Mercurial > libavcodec.hg
view qdm2.c @ 5876:731ee5ad6bde libavcodec
Correct assignment of interlaced_frame; was being set on output frames,
in display order, based on decoding information in decoding order. Now
set properly, immediately upon completion of decode.
Based on original patch from Reinhard Nissl, rnisssl % gmx , de
Original Thread: [FFmpeg-devel] H.264 + PAFF: BBC HD recording shows
extreme interlacing artefacts, Thu, 01 Nov 2007 22:43:09
| author | heydowns |
|---|---|
| date | Mon, 05 Nov 2007 18:16:42 +0000 |
| parents | 2b72f9bc4f06 |
| children | 61f95f3a62e0 |
line wrap: on
line source
/* * 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; 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 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 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,"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 < 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 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 indices 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; /* find subpacket 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*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 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; /* 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 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 = 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); // 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, 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, 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, };