From 150c9823e71a161e97003849cf8b2f55b21520bd Mon Sep 17 00:00:00 2001 From: Tim Redfern Date: Mon, 26 Aug 2013 15:10:18 +0100 Subject: adding ffmpeg specific version --- ffmpeg1/libavcodec/qdm2.c | 2014 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 2014 insertions(+) create mode 100644 ffmpeg1/libavcodec/qdm2.c (limited to 'ffmpeg1/libavcodec/qdm2.c') diff --git a/ffmpeg1/libavcodec/qdm2.c b/ffmpeg1/libavcodec/qdm2.c new file mode 100644 index 0000000..108c327 --- /dev/null +++ b/ffmpeg1/libavcodec/qdm2.c @@ -0,0 +1,2014 @@ +/* + * 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 +#include +#include + +#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 + + +#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"), +}; -- cgit v1.2.3