Mercurial > libavcodec.hg
view qdm2.c @ 7766:10242bdaac34 libavcodec
1000l Fix snow regression test that i broke in r15135.
| author | michael |
|---|---|
| date | Mon, 01 Sep 2008 00:22:00 +0000 |
| parents | 85ab7655ad4d |
| children | 7a463923ecd1 |
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/* * 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 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" #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 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 level; float *samples_im; float *samples_re; 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 { float re; float im; } QDM2Complex; typedef struct { DECLARE_ALIGNED_16(QDM2Complex, complex[256 + 1]); 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 const uint8_t *compressed_data; int compressed_size; float output_buffer[1024]; /// 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 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 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 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 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 (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 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->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 = 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; /* prerotation (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 */ static 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 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 = 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; 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 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 it's > 9 it can overflow s->exptab[] 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_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); } qdm2_init(s); avctx->sample_fmt = SAMPLE_FMT_S16; // 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; } static void qdm2_decode (QDM2Context *q, const 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,"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; } } /* 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, const uint8_t *buf, int buf_size) { QDM2Context *s = avctx->priv_data; if(!buf) return 0; if(buf_size < s->checksum_size) return -1; *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, .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), };