diff options
Diffstat (limited to 'ffmpeg1/libavcodec/qdm2.c')
| -rw-r--r-- | ffmpeg1/libavcodec/qdm2.c | 2014 |
1 files changed, 0 insertions, 2014 deletions
diff --git a/ffmpeg1/libavcodec/qdm2.c b/ffmpeg1/libavcodec/qdm2.c deleted file mode 100644 index 108c327..0000000 --- a/ffmpeg1/libavcodec/qdm2.c +++ /dev/null @@ -1,2014 +0,0 @@ -/* - * 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 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 BITSTREAM_READER_LE -#include "libavutil/channel_layout.h" -#include "avcodec.h" -#include "get_bits.h" -#include "internal.h" -#include "rdft.h" -#include "mpegaudiodsp.h" -#include "mpegaudio.h" - -#include "qdm2data.h" -#include "qdm2_tablegen.h" - -#undef NDEBUG -#include <assert.h> - - -#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 SAMPLES_NEEDED \ - av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n"); - -#define SAMPLES_NEEDED_2(why) \ - av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why); - -#define QDM2_MAX_FRAME_SIZE 512 - -typedef int8_t sb_int8_array[2][30][64]; - -/** - * Subpacket - */ -typedef struct { - int type; ///< subpacket type - unsigned int size; ///< subpacket size - const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy) -} QDM2SubPacket; - -/** - * A node in the subpacket list - */ -typedef struct QDM2SubPNode { - QDM2SubPacket *packet; ///< packet - struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node -} QDM2SubPNode; - -typedef struct { - float re; - float im; -} QDM2Complex; - -typedef struct { - float level; - QDM2Complex *complex; - const float *table; - int phase; - int phase_shift; - int duration; - short time_index; - short cutoff; -} FFTTone; - -typedef struct { - int16_t sub_packet; - uint8_t channel; - int16_t offset; - int16_t exp; - uint8_t phase; -} FFTCoefficient; - -typedef struct { - DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256]; -} QDM2FFT; - -/** - * QDM2 decoder context - */ -typedef struct { - /// Parameters from codec header, do not change during playback - int nb_channels; ///< number of channels - int channels; ///< number of channels - int group_size; ///< size of frame group (16 frames per group) - int fft_size; ///< size of FFT, in complex numbers - int checksum_size; ///< size of data block, used also for checksum - - /// Parameters built from header parameters, do not change during playback - int group_order; ///< order of frame group - int fft_order; ///< order of FFT (actually fftorder+1) - int frame_size; ///< size of data frame - int frequency_range; - int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */ - int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2 - int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4) - - /// Packets and packet lists - QDM2SubPacket sub_packets[16]; ///< the packets themselves - QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets - QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list - int sub_packets_B; ///< number of packets on 'B' list - QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors? - QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets - - /// FFT and tones - FFTTone fft_tones[1000]; - int fft_tone_start; - int fft_tone_end; - FFTCoefficient fft_coefs[1000]; - int fft_coefs_index; - int fft_coefs_min_index[5]; - int fft_coefs_max_index[5]; - int fft_level_exp[6]; - RDFTContext rdft_ctx; - QDM2FFT fft; - - /// I/O data - const uint8_t *compressed_data; - int compressed_size; - float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2]; - - /// Synthesis filter - MPADSPContext mpadsp; - DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2]; - int synth_buf_offset[MPA_MAX_CHANNELS]; - DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT]; - DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; - - /// 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 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 const uint16_t qdm2_vlc_offs[] = { - 0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838, -}; - -static av_cold void qdm2_init_vlc(void) -{ - static int vlcs_initialized = 0; - static VLC_TYPE qdm2_table[3838][2]; - - if (!vlcs_initialized) { - - vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]]; - vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0]; - init_vlc (&vlc_tab_level, 8, 24, - vlc_tab_level_huffbits, 1, 1, - vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - - vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]]; - vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1]; - init_vlc (&vlc_tab_diff, 8, 37, - vlc_tab_diff_huffbits, 1, 1, - vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - - vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]]; - vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2]; - init_vlc (&vlc_tab_run, 5, 6, - vlc_tab_run_huffbits, 1, 1, - vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - - fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]]; - fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3]; - 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_NEW_STATIC | INIT_VLC_LE); - - - fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]]; - fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4]; - init_vlc (&fft_level_exp_vlc, 8, 20, - fft_level_exp_huffbits, 1, 1, - fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - - fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]]; - fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5]; - init_vlc (&fft_stereo_exp_vlc, 6, 7, - fft_stereo_exp_huffbits, 1, 1, - fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - - fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]]; - fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6]; - init_vlc (&fft_stereo_phase_vlc, 6, 9, - fft_stereo_phase_huffbits, 1, 1, - fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - - vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]]; - vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7]; - 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_NEW_STATIC | INIT_VLC_LE); - - vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]]; - vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8]; - 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_NEW_STATIC | INIT_VLC_LE); - - vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]]; - vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9]; - 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_NEW_STATIC | INIT_VLC_LE); - - vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]]; - vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10]; - init_vlc (&vlc_tab_type30, 6, 9, - vlc_tab_type30_huffbits, 1, 1, - vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - - vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]]; - vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11]; - init_vlc (&vlc_tab_type34, 5, 10, - vlc_tab_type34_huffbits, 1, 1, - vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE); - - vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]]; - vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12]; - 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_NEW_STATIC | INIT_VLC_LE); - - vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]]; - vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13]; - 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_NEW_STATIC | INIT_VLC_LE); - - vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]]; - vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14]; - 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_NEW_STATIC | INIT_VLC_LE); - - vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]]; - vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15]; - 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_NEW_STATIC | INIT_VLC_LE); - - vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]]; - vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16]; - 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_NEW_STATIC | INIT_VLC_LE); - - vlcs_initialized=1; - } -} - -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; - - if (value >= 60) { - av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value); - return 0; - } - - 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); -} - - -/** - * Fill 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; -} - - -/** - * Replace 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] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j]; - q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j]; - } -} - - -/** - * 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; - static const 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; - - 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; - - if (!superblocktype_2_3) { - /* This case is untested, no samples available */ - avpriv_request_sample(NULL, "!superblocktype_2_3"); - return; - 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]; - - multres = 0x66666667LL * (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 int 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 0; - } - - 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 = (get_bits_left(gb) >= 1) ? get_bits1 (gb) : 0; - - if (joined_stereo) { - if (get_bits_left(gb) >= 16) - for (j = 0; j < 16; j++) - sign_bits[j] = get_bits1 (gb); - - if (q->coding_method[0][sb][0] <= 0) { - av_log(NULL, AV_LOG_ERROR, "coding method invalid\n"); - return AVERROR_INVALIDDATA; - } - - 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 = (get_bits_left(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 (get_bits_left(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); - if (n >= 243) { - av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n"); - return AVERROR_INVALIDDATA; - } - - 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 (get_bits_left(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 (get_bits_left(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); - if (n >= 243) { - av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n"); - return AVERROR_INVALIDDATA; - } - - 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 (get_bits_left(gb) >= 7) { - n = get_bits(gb, 7); - if (n >= 125) { - av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n"); - return AVERROR_INVALIDDATA; - } - - 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 (get_bits_left(gb) >= 4) { - unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1); - if (index >= FF_ARRAY_ELEMS(type30_dequant)) { - av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index); - return AVERROR_INVALIDDATA; - } - samples[0] = type30_dequant[index]; - } else - samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); - - run = 1; - break; - - case 34: - if (get_bits_left(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 { - unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1); - if (index >= FF_ARRAY_ELEMS(type34_delta)) { - av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index); - return AVERROR_INVALIDDATA; - } - samples[0] = type34_delta[index] / 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]; - if ((j + k) < 128) - 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] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs]; - } else { - for (k = 0; k < run; k++) - if ((j + k) < 128) - q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k]; - } - - j += run; - } // j loop - } // channel loop - } // subband loop - return 0; -} - - -/** - * 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 quantized_coeffs pointer to quantized_coeffs[ch][0] - * @param gb bitreader context - */ -static int init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb) -{ - int i, k, run, level, diff; - - if (get_bits_left(gb) < 16) - return -1; - level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); - - quantized_coeffs[0] = level; - - for (i = 0; i < 7; ) { - if (get_bits_left(gb) < 16) - return -1; - run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; - - if (i + run >= 8) - return -1; - - if (get_bits_left(gb) < 16) - return -1; - 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; - } - return 0; -} - - -/** - * 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 - */ -static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb) -{ - int sb, j, k, n, ch; - - for (ch = 0; ch < q->nb_channels; ch++) { - init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb); - - if (get_bits_left(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 (get_bits_left(gb) < 1) - break; - if (get_bits1(gb)) { - for (k=0; k < 8; k++) { - if (get_bits_left(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 (get_bits_left(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 (get_bits_left(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 int 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); - - if (j + run >= 8) - return -1; - - 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; - - return 0; -} - - -/** - * Process subpacket 10 if not null, else - * - * @param q context - * @param node pointer to node with packet - */ -static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node) -{ - GetBitContext gb; - - if (node) { - init_get_bits(&gb, node->packet->data, node->packet->size * 8); - init_tone_level_dequantization(q, &gb); - 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 - */ -static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node) -{ - GetBitContext gb; - int length = 0; - - if (node) { - length = node->packet->size * 8; - init_get_bits(&gb, node->packet->data, length); - } - - 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 - */ -static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node) -{ - GetBitContext gb; - int length = 0; - - if (node) { - length = node->packet->size * 8; - init_get_bits(&gb, node->packet->data, length); - } - - 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]); - else - process_subpacket_10(q, NULL); - - 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]); - else - process_subpacket_11(q, NULL); - - 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]); - else - process_subpacket_12(q, NULL); -} - - -/** - * 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); - csum += 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; - - if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) { - SAMPLES_NEEDED_2("too many packet bytes"); - return; - } - - 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); - process_subpacket_11(q, NULL); - process_subpacket_12(q, NULL); - } -/* **************************************************************** */ -} - - -static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet, - int offset, int duration, int channel, - int exp, int phase) -{ - if (q->fft_coefs_min_index[duration] < 0) - q->fft_coefs_min_index[duration] = q->fft_coefs_index; - - q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); - q->fft_coefs[q->fft_coefs_index].channel = channel; - q->fft_coefs[q->fft_coefs_index].offset = offset; - q->fft_coefs[q->fft_coefs_index].exp = exp; - q->fft_coefs[q->fft_coefs_index].phase = phase; - q->fft_coefs_index++; -} - - -static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b) -{ - int channel, stereo, phase, exp; - int local_int_4, local_int_8, stereo_phase, local_int_10; - int local_int_14, stereo_exp, local_int_20, local_int_28; - int n, offset; - - local_int_4 = 0; - local_int_28 = 0; - local_int_20 = 2; - local_int_8 = (4 - duration); - local_int_10 = 1 << (q->group_order - duration - 1); - offset = 1; - - while (get_bits_left(gb)>0) { - if (q->superblocktype_2_3) { - while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { - if (get_bits_left(gb)<0) { - if(local_int_4 < q->group_size) - av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n"); - return; - } - offset = 1; - if (n == 0) { - local_int_4 += local_int_10; - local_int_28 += (1 << local_int_8); - } else { - local_int_4 += 8*local_int_10; - local_int_28 += (8 << local_int_8); - } - } - offset += (n - 2); - } else { - offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2); - while (offset >= (local_int_10 - 1)) { - offset += (1 - (local_int_10 - 1)); - local_int_4 += local_int_10; - local_int_28 += (1 << local_int_8); - } - } - - if (local_int_4 >= q->group_size) - return; - - local_int_14 = (offset >> local_int_8); - if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table)) - return; - - if (q->nb_channels > 1) { - channel = get_bits1(gb); - stereo = get_bits1(gb); - } else { - channel = 0; - stereo = 0; - } - - exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); - exp += q->fft_level_exp[fft_level_index_table[local_int_14]]; - exp = (exp < 0) ? 0 : exp; - - phase = get_bits(gb, 3); - stereo_exp = 0; - stereo_phase = 0; - - if (stereo) { - stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1)); - stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1)); - if (stereo_phase < 0) - stereo_phase += 8; - } - - if (q->frequency_range > (local_int_14 + 1)) { - int sub_packet = (local_int_20 + local_int_28); - - qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase); - if (stereo) - qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase); - } - - offset++; - } -} - - -static void qdm2_decode_fft_packets (QDM2Context *q) -{ - int i, j, min, max, value, type, unknown_flag; - GetBitContext gb; - - if (q->sub_packet_list_B[0].packet == NULL) - return; - - /* reset minimum indexes for FFT coefficients */ - q->fft_coefs_index = 0; - for (i=0; i < 5; i++) - q->fft_coefs_min_index[i] = -1; - - /* process subpackets ordered by type, largest type first */ - for (i = 0, max = 256; i < q->sub_packets_B; i++) { - QDM2SubPacket *packet= NULL; - - /* find subpacket with largest type less than max */ - for (j = 0, min = 0; j < q->sub_packets_B; j++) { - value = q->sub_packet_list_B[j].packet->type; - if (value > min && value < max) { - min = value; - packet = q->sub_packet_list_B[j].packet; - } - } - - max = min; - - /* check for errors (?) */ - if (!packet) - return; - - if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16])) - return; - - /* decode FFT tones */ - init_get_bits (&gb, packet->data, packet->size*8); - - if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) - unknown_flag = 1; - else - unknown_flag = 0; - - type = packet->type; - - if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) { - int duration = q->sub_sampling + 5 - (type & 15); - - if (duration >= 0 && duration < 4) - qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); - } else if (type == 31) { - for (j=0; j < 4; j++) - qdm2_fft_decode_tones(q, j, &gb, unknown_flag); - } else if (type == 46) { - for (j=0; j < 6; j++) - q->fft_level_exp[j] = get_bits(&gb, 6); - for (j=0; j < 4; j++) - qdm2_fft_decode_tones(q, j, &gb, unknown_flag); - } - } // Loop on B packets - - /* calculate maximum indexes for FFT coefficients */ - for (i = 0, j = -1; i < 5; i++) - if (q->fft_coefs_min_index[i] >= 0) { - if (j >= 0) - q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i]; - j = i; - } - if (j >= 0) - q->fft_coefs_max_index[j] = q->fft_coefs_index; -} - - -static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone) -{ - float level, f[6]; - int i; - QDM2Complex c; - const double iscale = 2.0*M_PI / 512.0; - - tone->phase += tone->phase_shift; - - /* calculate current level (maximum amplitude) of tone */ - level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level; - c.im = level * sin(tone->phase*iscale); - c.re = level * cos(tone->phase*iscale); - - /* generate FFT coefficients for tone */ - if (tone->duration >= 3 || tone->cutoff >= 3) { - tone->complex[0].im += c.im; - tone->complex[0].re += c.re; - tone->complex[1].im -= c.im; - tone->complex[1].re -= c.re; - } else { - f[1] = -tone->table[4]; - f[0] = tone->table[3] - tone->table[0]; - f[2] = 1.0 - tone->table[2] - tone->table[3]; - f[3] = tone->table[1] + tone->table[4] - 1.0; - f[4] = tone->table[0] - tone->table[1]; - f[5] = tone->table[2]; - for (i = 0; i < 2; i++) { - tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i]; - tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]); - } - for (i = 0; i < 4; i++) { - tone->complex[i].re += c.re * f[i+2]; - tone->complex[i].im += c.im * f[i+2]; - } - } - - /* copy the tone if it has not yet died out */ - if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) { - memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone)); - q->fft_tone_end = (q->fft_tone_end + 1) % 1000; - } -} - - -static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet) -{ - int i, j, ch; - const double iscale = 0.25 * M_PI; - - for (ch = 0; ch < q->channels; ch++) { - memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex)); - } - - - /* apply FFT tones with duration 4 (1 FFT period) */ - if (q->fft_coefs_min_index[4] >= 0) - for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) { - float level; - QDM2Complex c; - - if (q->fft_coefs[i].sub_packet != sub_packet) - break; - - ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel; - level = (q->fft_coefs[i].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[i].exp & 63]; - - c.re = level * cos(q->fft_coefs[i].phase * iscale); - c.im = level * sin(q->fft_coefs[i].phase * iscale); - q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re; - q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im; - q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re; - q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im; - } - - /* generate existing FFT tones */ - for (i = q->fft_tone_end; i != q->fft_tone_start; ) { - qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]); - q->fft_tone_start = (q->fft_tone_start + 1) % 1000; - } - - /* create and generate new FFT tones with duration 0 (long) to 3 (short) */ - for (i = 0; i < 4; i++) - if (q->fft_coefs_min_index[i] >= 0) { - for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) { - int offset, four_i; - FFTTone tone; - - if (q->fft_coefs[j].sub_packet != sub_packet) - break; - - four_i = (4 - i); - offset = q->fft_coefs[j].offset >> four_i; - ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel; - - if (offset < q->frequency_range) { - if (offset < 2) - tone.cutoff = offset; - else - tone.cutoff = (offset >= 60) ? 3 : 2; - - tone.level = (q->fft_coefs[j].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[j].exp & 63]; - tone.complex = &q->fft.complex[ch][offset]; - tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)]; - tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128; - tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i); - tone.duration = i; - tone.time_index = 0; - - qdm2_fft_generate_tone(q, &tone); - } - } - q->fft_coefs_min_index[i] = j; - } -} - - -static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet) -{ - const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f; - float *out = q->output_buffer + channel; - int i; - q->fft.complex[channel][0].re *= 2.0f; - q->fft.complex[channel][0].im = 0.0f; - q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]); - /* add samples to output buffer */ - for (i = 0; i < FFALIGN(q->fft_size, 8); i++) { - out[0] += q->fft.complex[channel][i].re * gain; - out[q->channels] += q->fft.complex[channel][i].im * gain; - out += 2 * q->channels; - } -} - - -/** - * @param q context - * @param index subpacket number - */ -static void qdm2_synthesis_filter (QDM2Context *q, int index) -{ - 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++) { - float *samples_ptr = q->samples + ch; - - for (i = 0; i < 8; i++) { - ff_mpa_synth_filter_float(&q->mpadsp, - q->synth_buf[ch], &(q->synth_buf_offset[ch]), - ff_mpa_synth_window_float, &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] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch]; -} - - -/** - * Init static data (does not depend on specific file) - * - * @param q context - */ -static av_cold void qdm2_init(QDM2Context *q) { - static int initialized = 0; - - if (initialized != 0) - return; - initialized = 1; - - qdm2_init_vlc(); - ff_mpa_synth_init_float(ff_mpa_synth_window_float); - softclip_table_init(); - rnd_table_init(); - init_noise_samples(); - - av_log(NULL, AV_LOG_DEBUG, "init done\n"); -} - - -/** - * Init parameters from codec extradata - */ -static av_cold int qdm2_decode_init(AVCodecContext *avctx) -{ - QDM2Context *s = avctx->priv_data; - uint8_t *extradata; - int extradata_size; - int tmp_val, tmp, size; - - /* extradata parsing - - Structure: - wave { - frma (QDM2) - QDCA - QDCP - } - - 32 size (including this field) - 32 tag (=frma) - 32 type (=QDM2 or QDMC) - - 32 size (including this field, in bytes) - 32 tag (=QDCA) // maybe mandatory parameters - 32 unknown (=1) - 32 channels (=2) - 32 samplerate (=44100) - 32 bitrate (=96000) - 32 block size (=4096) - 32 frame size (=256) (for one channel) - 32 packet size (=1300) - - 32 size (including this field, in bytes) - 32 tag (=QDCP) // maybe some tuneable parameters - 32 float1 (=1.0) - 32 zero ? - 32 float2 (=1.0) - 32 float3 (=1.0) - 32 unknown (27) - 32 unknown (8) - 32 zero ? - */ - - if (!avctx->extradata || (avctx->extradata_size < 48)) { - av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n"); - return -1; - } - - extradata = avctx->extradata; - extradata_size = avctx->extradata_size; - - while (extradata_size > 7) { - if (!memcmp(extradata, "frmaQDM", 7)) - break; - extradata++; - extradata_size--; - } - - if (extradata_size < 12) { - av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n", - extradata_size); - return -1; - } - - if (memcmp(extradata, "frmaQDM", 7)) { - av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n"); - return -1; - } - - if (extradata[7] == 'C') { -// s->is_qdmc = 1; - av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n"); - return -1; - } - - extradata += 8; - extradata_size -= 8; - - size = AV_RB32(extradata); - - if(size > extradata_size){ - av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n", - extradata_size, size); - return -1; - } - - extradata += 4; - av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size); - if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) { - av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n"); - return -1; - } - - extradata += 8; - - avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata); - extradata += 4; - if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) { - av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n"); - return AVERROR_INVALIDDATA; - } - avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO : - AV_CH_LAYOUT_MONO; - - 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); - if (s->checksum_size >= 1U << 28) { - av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size); - return AVERROR_INVALIDDATA; - } - - s->fft_order = av_log2(s->fft_size) + 1; - - // something like max decodable tones - s->group_order = av_log2(s->group_size) + 1; - s->frame_size = s->group_size / 16; // 16 iterations per super block - - if (s->frame_size > QDM2_MAX_FRAME_SIZE) - return AVERROR_INVALIDDATA; - - s->sub_sampling = s->fft_order - 7; - s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); - - switch ((s->sub_sampling * 2 + s->channels - 1)) { - case 0: tmp = 40; break; - case 1: tmp = 48; break; - case 2: tmp = 56; break; - case 3: tmp = 72; break; - case 4: tmp = 80; break; - case 5: tmp = 100;break; - default: tmp=s->sub_sampling; break; - } - tmp_val = 0; - if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1; - if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2; - if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3; - if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4; - s->cm_table_select = tmp_val; - - if (s->sub_sampling == 0) - tmp = 7999; - else - tmp = ((-(s->sub_sampling -1)) & 8000) + 20000; - /* - 0: 7999 -> 0 - 1: 20000 -> 2 - 2: 28000 -> 2 - */ - if (tmp < 8000) - s->coeff_per_sb_select = 0; - else if (tmp <= 16000) - s->coeff_per_sb_select = 1; - else - s->coeff_per_sb_select = 2; - - // Fail on unknown fft order - if ((s->fft_order < 7) || (s->fft_order > 9)) { - av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order); - return -1; - } - - ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R); - ff_mpadsp_init(&s->mpadsp); - - qdm2_init(s); - - avctx->sample_fmt = AV_SAMPLE_FMT_S16; - - return 0; -} - - -static av_cold int qdm2_decode_close(AVCodecContext *avctx) -{ - QDM2Context *s = avctx->priv_data; - - ff_rdft_end(&s->rdft_ctx); - - return 0; -} - - -static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out) -{ - int ch, i; - const int frame_size = (q->frame_size * q->channels); - - if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2) - return -1; - - /* select input buffer */ - q->compressed_data = in; - q->compressed_size = q->checksum_size; - - /* copy old block, clear new block of output samples */ - memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float)); - memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float)); - - /* decode block of QDM2 compressed data */ - if (q->sub_packet == 0) { - q->has_errors = 0; // zero it for a new super block - av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n"); - qdm2_decode_super_block(q); - } - - /* parse subpackets */ - if (!q->has_errors) { - if (q->sub_packet == 2) - qdm2_decode_fft_packets(q); - - qdm2_fft_tone_synthesizer(q, q->sub_packet); - } - - /* sound synthesis stage 1 (FFT) */ - for (ch = 0; ch < q->channels; ch++) { - qdm2_calculate_fft(q, ch, q->sub_packet); - - if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) { - SAMPLES_NEEDED_2("has errors, and C list is not empty") - return -1; - } - } - - /* sound synthesis stage 2 (MPEG audio like synthesis filter) */ - if (!q->has_errors && q->do_synth_filter) - qdm2_synthesis_filter(q, q->sub_packet); - - q->sub_packet = (q->sub_packet + 1) % 16; - - /* clip and convert output float[] to 16bit signed samples */ - for (i = 0; i < frame_size; i++) { - int value = (int)q->output_buffer[i]; - - if (value > SOFTCLIP_THRESHOLD) - value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD]; - else if (value < -SOFTCLIP_THRESHOLD) - value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD]; - - out[i] = value; - } - - return 0; -} - - -static int qdm2_decode_frame(AVCodecContext *avctx, void *data, - int *got_frame_ptr, AVPacket *avpkt) -{ - AVFrame *frame = data; - const uint8_t *buf = avpkt->data; - int buf_size = avpkt->size; - QDM2Context *s = avctx->priv_data; - int16_t *out; - int i, ret; - - if(!buf) - return 0; - if(buf_size < s->checksum_size) - return -1; - - /* get output buffer */ - frame->nb_samples = 16 * s->frame_size; - if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) - return ret; - out = (int16_t *)frame->data[0]; - - for (i = 0; i < 16; i++) { - if (qdm2_decode(s, buf, out) < 0) - return -1; - out += s->channels * s->frame_size; - } - - *got_frame_ptr = 1; - - return s->checksum_size; -} - -AVCodec ff_qdm2_decoder = -{ - .name = "qdm2", - .type = AVMEDIA_TYPE_AUDIO, - .id = AV_CODEC_ID_QDM2, - .priv_data_size = sizeof(QDM2Context), - .init = qdm2_decode_init, - .close = qdm2_decode_close, - .decode = qdm2_decode_frame, - .capabilities = CODEC_CAP_DR1, - .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), -}; |