Mercurial > libavcodec.hg
changeset 2914:4a52affac0e0 libavcodec
QDM2 compatible decoder
| author | rtognimp |
|---|---|
| date | Tue, 18 Oct 2005 20:31:12 +0000 |
| parents | cc55bc1f8d92 |
| children | aa98fe99148e |
| files | qdm2.c qdm2data.h |
| diffstat | 2 files changed, 2567 insertions(+), 0 deletions(-) [+] |
line wrap: on
line diff
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/qdm2.c Tue Oct 18 20:31:12 2005 +0000 @@ -0,0 +1,2039 @@ +/* + * QDM2 compatible decoder + * Copyright (c) 2003 Ewald Snel + * Copyright (c) 2005 Benjamin Larsson + * Copyright (c) 2005 Alex Beregszaszi + * Copyright (c) 2005 Roberto Togni + * + * This library 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 of the License, or (at your option) any later version. + * + * This library 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 this library; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +/** + * @file qdm2.c + * QDM2 decoder + * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni + * The decoder is not perfect yet, there are still some distorions expecially + * 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" + +#ifdef CONFIG_MPEGAUDIO_HP +#define USE_HIGHPRECISION +#endif + +#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); + + +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 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 level; + float *samples_im; + float *samples_re; + 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 { + float re; + float im; +} QDM2Complex; + +typedef struct { + QDM2Complex complex[256 + 1] __attribute__((aligned(16))); + float samples_im[MPA_MAX_CHANNELS][256]; + float samples_re[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]; + FFTContext fft_ctx; + FFTComplex exptab[128]; + QDM2FFT fft; + + /// I/O data + uint8_t *compressed_data; + int compressed_size; + float output_buffer[1024]; + + /// Synthesis filter + MPA_INT synth_buf[MPA_MAX_CHANNELS][512*2] __attribute__((aligned(16))); + int synth_buf_offset[MPA_MAX_CHANNELS]; + int32_t sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT] __attribute__((aligned(16))); + + /// 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 have 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 MPA_INT mpa_window[512] __attribute__((aligned(16))); + + +static void softclip_table_init() { + 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 void rnd_table_init() { + 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 void init_noise_samples() { + 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 void qdm2_init_vlc() +{ + 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 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 (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,"Sub packet: 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 subpacket 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 subblocks 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] + */ + 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]]) { + 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 < 64; j++) { + 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 bit + * @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 bit + */ +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 bit + */ +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); + + 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 bit + */ +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)); + + 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)); + 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 bit + */ +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)); + 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); + 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); + + 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); + skip_bits(&gb, next_index*8); + + if (next_index >= header.size) + break; + } + + /* decode sub packet */ + 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 sub packet to 'all sub packets' list */ + q->sub_packet_list_A[i].packet = packet; + + /* add sub packet 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 (1) { + 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 (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 indices for FFT coefficients */ + q->fft_coefs_index = 0; + for (i=0; i < 5; i++) + q->fft_coefs_min_index[i] = -1; + + /* process sub packets ordered by type, largest type first */ + for (i = 0, max = 256; i < q->sub_packets_B; i++) { + QDM2SubPacket *packet; + + /* find sub packet with largest type less than max */ + for (j = 0, min = 0, packet = NULL; 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 (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); + + 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 (i=0; i < 4; i++) + qdm2_fft_decode_tones(q, i, &gb, unknown_flag); + } else if (type == 46) { + for (i=0; i < 6; i++) + q->fft_level_exp[i] = get_bits(&gb, 6); + for (i=0; i < 4; i++) + qdm2_fft_decode_tones(q, i, &gb, unknown_flag); + } + } // Loop on B packets + + /* calculate maximum indices 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->samples_im[0] += c.im; + tone->samples_re[0] += c.re; + tone->samples_im[1] -= c.im; + tone->samples_re[1] -= 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->samples_re[fft_cutoff_index_table[tone->cutoff][i]] += c.re * f[i]; + tone->samples_im[fft_cutoff_index_table[tone->cutoff][i]] += c.im *((tone->cutoff <= i) ? -f[i] : f[i]); + } + for (i = 0; i < 4; i++) { + tone->samples_re[i] += c.re * f[i+2]; + tone->samples_im[i] += 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.samples_im[ch], 0, q->fft_size * sizeof(float)); + memset(q->fft.samples_re[ch], 0, q->fft_size * sizeof(float)); + } + + + /* 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.samples_re[ch][q->fft_coefs[i].offset + 0] += c.re; + q->fft.samples_im[ch][q->fft_coefs[i].offset + 0] += c.im; + q->fft.samples_re[ch][q->fft_coefs[i].offset + 1] -= c.re; + q->fft.samples_im[ch][q->fft_coefs[i].offset + 1] -= 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.samples_im = &q->fft.samples_im[ch][offset]; + tone.samples_re = &q->fft.samples_re[ch][offset]; + tone.table = (float*)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 int n = 1 << (q->fft_order - 1); + const int n2 = n >> 1; + const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.25f : 0.50f; + float c, s, f0, f1, f2, f3; + int i, j; + + /* pre rotation (or something like that) */ + for (i=1; i < n2; i++) { + j = (n - i); + c = q->exptab[i].re; + s = -q->exptab[i].im; + f0 = (q->fft.samples_re[channel][i] - q->fft.samples_re[channel][j]) * gain; + f1 = (q->fft.samples_im[channel][i] + q->fft.samples_im[channel][j]) * gain; + f2 = (q->fft.samples_re[channel][i] + q->fft.samples_re[channel][j]) * gain; + f3 = (q->fft.samples_im[channel][i] - q->fft.samples_im[channel][j]) * gain; + q->fft.complex[i].re = s * f0 - c * f1 + f2; + q->fft.complex[i].im = c * f0 + s * f1 + f3; + q->fft.complex[j].re = -s * f0 + c * f1 + f2; + q->fft.complex[j].im = c * f0 + s * f1 - f3; + } + + q->fft.complex[ 0].re = q->fft.samples_re[channel][ 0] * gain * 2.0; + q->fft.complex[ 0].im = q->fft.samples_re[channel][ 0] * gain * 2.0; + q->fft.complex[n2].re = q->fft.samples_re[channel][n2] * gain * 2.0; + q->fft.complex[n2].im = -q->fft.samples_im[channel][n2] * gain * 2.0; + + ff_fft_permute(&q->fft_ctx, (FFTComplex *) q->fft.complex); + ff_fft_calc (&q->fft_ctx, (FFTComplex *) q->fft.complex); + /* 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)[i]; +} + + +/** + * @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 + */ +void qdm2_init(QDM2Context *q) { + static int inited = 0; + + if (inited != 0) + return; + inited = 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 int qdm2_decode_init(AVCodecContext *avctx) +{ + QDM2Context *s = avctx->priv_data; + uint8_t *extradata; + int extradata_size; + int tmp_val, tmp, size; + int i; + float alpha; + + /* 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 = BE_32(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 (BE_32(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 = BE_32(extradata); + extradata += 4; + + avctx->sample_rate = BE_32(extradata); + extradata += 4; + + avctx->bit_rate = BE_32(extradata); + extradata += 4; + + s->group_size = BE_32(extradata); + extradata += 4; + + s->fft_size = BE_32(extradata); + extradata += 4; + + s->checksum_size = BE_32(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->fft_order == 8) + s->sub_sampling = 1; + else + s->sub_sampling = 2; + 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 = 16000; + else + tmp = ((-(s->sub_sampling -1)) & 8000) + 20000; + /* + 0: 16000 -> 1 + 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; + + if (s->fft_order != 8 && s->fft_order != 9) + av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order); + + ff_fft_init(&s->fft_ctx, s->fft_order - 1, 1); + + for (i = 1; i < (1 << (s->fft_order - 2)); i++) { + alpha = 2 * M_PI * (float)i / (float)(1 << (s->fft_order - 1)); + s->exptab[i].re = cos(alpha); + s->exptab[i].im = sin(alpha); + } + + ff_fft_init(&s->fft_ctx, s->fft_order - 1, 1); + qdm2_init(s); + +// dump_context(s); + return 0; +} + + +static int qdm2_decode_close(AVCodecContext *avctx) +{ + QDM2Context *s = avctx->priv_data; + + ff_fft_end(&s->fft_ctx); + + return 0; +} + + +void qdm2_decode (QDM2Context *q, uint8_t *in, int16_t *out) +{ + int ch, i; + const int frame_size = (q->frame_size * q->channels); + + /* 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,"Super block follows\n"); + qdm2_decode_super_block(q); + } + + /* parse sub packets */ + 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; + } + } + + /* 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; + } +} + + +static int qdm2_decode_frame(AVCodecContext *avctx, + void *data, int *data_size, + uint8_t *buf, int buf_size) +{ + QDM2Context *s = avctx->priv_data; + + if((buf == NULL) || (buf_size < s->checksum_size)) + return 0; + + *data_size = s->channels * s->frame_size * sizeof(int16_t); + + av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n", + buf_size, buf, s->checksum_size, data, *data_size); + + qdm2_decode(s, buf, data); + + // reading only when next superblock found + if (s->sub_packet == 0) { + return s->checksum_size; + } + + return 0; +} + +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, +};
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/qdm2data.h Tue Oct 18 20:31:12 2005 +0000 @@ -0,0 +1,528 @@ +/* + * QDM2 compatible decoder + * Copyright (c) 2003 Ewald Snel + * Copyright (c) 2005 Benjamin Larsson + * Copyright (c) 2005 Alex Beregszaszi + * Copyright (c) 2005 Roberto Togni + * + * This library 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 of the License, or (at your option) any later version. + * + * This library 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 this library; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + + /** + * @file qdm2data.h + * Various QDM2 tables. + */ + +#ifndef QDM2DATA_H +#define QDM2DATA_H + +/** VLC TABLES **/ + +/* values in this table range from -1..23; adjust retrieved value by -1 */ +static uint16_t vlc_tab_level_huffcodes[24] = { + 0x037c, 0x0004, 0x003c, 0x004c, 0x003a, 0x002c, 0x001c, 0x001a, + 0x0024, 0x0014, 0x0001, 0x0002, 0x0000, 0x0003, 0x0007, 0x0005, + 0x0006, 0x0008, 0x0009, 0x000a, 0x000c, 0x00fc, 0x007c, 0x017c +}; + +static uint8_t vlc_tab_level_huffbits[24] = { + 10, 6, 7, 7, 6, 6, 6, 6, 6, 5, 4, 4, 4, 3, 3, 3, 3, 4, 4, 5, 7, 8, 9, 10 +}; + +/* values in this table range from -1..36; adjust retrieved value by -1 */ +static uint16_t vlc_tab_diff_huffcodes[37] = { + 0x1c57, 0x0004, 0x0000, 0x0001, 0x0003, 0x0002, 0x000f, 0x000e, + 0x0007, 0x0016, 0x0037, 0x0027, 0x0026, 0x0066, 0x0006, 0x0097, + 0x0046, 0x01c6, 0x0017, 0x0786, 0x0086, 0x0257, 0x00d7, 0x0357, + 0x00c6, 0x0386, 0x0186, 0x0000, 0x0157, 0x0c57, 0x0057, 0x0000, + 0x0b86, 0x0000, 0x1457, 0x0000, 0x0457 +}; + +static uint8_t vlc_tab_diff_huffbits[37] = { + 13, 3, 3, 2, 3, 3, 4, 4, 6, 5, 6, 6, 7, 7, 8, 8, + 8, 9, 8, 11, 9, 10, 8, 10, 9, 12, 10, 0, 10, 13, 11, 0, + 12, 0, 13, 0, 13 +}; + +/* values in this table range from -1..5; adjust retrieved value by -1 */ +static uint8_t vlc_tab_run_huffcodes[6] = { + 0x1f, 0x00, 0x01, 0x03, 0x07, 0x0f +}; + +static uint8_t vlc_tab_run_huffbits[6] = { + 5, 1, 2, 3, 4, 5 +}; + +/* values in this table range from -1..19; adjust retrieved value by -1 */ +static uint16_t vlc_tab_tone_level_idx_hi1_huffcodes[20] = { + 0x5714, 0x000c, 0x0002, 0x0001, 0x0000, 0x0004, 0x0034, 0x0054, + 0x0094, 0x0014, 0x0114, 0x0214, 0x0314, 0x0614, 0x0e14, 0x0f14, + 0x2714, 0x0714, 0x1714, 0x3714 +}; + +static uint8_t vlc_tab_tone_level_idx_hi1_huffbits[20] = { + 15, 4, 2, 1, 3, 5, 6, 7, 8, 10, 10, 11, 11, 12, 12, 12, 14, 14, 15, 14 +}; + +/* values in this table range from -1..23; adjust retrieved value by -1 */ +static uint16_t vlc_tab_tone_level_idx_mid_huffcodes[24] = { + 0x0fea, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, 0x03ea, 0x00ea, 0x002a, 0x001a, + 0x0006, 0x0001, 0x0000, 0x0002, 0x000a, 0x006a, 0x01ea, 0x07ea +}; + +static uint8_t vlc_tab_tone_level_idx_mid_huffbits[24] = { + 12, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 11, 9, 7, 5, 3, 1, 2, 4, 6, 8, 10, 12 +}; + +/* values in this table range from -1..23; adjust retrieved value by -1 */ +static uint16_t vlc_tab_tone_level_idx_hi2_huffcodes[24] = { + 0x0664, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0064, 0x00e4, + 0x00a4, 0x0068, 0x0004, 0x0008, 0x0014, 0x0018, 0x0000, 0x0001, + 0x0002, 0x0003, 0x000c, 0x0028, 0x0024, 0x0164, 0x0000, 0x0264 +}; + +static uint8_t vlc_tab_tone_level_idx_hi2_huffbits[24] = { + 11, 0, 0, 0, 0, 0, 10, 8, 8, 7, 6, 6, 5, 5, 4, 2, 2, 2, 4, 7, 8, 9, 0, 11 +}; + +/* values in this table range from -1..8; adjust retrieved value by -1 */ +static uint8_t vlc_tab_type30_huffcodes[9] = { + 0x3c, 0x06, 0x00, 0x01, 0x03, 0x02, 0x04, 0x0c, 0x1c +}; + +static uint8_t vlc_tab_type30_huffbits[9] = { + 6, 3, 3, 2, 2, 3, 4, 5, 6 +}; + +/* values in this table range from -1..9; adjust retrieved value by -1 */ +static uint8_t vlc_tab_type34_huffcodes[10] = { + 0x18, 0x00, 0x01, 0x04, 0x05, 0x07, 0x03, 0x02, 0x06, 0x08 +}; + +static uint8_t vlc_tab_type34_huffbits[10] = { + 5, 4, 3, 3, 3, 3, 3, 3, 3, 5 +}; + +/* values in this table range from -1..22; adjust retrieved value by -1 */ +static uint16_t vlc_tab_fft_tone_offset_0_huffcodes[23] = { + 0x038e, 0x0001, 0x0000, 0x0022, 0x000a, 0x0006, 0x0012, 0x0002, + 0x001e, 0x003e, 0x0056, 0x0016, 0x000e, 0x0032, 0x0072, 0x0042, + 0x008e, 0x004e, 0x00f2, 0x002e, 0x0036, 0x00c2, 0x018e +}; + +static uint8_t vlc_tab_fft_tone_offset_0_huffbits[23] = { + 10, 1, 2, 6, 4, 5, 6, 7, 6, 6, 7, 7, 8, 7, 8, 8, 9, 7, 8, 6, 6, 8, 10 +}; + +/* values in this table range from -1..27; adjust retrieved value by -1 */ +static uint16_t vlc_tab_fft_tone_offset_1_huffcodes[28] = { + 0x07a4, 0x0001, 0x0020, 0x0012, 0x001c, 0x0008, 0x0006, 0x0010, + 0x0000, 0x0014, 0x0004, 0x0032, 0x0070, 0x000c, 0x0002, 0x003a, + 0x001a, 0x002c, 0x002a, 0x0022, 0x0024, 0x000a, 0x0064, 0x0030, + 0x0062, 0x00a4, 0x01a4, 0x03a4 +}; + +static uint8_t vlc_tab_fft_tone_offset_1_huffbits[28] = { + 11, 1, 6, 6, 5, 4, 3, 6, 6, 5, 6, 6, 7, 6, 6, 6, + 6, 6, 6, 7, 8, 6, 7, 7, 7, 9, 10, 11 +}; + +/* values in this table range from -1..31; adjust retrieved value by -1 */ +static uint16_t vlc_tab_fft_tone_offset_2_huffcodes[32] = { + 0x1760, 0x0001, 0x0000, 0x0082, 0x000c, 0x0006, 0x0003, 0x0007, + 0x0008, 0x0004, 0x0010, 0x0012, 0x0022, 0x001a, 0x0000, 0x0020, + 0x000a, 0x0040, 0x004a, 0x006a, 0x002a, 0x0042, 0x0002, 0x0060, + 0x00aa, 0x00e0, 0x00c2, 0x01c2, 0x0160, 0x0360, 0x0760, 0x0f60 +}; + +static uint8_t vlc_tab_fft_tone_offset_2_huffbits[32] = { + 13, 2, 0, 8, 4, 3, 3, 3, 4, 4, 5, 5, 6, 5, 7, 7, + 7, 7, 7, 7, 8, 8, 8, 9, 8, 8, 9, 9, 10, 11, 13, 12 +}; + +/* values in this table range from -1..34; adjust retrieved value by -1 */ +static uint16_t vlc_tab_fft_tone_offset_3_huffcodes[35] = { + 0x33ea, 0x0005, 0x0000, 0x000c, 0x0000, 0x0006, 0x0003, 0x0008, + 0x0002, 0x0001, 0x0004, 0x0007, 0x001a, 0x000f, 0x001c, 0x002c, + 0x000a, 0x001d, 0x002d, 0x002a, 0x000d, 0x004c, 0x008c, 0x006a, + 0x00cd, 0x004d, 0x00ea, 0x020c, 0x030c, 0x010c, 0x01ea, 0x07ea, + 0x0bea, 0x03ea, 0x13ea +}; + +static uint8_t vlc_tab_fft_tone_offset_3_huffbits[35] = { + 14, 4, 0, 10, 4, 3, 3, 4, 4, 3, 4, 4, 5, 4, 5, 6, + 6, 5, 6, 7, 7, 7, 8, 8, 8, 8, 9, 10, 10, 10, 10, 11, + 12, 13, 14 +}; + +/* values in this table range from -1..37; adjust retrieved value by -1 */ +static uint16_t vlc_tab_fft_tone_offset_4_huffcodes[38] = { + 0x5282, 0x0016, 0x0000, 0x0136, 0x0004, 0x0000, 0x0007, 0x000a, + 0x000e, 0x0003, 0x0001, 0x000d, 0x0006, 0x0009, 0x0012, 0x0005, + 0x0025, 0x0022, 0x0015, 0x0002, 0x0076, 0x0035, 0x0042, 0x00c2, + 0x0182, 0x00b6, 0x0036, 0x03c2, 0x0482, 0x01c2, 0x0682, 0x0882, + 0x0a82, 0x0082, 0x0282, 0x1282, 0x3282, 0x2282 +}; + +static uint8_t vlc_tab_fft_tone_offset_4_huffbits[38] = { + 15, 6, 0, 9, 3, 3, 3, 4, 4, 3, 4, 4, 5, 4, 5, 6, + 6, 6, 6, 8, 7, 6, 8, 9, 9, 8, 9, 10, 11, 10, 11, 12, + 12, 12, 14, 15, 14, 14 +}; + +/** FFT TABLES **/ + +/* values in this table range from -1..27; adjust retrieved value by -1 */ +static uint16_t fft_level_exp_alt_huffcodes[28] = { + 0x1ec6, 0x0006, 0x00c2, 0x0142, 0x0242, 0x0246, 0x00c6, 0x0046, + 0x0042, 0x0146, 0x00a2, 0x0062, 0x0026, 0x0016, 0x000e, 0x0005, + 0x0004, 0x0003, 0x0000, 0x0001, 0x000a, 0x0012, 0x0002, 0x0022, + 0x01c6, 0x02c6, 0x06c6, 0x0ec6 +}; + +static uint8_t fft_level_exp_alt_huffbits[28] = { + 13, 7, 8, 9, 10, 10, 10, 10, 10, 9, 8, 7, 6, 5, 4, 3, + 3, 2, 3, 3, 4, 5, 7, 8, 9, 11, 12, 13 +}; + +/* values in this table range from -1..19; adjust retrieved value by -1 */ +static uint16_t fft_level_exp_huffcodes[20] = { + 0x0f24, 0x0001, 0x0002, 0x0000, 0x0006, 0x0005, 0x0007, 0x000c, + 0x000b, 0x0014, 0x0013, 0x0004, 0x0003, 0x0023, 0x0064, 0x00a4, + 0x0024, 0x0124, 0x0324, 0x0724 +}; + +static uint8_t fft_level_exp_huffbits[20] = { + 12, 3, 3, 3, 3, 3, 3, 4, 4, 5, 5, 6, 6, 6, 7, 8, 9, 10, 11, 12 +}; + +/* values in this table range from -1..6; adjust retrieved value by -1 */ +static uint8_t fft_stereo_exp_huffcodes[7] = { + 0x3e, 0x01, 0x00, 0x02, 0x06, 0x0e, 0x1e +}; + +static uint8_t fft_stereo_exp_huffbits[7] = { + 6, 1, 2, 3, 4, 5, 6 +}; + +/* values in this table range from -1..8; adjust retrieved value by -1 */ +static uint8_t fft_stereo_phase_huffcodes[9] = { + 0x35, 0x02, 0x00, 0x01, 0x0d, 0x15, 0x05, 0x09, 0x03 +}; + +static uint8_t fft_stereo_phase_huffbits[9] = { + 6, 2, 2, 4, 4, 6, 5, 4, 2 +}; + +static const int fft_cutoff_index_table[4][2] = { + { 1, 2 }, {-1, 0 }, {-1,-2 }, { 0, 0 } +}; + +static const int16_t fft_level_index_table[256] = { + 0, 0, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1, + 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, + 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, + 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, + 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, + 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, + 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, + 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, + 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, + 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, + 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, + 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, + 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, + 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, + 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, + 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, +}; + +static uint8_t last_coeff[3] = { + 4, 7, 10 +}; + +static uint8_t coeff_per_sb_for_avg[3][30] = { + { 0, 1, 1, 1, 1, 2, 2, 2, 2, 2, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3 }, + { 0, 1, 2, 2, 3, 3, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5, 5, 5, 5, 5, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6 }, + { 0, 1, 2, 3, 4, 4, 5, 5, 6, 6, 6, 6, 7, 7, 7, 7, 8, 8, 8, 8, 8, 8, 9, 9, 9, 9, 9, 9, 9, 9 } +}; + +static uint32_t dequant_table[3][10][30] = { + { { 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 256, 256, 205, 154, 102, 51, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 0, 51, 102, 154, 205, 256, 238, 219, 201, 183, 165, 146, 128, 110, 91, 73, 55, 37, 18, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 0, 0, 0, 0, 0, 0, 18, 37, 55, 73, 91, 110, 128, 146, 165, 183, 201, 219, 238, 256, 228, 199, 171, 142, 114, 85, 57, 28 }, + { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 } }, + { { 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 256, 171, 85, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 0, 85, 171, 256, 171, 85, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 0, 0, 0, 0, 85, 171, 256, 219, 183, 146, 110, 73, 37, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 0, 0, 0, 0, 0, 0, 0, 37, 73, 110, 146, 183, 219, 256, 228, 199, 171, 142, 114, 85, 57, 28, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 28, 57, 85, 114, 142, 171, 199, 228, 256, 213, 171, 128, 85, 43 }, + { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 } }, + { { 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 0, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 0, 0, 256, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 0, 0, 0, 0, 256, 171, 85, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 0, 0, 0, 0, 0, 85, 171, 256, 192, 128, 64, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 64, 128, 192, 256, 205, 154, 102, 51, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 51, 102, 154, 205, 256, 213, 171, 128, 85, 43, 0, 0, 0, 0, 0, 0 }, + { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 43, 85, 128, 171, 213, 256, 213, 171, 128, 85, 43 } } +}; + +static uint8_t coeff_per_sb_for_dequant[3][30] = { + { 0, 1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3, 3 }, + { 0, 1, 2, 2, 2, 3, 3, 3, 4, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5, 5, 5, 5, 5, 5, 6, 6, 6, 6, 6, 6 }, + { 0, 1, 2, 3, 4, 4, 5, 5, 5, 6, 6, 6, 6, 7, 7, 7, 7, 7, 8, 8, 8, 8, 8, 8, 9, 9, 9, 9, 9, 9 } +}; + +/* first index is subband, 2nd index is 0, 1 or 3 (2 is unused) */ +static int8_t tone_level_idx_offset_table[30][4] = { + { -50, -50, 0, -50 }, + { -50, -50, 0, -50 }, + { -50, -9, 0, -19 }, + { -16, -6, 0, -12 }, + { -11, -4, 0, -8 }, + { -8, -3, 0, -6 }, + { -7, -3, 0, -5 }, + { -6, -2, 0, -4 }, + { -5, -2, 0, -3 }, + { -4, -1, 0, -3 }, + { -4, -1, 0, -2 }, + { -3, -1, 0, -2 }, + { -3, -1, 0, -2 }, + { -3, -1, 0, -2 }, + { -2, -1, 0, -1 }, + { -2, -1, 0, -1 }, + { -2, -1, 0, -1 }, + { -2, 0, 0, -1 }, + { -2, 0, 0, -1 }, + { -1, 0, 0, -1 }, + { -1, 0, 0, -1 }, + { -1, 0, 0, -1 }, + { -1, 0, 0, -1 }, + { -1, 0, 0, -1 }, + { -1, 0, 0, -1 }, + { -1, 0, 0, -1 }, + { -1, 0, 0, 0 }, + { -1, 0, 0, 0 }, + { -1, 0, 0, 0 }, + { -1, 0, 0, 0 } +}; + +/* all my samples have 1st index 0 or 1 */ +/* second index is subband, only indexes 0-29 seem to be used */ +static int8_t coding_method_table[5][30] = { + { 34, 30, 24, 24, 16, 16, 16, 16, 10, 10, 10, 10, 10, 10, 10, + 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10 + }, + { 34, 30, 24, 24, 16, 16, 16, 16, 10, 10, 10, 10, 10, 10, 10, + 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10 + }, + { 34, 30, 30, 30, 24, 24, 16, 16, 16, 16, 16, 16, 10, 10, 10, + 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10 + }, + { 34, 34, 30, 30, 24, 24, 24, 24, 16, 16, 16, 16, 16, 16, 16, + 16, 16, 16, 16, 16, 16, 16, 10, 10, 10, 10, 10, 10, 10, 10 + }, + { 34, 34, 30, 30, 30, 30, 30, 30, 24, 24, 24, 24, 24, 24, 24, + 24, 24, 24, 24, 24, 16, 16, 16, 16, 16, 16, 16, 16, 16, 16 + }, +}; + +static const int vlc_stage3_values[60] = { + 0, 1, 2, 3, 4, 6, 8, 10, 12, 16, 20, 24, + 28, 36, 44, 52, 60, 76, 92, 108, 124, 156, 188, 220, + 252, 316, 380, 444, 508, 636, 764, 892, 1020, 1276, 1532, 1788, + 2044, 2556, 3068, 3580, 4092, 5116, 6140, 7164, 8188, 10236, 12284, 14332, + 16380, 20476, 24572, 28668, 32764, 40956, 49148, 57340, 65532, 81916, 98300,114684 +}; + +static const float fft_tone_sample_table[4][16][5] = { + { { .0100000000f,-.0037037037f,-.0020000000f,-.0069444444f,-.0018416207f }, + { .0416666667f, .0000000000f, .0000000000f,-.0208333333f,-.0123456791f }, + { .1250000000f, .0558035709f, .0330687836f,-.0164473690f,-.0097465888f }, + { .1562500000f, .0625000000f, .0370370370f,-.0062500000f,-.0037037037f }, + { .1996007860f, .0781250000f, .0462962948f, .0022727272f, .0013468013f }, + { .2000000000f, .0625000000f, .0370370373f, .0208333333f, .0074074073f }, + { .2127659619f, .0555555556f, .0329218097f, .0208333333f, .0123456791f }, + { .2173913121f, .0473484844f, .0280583613f, .0347222239f, .0205761325f }, + { .2173913121f, .0347222239f, .0205761325f, .0473484844f, .0280583613f }, + { .2127659619f, .0208333333f, .0123456791f, .0555555556f, .0329218097f }, + { .2000000000f, .0208333333f, .0074074073f, .0625000000f, .0370370370f }, + { .1996007860f, .0022727272f, .0013468013f, .0781250000f, .0462962948f }, + { .1562500000f,-.0062500000f,-.0037037037f, .0625000000f, .0370370370f }, + { .1250000000f,-.0164473690f,-.0097465888f, .0558035709f, .0330687836f }, + { .0416666667f,-.0208333333f,-.0123456791f, .0000000000f, .0000000000f }, + { .0100000000f,-.0069444444f,-.0018416207f,-.0037037037f,-.0020000000f } }, + + { { .0050000000f,-.0200000000f, .0125000000f,-.3030303030f, .0020000000f }, + { .1041666642f, .0400000000f,-.0250000000f, .0333333333f,-.0200000000f }, + { .1250000000f, .0100000000f, .0142857144f,-.0500000007f,-.0200000000f }, + { .1562500000f,-.0006250000f,-.00049382716f,-.000625000f,-.00049382716f }, + { .1562500000f,-.0006250000f,-.00049382716f,-.000625000f,-.00049382716f }, + { .1250000000f,-.0500000000f,-.0200000000f, .0100000000f, .0142857144f }, + { .1041666667f, .0333333333f,-.0200000000f, .0400000000f,-.0250000000f }, + { .0050000000f,-.3030303030f, .0020000001f,-.0200000000f, .0125000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f } }, + + { { .1428571492f, .1250000000f,-.0285714287f,-.0357142873f, .0208333333f }, + { .1818181818f, .0588235296f, .0333333333f, .0212765951f, .0100000000f }, + { .1818181818f, .0212765951f, .0100000000f, .0588235296f, .0333333333f }, + { .1428571492f,-.0357142873f, .0208333333f, .1250000000f,-.0285714287f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f } }, + + { { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f }, + { .0000000000f, .0000000000f, .0000000000f, .0000000000f, .0000000000f } } +}; + +static const float fft_tone_level_table[2][64] = { { +/* pow ~ (i > 46) ? 0 : (((((i & 1) ? 431 : 304) << (i >> 1))) / 1024.0); */ + 0.17677669f, 0.42677650f, 0.60355347f, 0.85355347f, + 1.20710683f, 1.68359375f, 2.37500000f, 3.36718750f, + 4.75000000f, 6.73437500f, 9.50000000f, 13.4687500f, + 19.0000000f, 26.9375000f, 38.0000000f, 53.8750000f, + 76.0000000f, 107.750000f, 152.000000f, 215.500000f, + 304.000000f, 431.000000f, 608.000000f, 862.000000f, + 1216.00000f, 1724.00000f, 2432.00000f, 3448.00000f, + 4864.00000f, 6896.00000f, 9728.00000f, 13792.0000f, + 19456.0000f, 27584.0000f, 38912.0000f, 55168.0000f, + 77824.0000f, 110336.000f, 155648.000f, 220672.000f, + 311296.000f, 441344.000f, 622592.000f, 882688.000f, + 1245184.00f, 1765376.00f, 2490368.00f, 0.00000000f, + 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, + 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, + 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, + 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, + }, { +/* pow = (i > 45) ? 0 : ((((i & 1) ? 431 : 304) << (i >> 1)) / 512.0); */ + 0.59375000f, 0.84179688f, 1.18750000f, 1.68359375f, + 2.37500000f, 3.36718750f, 4.75000000f, 6.73437500f, + 9.50000000f, 13.4687500f, 19.0000000f, 26.9375000f, + 38.0000000f, 53.8750000f, 76.0000000f, 107.750000f, + 152.000000f, 215.500000f, 304.000000f, 431.000000f, + 608.000000f, 862.000000f, 1216.00000f, 1724.00000f, + 2432.00000f, 3448.00000f, 4864.00000f, 6896.00000f, + 9728.00000f, 13792.0000f, 19456.0000f, 27584.0000f, + 38912.0000f, 55168.0000f, 77824.0000f, 110336.000f, + 155648.000f, 220672.000f, 311296.000f, 441344.000f, + 622592.000f, 882688.000f, 1245184.00f, 1765376.00f, + 2490368.00f, 3530752.00f, 0.00000000f, 0.00000000f, + 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, + 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, + 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f, + 0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f +} }; + +static const float fft_tone_envelope_table[4][31] = { + { .009607375f, .038060248f, .084265202f, .146446645f, .222214907f, .308658302f, + .402454883f, .500000060f, .597545207f, .691341758f, .777785182f, .853553414f, + .915734828f, .961939812f, .990392685f, 1.00000000f, .990392625f, .961939752f, + .915734768f, .853553295f, .777785063f, .691341639f, .597545087f, .500000000f, + .402454853f, .308658272f, .222214878f, .146446615f, .084265172f, .038060218f, + .009607345f }, + { .038060248f, .146446645f, .308658302f, .500000060f, .691341758f, .853553414f, + .961939812f, 1.00000000f, .961939752f, .853553295f, .691341639f, .500000000f, + .308658272f, .146446615f, .038060218f, .000000000f, .000000000f, .000000000f, + .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, + .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, + .000000000f }, + { .146446645f, .500000060f, .853553414f, 1.00000000f, .853553295f, .500000000f, + .146446615f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, + .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, + .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, + .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, + .000000000f }, + { .500000060f, 1.00000000f, .500000000f, .000000000f, .000000000f, .000000000f, + .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, + .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, + .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, + .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, .000000000f, + .000000000f } +}; + +static const float sb_noise_attenuation[32] = { + 0.0f, 0.0f, 0.3f, 0.4f, 0.5f, 0.7f, 1.0f, 1.0f, + 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, + 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, + 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, 1.0f, +}; + +static const uint8_t fft_subpackets[32] = { + 0, 0, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 0, + 0, 0, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 0, 0 +}; + +/* first index is joined_stereo, second index is 0 or 2 (1 is unused) */ +static float dequant_1bit[2][3] = { + {-0.920000f, 0.000000f, 0.920000f }, + {-0.890000f, 0.000000f, 0.890000f } +}; + +static const float type30_dequant[8] = { + -1.0f,-0.625f,-0.291666656732559f,0.0f, + 0.25f,0.5f,0.75f,1.0f, +}; + +static const float type34_delta[10] = { // FIXME: covers 8 entries.. + -1.0f,-0.60947573184967f,-0.333333343267441f,-0.138071194291115f,0.0f, + 0.138071194291115f,0.333333343267441f,0.60947573184967f,1.0f,0.0f, +}; + +#endif /* QDM2DATA_H */