min; sb < sb_max; sb++) {
         FIX_NOISE_IDX(q->noise_idx);
 
         channels = q->nb_channels;
 
         if (q->nb_channels <= 1 || sb < 12)
             joined_stereo = 0;
         else if (sb >= 24)
             joined_stereo = 1;
         else
             joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
 
         if (joined_stereo) {
             if (BITS_LEFT(length,gb) >= 16)
                 for (j = 0; j < 16; j++)
                     sign_bits[j] = get_bits1 (gb);
 
             for (j = 0; j < 64; j++)
                 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
                     q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
 
             fix_coding_method_array(sb, q->nb_channels, q->coding_method);
             channels = 1;
         }
 
         for (ch = 0; ch < channels; ch++) {
             zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
             type34_predictor = 0.0;
             type34_first = 1;
 
             for (j = 0; j < 128; ) {
                 switch (q->coding_method[ch][sb][j / 2]) {
                     case 8:
                         if (BITS_LEFT(length,gb) >= 10) {
                             if (zero_encoding) {
                                 for (k = 0; k < 5; k++) {
                                     if ((j + 2 * k) >= 128)
                                         break;
                                     samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
                                 }
                             } else {
                                 n = get_bits(gb, 8);
                                 for (k = 0; k < 5; k++)
                                     samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
                             }
                             for (k = 0; k < 5; k++)
                                 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
                         } else {
                             for (k = 0; k < 10; k++)
                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
                         }
                         run = 10;
                         break;
 
                     case 10:
                         if (BITS_LEFT(length,gb) >= 1) {
                             float f = 0.81;
 
                             if (get_bits1(gb))
                                 f = -f;
                             f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
                             samples[0] = f;
                         } else {
                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
                         }
                         run = 1;
                         break;
 
                     case 16:
                         if (BITS_LEFT(length,gb) >= 10) {
                             if (zero_encoding) {
                                 for (k = 0; k < 5; k++) {
                                     if ((j + k) >= 128)
                                         break;
                                     samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
                                 }
                             } else {
                                 n = get_bits (gb, 8);
                                 for (k = 0; k < 5; k++)
                                     samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
                             }
                         } else {
                             for (k = 0; k < 5; k++)
                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
                         }
                         run = 5;
                         break;
 
                     case 24:
                         if (BITS_LEFT(length,gb) >= 7) {
                             n = get_bits(gb, 7);
                             for (k = 0; k < 3; k++)
                                 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
                         } else {
                             for (k = 0; k < 3; k++)
                                 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
                         }
                         run = 3;
                         break;
 
                     case 30:
                         if (BITS_LEFT(length,gb) >= 4)
                             samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
                         else
                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);

                         run = 1;
                         break;
 
                     case 34:
                         if (BITS_LEFT(length,gb) >= 7) {
                             if (type34_first) {
                                 type34_div = (float)(1 << get_bits(gb, 2));
                                 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
                                 type34_predictor = samples[0];
                                 type34_first = 0;
                             } else {
                                 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
                                 type34_predictor = samples[0];
                             }
                         } else {
                             samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
                         }
                         run = 1;
                         break;
 
                     default:
                         samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
                         run = 1;
                         break;
                 }
 
                 if (joined_stereo) {
                     float tmp[10][MPA_MAX_CHANNELS];
 
                     for (k = 0; k < run; k++) {
                         tmp[k][0] = samples[k];
                         tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
                     }
                     for (chs = 0; chs < q->nb_channels; chs++)
                         for (k = 0; k < run; k++)
                             if ((j + k) < 128)
                                 q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
                 } else {
                     for (k = 0; k < run; k++)
                         if ((j + k) < 128)
                             q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
                 }
 
                 j += run;
             } // j loop
         } // channel loop
     } // subband loop
 }
 
 
 /**

  * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).

  * This is similar to process_subpacket_9, but for a single channel and for element [0]

  * same VLC tables as process_subpacket_9 are used.

  *
  * @param q         context
  * @param quantized_coeffs    pointer to quantized_coeffs[ch][0]
  * @param gb        bitreader context

  * @param length    packet length in bits

  */
 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
 {
     int i, k, run, level, diff;
 
     if (BITS_LEFT(length,gb) < 16)
         return;
     level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
 
     quantized_coeffs[0] = level;
 
     for (i = 0; i < 7; ) {
         if (BITS_LEFT(length,gb) < 16)
             break;
         run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
 
         if (BITS_LEFT(length,gb) < 16)
             break;
         diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);

         for (k = 1; k <= run; k++)
             quantized_coeffs[i + k] = (level + ((k * diff) / run));

         level += diff;
         i += run;
     }
 }
 
 
 /**
  * Related to synthesis filter, process data from packet 10
  * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
  * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
  *
  * @param q         context
  * @param gb        bitreader context

  * @param length    packet length in bits

  */
 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
 {
     int sb, j, k, n, ch;
 
     for (ch = 0; ch < q->nb_channels; ch++) {
         init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
 
         if (BITS_LEFT(length,gb) < 16) {
             memset(q->quantized_coeffs[ch][0], 0, 8);
             break;
         }
     }
 
     n = q->sub_sampling + 1;
 
     for (sb = 0; sb < n; sb++)
         for (ch = 0; ch < q->nb_channels; ch++)
             for (j = 0; j < 8; j++) {
                 if (BITS_LEFT(length,gb) < 1)
                     break;
                 if (get_bits1(gb)) {
                     for (k=0; k < 8; k++) {
                         if (BITS_LEFT(length,gb) < 16)
                             break;
                         q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
                     }
                 } else {
                     for (k=0; k < 8; k++)
                         q->tone_level_idx_hi1[ch][sb][j][k] = 0;
                 }
             }
 
     n = QDM2_SB_USED(q->sub_sampling) - 4;
 
     for (sb = 0; sb < n; sb++)
         for (ch = 0; ch < q->nb_channels; ch++) {
             if (BITS_LEFT(length,gb) < 16)
                 break;
             q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
             if (sb > 19)
                 q->tone_level_idx_hi2[ch][sb] -= 16;
             else
                 for (j = 0; j < 8; j++)
                     q->tone_level_idx_mid[ch][sb][j] = -16;
         }
 
     n = QDM2_SB_USED(q->sub_sampling) - 5;
 
     for (sb = 0; sb < n; sb++)
         for (ch = 0; ch < q->nb_channels; ch++)
             for (j = 0; j < 8; j++) {
                 if (BITS_LEFT(length,gb) < 16)
                     break;
                 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
             }
 }
 
 /**
  * Process subpacket 9, init quantized_coeffs with data from it
  *
  * @param q       context
  * @param node    pointer to node with packet
  */
 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
 {
     GetBitContext gb;
     int i, j, k, n, ch, run, level, diff;
 

     init_get_bits(&gb, node->packet->data, node->packet->size*8);

 
     n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
 
     for (i = 1; i < n; i++)
         for (ch=0; ch < q->nb_channels; ch++) {
             level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
             q->quantized_coeffs[ch][i][0] = level;
 
             for (j = 0; j < (8 - 1); ) {
                 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
                 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
 
                 for (k = 1; k <= run; k++)
                     q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
 
                 level += diff;
                 j += run;
             }
         }
 
     for (ch = 0; ch < q->nb_channels; ch++)
         for (i = 0; i < 8; i++)
             q->quantized_coeffs[ch][0][i] = 0;
 }
 
 
 /**
  * Process subpacket 10 if not null, else
  *
  * @param q         context
  * @param node      pointer to node with packet

  * @param length    packet length in bits

  */
 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
 {
     GetBitContext gb;
 

     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));

 
     if (length != 0) {
         init_tone_level_dequantization(q, &gb, length);
         fill_tone_level_array(q, 1);
     } else {
         fill_tone_level_array(q, 0);
     }
 }
 
 
 /**
  * Process subpacket 11
  *
  * @param q         context
  * @param node      pointer to node with packet
  * @param length    packet length in bit
  */
 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
 {
     GetBitContext gb;
 

     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));

     if (length >= 32) {
         int c = get_bits (&gb, 13);
 
         if (c > 3)
             fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
                                       q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
     }
 
     synthfilt_build_sb_samples(q, &gb, length, 0, 8);
 }
 
 
 /**
  * Process subpacket 12
  *
  * @param q         context
  * @param node      pointer to node with packet

  * @param length    packet length in bits

  */
 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
 {
     GetBitContext gb;
 

     init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));

     synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
 }
 
 /*
  * Process new subpackets for synthesis filter
  *
  * @param q       context
  * @param list    list with synthesis filter packets (list D)
  */
 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
 {
     QDM2SubPNode *nodes[4];
 
     nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
     if (nodes[0] != NULL)
         process_subpacket_9(q, nodes[0]);
 
     nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
     if (nodes[1] != NULL)
         process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
     else
         process_subpacket_10(q, NULL, 0);
 
     nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
     if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
         process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
     else
         process_subpacket_11(q, NULL, 0);
 
     nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
     if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
         process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
     else
         process_subpacket_12(q, NULL, 0);
 }
 
 
 /*

  * Decode superblock, fill packet lists.

  *
  * @param q    context
  */
 static void qdm2_decode_super_block (QDM2Context *q)
 {
     GetBitContext gb;
     QDM2SubPacket header, *packet;
     int i, packet_bytes, sub_packet_size, sub_packets_D;
     unsigned int next_index = 0;
 
     memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
     memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
     memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
 
     q->sub_packets_B = 0;
     sub_packets_D = 0;
 
     average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
 

     init_get_bits(&gb, q->compressed_data, q->compressed_size*8);

     qdm2_decode_sub_packet_header(&gb, &header);
 
     if (header.type < 2 || header.type >= 8) {
         q->has_errors = 1;
         av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
         return;
     }
 
     q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
     packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
 

     init_get_bits(&gb, header.data, header.size*8);

 
     if (header.type == 2 || header.type == 4 || header.type == 5) {
         int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8);
 
         csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
 
         if (csum != 0) {
             q->has_errors = 1;
             av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
             return;
         }
     }
 
     q->sub_packet_list_B[0].packet = NULL;
     q->sub_packet_list_D[0].packet = NULL;
 
     for (i = 0; i < 6; i++)
         if (--q->fft_level_exp[i] < 0)
             q->fft_level_exp[i] = 0;
 
     for (i = 0; packet_bytes > 0; i++) {
         int j;
 
         q->sub_packet_list_A[i].next = NULL;
 
         if (i > 0) {
             q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
 
             /* seek to next block */

             init_get_bits(&gb, header.data, header.size*8);

             skip_bits(&gb, next_index*8);
 
             if (next_index >= header.size)
                 break;
         }
 

         /* decode subpacket */

         packet = &q->sub_packets[i];
         qdm2_decode_sub_packet_header(&gb, packet);
         next_index = packet->size + get_bits_count(&gb) / 8;
         sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
 
         if (packet->type == 0)
             break;
 
         if (sub_packet_size > packet_bytes) {
             if (packet->type != 10 && packet->type != 11 && packet->type != 12)
                 break;
             packet->size += packet_bytes - sub_packet_size;
         }
 
         packet_bytes -= sub_packet_size;
 

         /* add subpacket to 'all subpackets' list */

         q->sub_packet_list_A[i].packet = packet;
 

         /* add subpacket to related list */

         if (packet->type == 8) {
             SAMPLES_NEEDED_2("packet type 8");
             return;
         } else if (packet->type >= 9 && packet->type <= 12) {
             /* packets for MPEG Audio like Synthesis Filter */
             QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
         } else if (packet->type == 13) {
             for (j = 0; j < 6; j++)
                 q->fft_level_exp[j] = get_bits(&gb, 6);
         } else if (packet->type == 14) {
             for (j = 0; j < 6; j++)
                 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
         } else if (packet->type == 15) {
             SAMPLES_NEEDED_2("packet type 15")
             return;
         } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
             /* packets for FFT */
             QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
         }
     } // Packet bytes loop
 
 /* **************************************************************** */
     if (q->sub_packet_list_D[0].packet != NULL) {
         process_synthesis_subpackets(q, q->sub_packet_list_D);
         q->do_synth_filter = 1;
     } else if (q->do_synth_filter) {
         process_subpacket_10(q, NULL, 0);
         process_subpacket_11(q, NULL, 0);
         process_subpacket_12(q, NULL, 0);
     }
 /* **************************************************************** */
 }
 
 
 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
                        int offset, int duration, int channel,
                        int exp, int phase)
 {
     if (q->fft_coefs_min_index[duration] < 0)
         q->fft_coefs_min_index[duration] = q->fft_coefs_index;
 
     q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
     q->fft_coefs[q->fft_coefs_index].channel = channel;
     q->fft_coefs[q->fft_coefs_index].offset = offset;
     q->fft_coefs[q->fft_coefs_index].exp = exp;
     q->fft_coefs[q->fft_coefs_index].phase = phase;
     q->fft_coefs_index++;
 }
 
 
 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
 {
     int channel, stereo, phase, exp;
     int local_int_4,  local_int_8,  stereo_phase,  local_int_10;
     int local_int_14, stereo_exp, local_int_20, local_int_28;
     int n, offset;
 
     local_int_4 = 0;
     local_int_28 = 0;
     local_int_20 = 2;
     local_int_8 = (4 - duration);
     local_int_10 = 1 << (q->group_order - duration - 1);
     offset = 1;
 

     while (get_bits_left(gb)>0) {

         if (q->superblocktype_2_3) {
             while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
                 offset = 1;
                 if (n == 0) {
                     local_int_4 += local_int_10;
                     local_int_28 += (1 << local_int_8);
                 } else {
                     local_int_4 += 8*local_int_10;
                     local_int_28 += (8 << local_int_8);
                 }
             }
             offset += (n - 2);
         } else {
             offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
             while (offset >= (local_int_10 - 1)) {
                 offset += (1 - (local_int_10 - 1));
                 local_int_4  += local_int_10;
                 local_int_28 += (1 << local_int_8);
             }
         }
 
         if (local_int_4 >= q->group_size)
             return;
 
         local_int_14 = (offset >> local_int_8);

         if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
             return;

 
         if (q->nb_channels > 1) {
             channel = get_bits1(gb);
             stereo = get_bits1(gb);
         } else {
             channel = 0;
             stereo = 0;
         }
 
         exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
         exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
         exp = (exp < 0) ? 0 : exp;
 
         phase = get_bits(gb, 3);
         stereo_exp = 0;
         stereo_phase = 0;
 
         if (stereo) {
             stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
             stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
             if (stereo_phase < 0)
                 stereo_phase += 8;
         }
 
         if (q->frequency_range > (local_int_14 + 1)) {
             int sub_packet = (local_int_20 + local_int_28);
 
             qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
             if (stereo)
                 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
         }
 
         offset++;
     }
 }
 
 
 static void qdm2_decode_fft_packets (QDM2Context *q)
 {
     int i, j, min, max, value, type, unknown_flag;
     GetBitContext gb;
 
     if (q->sub_packet_list_B[0].packet == NULL)
         return;
 

     /* reset minimum indexes for FFT coefficients */

     q->fft_coefs_index = 0;
     for (i=0; i < 5; i++)
         q->fft_coefs_min_index[i] = -1;
 

     /* process subpackets ordered by type, largest type first */

     for (i = 0, max = 256; i < q->sub_packets_B; i++) {

         QDM2SubPacket *packet= NULL;

         /* find subpacket with largest type less than max */

         for (j = 0, min = 0; j < q->sub_packets_B; j++) {

             value = q->sub_packet_list_B[j].packet->type;
             if (value > min && value < max) {
                 min = value;
                 packet = q->sub_packet_list_B[j].packet;
             }
         }
 
         max = min;
 
         /* check for errors (?) */

         if (!packet)
             return;
 

         if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
             return;
 
         /* decode FFT tones */

         init_get_bits (&gb, packet->data, packet->size*8);

 
         if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
             unknown_flag = 1;
         else
             unknown_flag = 0;
 
         type = packet->type;
 
         if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
             int duration = q->sub_sampling + 5 - (type & 15);
 
             if (duration >= 0 && duration < 4)
                 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
         } else if (type == 31) {

             for (j=0; j < 4; j++)
                 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);

         } else if (type == 46) {

             for (j=0; j < 6; j++)
                 q->fft_level_exp[j] = get_bits(&gb, 6);
             for (j=0; j < 4; j++)
             qdm2_fft_decode_tones(q, j, &gb, unknown_flag);

         }
     } // Loop on B packets
 

     /* calculate maximum indexes for FFT coefficients */

     for (i = 0, j = -1; i < 5; i++)
         if (q->fft_coefs_min_index[i] >= 0) {
             if (j >= 0)
                 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
             j = i;
         }
     if (j >= 0)
         q->fft_coefs_max_index[j] = q->fft_coefs_index;
 }
 
 
 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
 {
    float level, f[6];
    int i;
    QDM2Complex c;
    const double iscale = 2.0*M_PI / 512.0;
 
     tone->phase += tone->phase_shift;
 
     /* calculate current level (maximum amplitude) of tone */
     level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
     c.im = level * sin(tone->phase*iscale);
     c.re = level * cos(tone->phase*iscale);
 
     /* generate FFT coefficients for tone */
     if (tone->duration >= 3 || tone->cutoff >= 3) {

         tone->complex[0].im += c.im;
         tone->complex[0].re += c.re;
         tone->complex[1].im -= c.im;
         tone->complex[1].re -= c.re;

     } else {
         f[1] = -tone->table[4];
         f[0] =  tone->table[3] - tone->table[0];
         f[2] =  1.0 - tone->table[2] - tone->table[3];
         f[3] =  tone->table[1] + tone->table[4] - 1.0;
         f[4] =  tone->table[0] - tone->table[1];
         f[5] =  tone->table[2];
         for (i = 0; i < 2; i++) {

             tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
             tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);

         }
         for (i = 0; i < 4; i++) {

             tone->complex[i].re += c.re * f[i+2];
             tone->complex[i].im += c.im * f[i+2];

         }
     }
 
     /* copy the tone if it has not yet died out */
     if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
       memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
       q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
     }
 }
 
 
 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
 {
     int i, j, ch;
     const double iscale = 0.25 * M_PI;
 
     for (ch = 0; ch < q->channels; ch++) {

         memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));

     }
 
 
     /* apply FFT tones with duration 4 (1 FFT period) */
     if (q->fft_coefs_min_index[4] >= 0)
         for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
             float level;
             QDM2Complex c;
 
             if (q->fft_coefs[i].sub_packet != sub_packet)
                 break;
 
             ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
             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];
 
             c.re = level * cos(q->fft_coefs[i].phase * iscale);
             c.im = level * sin(q->fft_coefs[i].phase * iscale);

             q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
             q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
             q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
             q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;

         }
 
     /* generate existing FFT tones */
     for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
         qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
         q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
     }
 
     /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
     for (i = 0; i < 4; i++)
         if (q->fft_coefs_min_index[i] >= 0) {
             for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
                 int offset, four_i;
                 FFTTone tone;
 
                 if (q->fft_coefs[j].sub_packet != sub_packet)
                     break;
 
                 four_i = (4 - i);
                 offset = q->fft_coefs[j].offset >> four_i;
                 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
 
                 if (offset < q->frequency_range) {
                     if (offset < 2)
                         tone.cutoff = offset;
                     else
                         tone.cutoff = (offset >= 60) ? 3 : 2;
 
                     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];

                     tone.complex = &q->fft.complex[ch][offset];

                     tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];

                     tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
                     tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
                     tone.duration = i;
                     tone.time_index = 0;
 
                     qdm2_fft_generate_tone(q, &tone);
                 }
             }
             q->fft_coefs_min_index[i] = j;
         }
 }
 
 
 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
 {

     const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
     int i;
     q->fft.complex[channel][0].re *= 2.0f;
     q->fft.complex[channel][0].im = 0.0f;
     ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);

     /* add samples to output buffer */
     for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)

         q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;

 }
 
 
 /**
  * @param q        context
  * @param index    subpacket number
  */
 static void qdm2_synthesis_filter (QDM2Context *q, int index)
 {
     OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
     int i, k, ch, sb_used, sub_sampling, dither_state = 0;
 
     /* copy sb_samples */
     sb_used = QDM2_SB_USED(q->sub_sampling);
 
     for (ch = 0; ch < q->channels; ch++)
         for (i = 0; i < 8; i++)
             for (k=sb_used; k < SBLIMIT; k++)
                 q->sb_samples[ch][(8 * index) + i][k] = 0;
 
     for (ch = 0; ch < q->nb_channels; ch++) {
         OUT_INT *samples_ptr = samples + ch;
 
         for (i = 0; i < 8; i++) {
             ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
                 mpa_window, &dither_state,
                 samples_ptr, q->nb_channels,
                 q->sb_samples[ch][(8 * index) + i]);
             samples_ptr += 32 * q->nb_channels;
         }
     }
 
     /* add samples to output buffer */
     sub_sampling = (4 >> q->sub_sampling);
 
     for (ch = 0; ch < q->channels; ch++)
         for (i = 0; i < q->frame_size; i++)
             q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
 }
 
 
 /**
  * Init static data (does not depend on specific file)
  *
  * @param q    context
  */

 static av_cold void qdm2_init(QDM2Context *q) {

     static int initialized = 0;

     if (initialized != 0)

         return;

     initialized = 1;

 
     qdm2_init_vlc();
     ff_mpa_synth_init(mpa_window);
     softclip_table_init();
     rnd_table_init();
     init_noise_samples();
 
     av_log(NULL, AV_LOG_DEBUG, "init done\n");
 }
 
 
 #if 0
 static void dump_context(QDM2Context *q)
 {
     int i;
 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
     PRINT("compressed_data",q->compressed_data);
     PRINT("compressed_size",q->compressed_size);
     PRINT("frame_size",q->frame_size);
     PRINT("checksum_size",q->checksum_size);
     PRINT("channels",q->channels);
     PRINT("nb_channels",q->nb_channels);
     PRINT("fft_frame_size",q->fft_frame_size);
     PRINT("fft_size",q->fft_size);
     PRINT("sub_sampling",q->sub_sampling);
     PRINT("fft_order",q->fft_order);
     PRINT("group_order",q->group_order);
     PRINT("group_size",q->group_size);
     PRINT("sub_packet",q->sub_packet);
     PRINT("frequency_range",q->frequency_range);
     PRINT("has_errors",q->has_errors);
     PRINT("fft_tone_end",q->fft_tone_end);
     PRINT("fft_tone_start",q->fft_tone_start);
     PRINT("fft_coefs_index",q->fft_coefs_index);
     PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
     PRINT("cm_table_select",q->cm_table_select);
     PRINT("noise_idx",q->noise_idx);
 
     for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
     {
     FFTTone *t = &q->fft_tones[i];

     av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
     av_log(NULL,AV_LOG_DEBUG,"  level = %f\n", t->level);
 //  PRINT(" level", t->level);
     PRINT(" phase", t->phase);
     PRINT(" phase_shift", t->phase_shift);
     PRINT(" duration", t->duration);
     PRINT(" samples_im", t->samples_im);
     PRINT(" samples_re", t->samples_re);
     PRINT(" table", t->table);
     }
 
 }
 #endif
 
 
 /**
  * Init parameters from codec extradata
  */

 static av_cold int qdm2_decode_init(AVCodecContext *avctx)

 {
     QDM2Context *s = avctx->priv_data;
     uint8_t *extradata;
     int extradata_size;
     int tmp_val, tmp, size;

     /* extradata parsing

     Structure:
     wave {
         frma (QDM2)
         QDCA
         QDCP
     }

     32  size (including this field)
     32  tag (=frma)
     32  type (=QDM2 or QDMC)

     32  size (including this field, in bytes)
     32  tag (=QDCA) // maybe mandatory parameters
     32  unknown (=1)
     32  channels (=2)
     32  samplerate (=44100)
     32  bitrate (=96000)
     32  block size (=4096)
     32  frame size (=256) (for one channel)
     32  packet size (=1300)

     32  size (including this field, in bytes)
     32  tag (=QDCP) // maybe some tuneable parameters
     32  float1 (=1.0)
     32  zero ?
     32  float2 (=1.0)
     32  float3 (=1.0)
     32  unknown (27)
     32  unknown (8)
     32  zero ?
     */
 
     if (!avctx->extradata || (avctx->extradata_size < 48)) {
         av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
         return -1;
     }
 
     extradata = avctx->extradata;
     extradata_size = avctx->extradata_size;
 
     while (extradata_size > 7) {
         if (!memcmp(extradata, "frmaQDM", 7))
             break;
         extradata++;
         extradata_size--;
     }
 
     if (extradata_size < 12) {
         av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
                extradata_size);
         return -1;
     }
 
     if (memcmp(extradata, "frmaQDM", 7)) {
         av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
         return -1;
     }
 
     if (extradata[7] == 'C') {
 //        s->is_qdmc = 1;
         av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
         return -1;
     }
 
     extradata += 8;
     extradata_size -= 8;
 

     size = AV_RB32(extradata);

 
     if(size > extradata_size){
         av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
                extradata_size, size);
         return -1;
     }
 
     extradata += 4;
     av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);

     if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {

         av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
         return -1;
     }
 
     extradata += 8;
 

     avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);

     extradata += 4;

     if (s->channels > MPA_MAX_CHANNELS)
         return AVERROR_INVALIDDATA;

     avctx->sample_rate = AV_RB32(extradata);

     extradata += 4;
 

     avctx->bit_rate = AV_RB32(extradata);

     extradata += 4;
 

     s->group_size = AV_RB32(extradata);

     extradata += 4;
 

     s->fft_size = AV_RB32(extradata);

     extradata += 4;
 

     s->checksum_size = AV_RB32(extradata);

     extradata += 4;
 
     s->fft_order = av_log2(s->fft_size) + 1;
     s->fft_frame_size = 2 * s->fft_size; // complex has two floats
 
     // something like max decodable tones
     s->group_order = av_log2(s->group_size) + 1;
     s->frame_size = s->group_size / 16; // 16 iterations per super block

 
     if (s->frame_size > QDM2_MAX_FRAME_SIZE)

         return AVERROR_INVALIDDATA;

     s->sub_sampling = s->fft_order - 7;

     s->frequency_range = 255 / (1 << (2 - s->sub_sampling));

     switch ((s->sub_sampling * 2 + s->channels - 1)) {
         case 0: tmp = 40; break;
         case 1: tmp = 48; break;
         case 2: tmp = 56; break;
         case 3: tmp = 72; break;
         case 4: tmp = 80; break;
         case 5: tmp = 100;break;
         default: tmp=s->sub_sampling; break;
     }
     tmp_val = 0;
     if ((tmp * 1000) < avctx->bit_rate)  tmp_val = 1;
     if ((tmp * 1440) < avctx->bit_rate)  tmp_val = 2;
     if ((tmp * 1760) < avctx->bit_rate)  tmp_val = 3;
     if ((tmp * 2240) < avctx->bit_rate)  tmp_val = 4;
     s->cm_table_select = tmp_val;
 
     if (s->sub_sampling == 0)

         tmp = 7999;

     else
         tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
     /*

     0: 7999 -> 0

     1: 20000 -> 2
     2: 28000 -> 2
     */
     if (tmp < 8000)
         s->coeff_per_sb_select = 0;
     else if (tmp <= 16000)
         s->coeff_per_sb_select = 1;
     else
         s->coeff_per_sb_select = 2;
 

     // Fail on unknown fft order

     if ((s->fft_order < 7) || (s->fft_order > 9)) {

         av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);

         return -1;
     }

     ff_rdft_init(&s->rdft_ctx, s->fft_order, IRDFT);

 
     qdm2_init(s);

     avctx->sample_fmt = SAMPLE_FMT_S16;
 

 //    dump_context(s);
     return 0;
 }
 
 

 static av_cold int qdm2_decode_close(AVCodecContext *avctx)

 {
     QDM2Context *s = avctx->priv_data;
 

     ff_rdft_end(&s->rdft_ctx);

     return 0;
 }
 
 

 static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)

 {
     int ch, i;
     const int frame_size = (q->frame_size * q->channels);

     if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
         return -1;
 

     /* select input buffer */
     q->compressed_data = in;
     q->compressed_size = q->checksum_size;
 
 //  dump_context(q);
 
     /* copy old block, clear new block of output samples */
     memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
     memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
 
     /* decode block of QDM2 compressed data */
     if (q->sub_packet == 0) {
         q->has_errors = 0; // zero it for a new super block

         av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");

         qdm2_decode_super_block(q);
     }
 

     /* parse subpackets */

     if (!q->has_errors) {
         if (q->sub_packet == 2)
             qdm2_decode_fft_packets(q);
 
         qdm2_fft_tone_synthesizer(q, q->sub_packet);
     }
 
     /* sound synthesis stage 1 (FFT) */
     for (ch = 0; ch < q->channels; ch++) {
         qdm2_calculate_fft(q, ch, q->sub_packet);
 
         if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
             SAMPLES_NEEDED_2("has errors, and C list is not empty")

             return -1;

         }
     }
 
     /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
     if (!q->has_errors && q->do_synth_filter)
         qdm2_synthesis_filter(q, q->sub_packet);
 
     q->sub_packet = (q->sub_packet + 1) % 16;
 
     /* clip and convert output float[] to 16bit signed samples */
     for (i = 0; i < frame_size; i++) {
         int value = (int)q->output_buffer[i];
 
         if (value > SOFTCLIP_THRESHOLD)
             value = (value >  HARDCLIP_THRESHOLD) ?  32767 :  softclip_table[ value - SOFTCLIP_THRESHOLD];
         else if (value < -SOFTCLIP_THRESHOLD)
             value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
 
         out[i] = value;
     }

 
     return 0;

 }
 
 
 static int qdm2_decode_frame(AVCodecContext *avctx,
             void *data, int *data_size,

             const uint8_t *buf, int buf_size)

 {
     QDM2Context *s = avctx->priv_data;

     int16_t *out = data;

     int i, out_size;

     if(!buf)

         return 0;

     if(buf_size < s->checksum_size)
         return -1;

     out_size = 16 * s->channels * s->frame_size *
                av_get_bits_per_sample_format(avctx->sample_fmt)/8;
     if (*data_size < out_size) {
         av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
         return AVERROR(EINVAL);
     }

 
     av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
        buf_size, buf, s->checksum_size, data, *data_size);
 

     for (i = 0; i < 16; i++) {
         if (qdm2_decode(s, buf, out) < 0)
             return -1;
         out += s->channels * s->frame_size;

     }
 

     *data_size = out_size;

 
     return buf_size;

 }
 
 AVCodec qdm2_decoder =
 {
     .name = "qdm2",
     .type = CODEC_TYPE_AUDIO,
     .id = CODEC_ID_QDM2,
     .priv_data_size = sizeof(QDM2Context),
     .init = qdm2_decode_init,
     .close = qdm2_decode_close,
     .decode = qdm2_decode_frame,

     .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),

 };