| /* |
| * 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> |
| |
| #include "libavutil/channel_layout.h" |
| #include "libavutil/thread.h" |
| |
| #define BITSTREAM_READER_LE |
| #include "avcodec.h" |
| #include "get_bits.h" |
| #include "bytestream.h" |
| #include "internal.h" |
| #include "mpegaudio.h" |
| #include "mpegaudiodsp.h" |
| #include "rdft.h" |
| |
| #include "qdm2_tablegen.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 QDM2SubPacket { |
| 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 QDM2Complex { |
| float re; |
| float im; |
| } QDM2Complex; |
| |
| typedef struct FFTTone { |
| float level; |
| QDM2Complex *complex; |
| const float *table; |
| int phase; |
| int phase_shift; |
| int duration; |
| short time_index; |
| short cutoff; |
| } FFTTone; |
| |
| typedef struct FFTCoefficient { |
| int16_t sub_packet; |
| uint8_t channel; |
| int16_t offset; |
| int16_t exp; |
| uint8_t phase; |
| } FFTCoefficient; |
| |
| typedef struct QDM2FFT { |
| DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256]; |
| } QDM2FFT; |
| |
| /** |
| * QDM2 decoder context |
| */ |
| typedef struct QDM2Context { |
| /// 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 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 |
| }; |
| |
| static int qdm2_get_vlc(GetBitContext *gb, const 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(const 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 checksummed |
| * @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); |
| |
| // FIXME: this depends on bitreader-internal data |
| sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; |
| } |
| |
| 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 && list->packet) { |
| 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 int fix_coding_method_array(int sb, int channels, |
| sb_int8_array coding_method) |
| { |
| int j, k; |
| int ch; |
| int run, case_val; |
| |
| for (ch = 0; ch < channels; ch++) { |
| for (j = 0; j < 64; ) { |
| if (coding_method[ch][sb][j] < 8) |
| return -1; |
| 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) { |
| int sbjk = sb + (j + k) / 64; |
| if (sbjk > 29) { |
| SAMPLES_NEEDED |
| continue; |
| } |
| if (coding_method[ch][sbjk][(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; |
| } |
| } |
| return 0; |
| } |
| |
| /** |
| * 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]; |
| } |
| } |
| } |
| } |
| } |
| } |
| |
| /** |
| * 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]; |
| } |
| } |
| |
| /** |
| * 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; |
| int type34_first; |
| float type34_div = 0; |
| float type34_predictor; |
| float samples[10]; |
| int sign_bits[16] = {0}; |
| |
| 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++) { |
| 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); |
| |
| 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]; |
| |
| if (fix_coding_method_array(sb, q->nb_channels, |
| q->coding_method)) { |
| av_log(NULL, AV_LOG_ERROR, "coding method invalid\n"); |
| build_sb_samples_from_noise(q, sb); |
| continue; |
| } |
| channels = 1; |
| } |
| |
| for (ch = 0; ch < channels; ch++) { |
| FIX_NOISE_IDX(q->noise_idx); |
| 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) { |
| for (k = 0; k < run && j + k < 128; k++) { |
| q->sb_samples[0][j + k][sb] = |
| q->tone_level[0][sb][(j + k) / 2] * samples[k]; |
| if (q->nb_channels == 2) { |
| if (sign_bits[(j + k) / 8]) |
| q->sb_samples[1][j + k][sb] = |
| q->tone_level[1][sb][(j + k) / 2] * -samples[k]; |
| else |
| q->sb_samples[1][j + k][sb] = |
| q->tone_level[1][sb][(j + k) / 2] * samples[k]; |
| } |
| } |
| } 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; |
| |
| 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]) |
| process_subpacket_9(q, nodes[0]); |
| |
| nodes[1] = qdm2_search_subpacket_type_in_list(list, 10); |
| if (nodes[1]) |
| 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] && nodes[1] && nodes[2]) |
| 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] && nodes[1] && nodes[3]) |
| 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) { |
| 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 { |
| if (local_int_10 <= 2) { |
| av_log(NULL, AV_LOG_ERROR, "qdm2_fft_decode_tones() stuck\n"); |
| return; |
| } |
| 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); |
| |
| if (q->fft_coefs_index + stereo >= FF_ARRAY_ELEMS(q->fft_coefs)) |
| return; |
| |
| 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) |
| 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) |
| */ |
| static av_cold void qdm2_init_static_data(void) { |
| qdm2_init_vlc(); |
| softclip_table_init(); |
| rnd_table_init(); |
| init_noise_samples(); |
| |
| ff_mpa_synth_init_float(); |
| } |
| |
| /** |
| * Init parameters from codec extradata |
| */ |
| static av_cold int qdm2_decode_init(AVCodecContext *avctx) |
| { |
| static AVOnce init_static_once = AV_ONCE_INIT; |
| QDM2Context *s = avctx->priv_data; |
| int tmp_val, tmp, size; |
| GetByteContext gb; |
| |
| /* 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 AVERROR_INVALIDDATA; |
| } |
| |
| bytestream2_init(&gb, avctx->extradata, avctx->extradata_size); |
| |
| while (bytestream2_get_bytes_left(&gb) > 8) { |
| if (bytestream2_peek_be64(&gb) == (((uint64_t)MKBETAG('f','r','m','a') << 32) | |
| (uint64_t)MKBETAG('Q','D','M','2'))) |
| break; |
| bytestream2_skip(&gb, 1); |
| } |
| |
| if (bytestream2_get_bytes_left(&gb) < 12) { |
| av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n", |
| bytestream2_get_bytes_left(&gb)); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| bytestream2_skip(&gb, 8); |
| size = bytestream2_get_be32(&gb); |
| |
| if (size > bytestream2_get_bytes_left(&gb)) { |
| av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n", |
| bytestream2_get_bytes_left(&gb), size); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size); |
| if (bytestream2_get_be32(&gb) != MKBETAG('Q','D','C','A')) { |
| av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| bytestream2_skip(&gb, 4); |
| |
| avctx->channels = s->nb_channels = s->channels = bytestream2_get_be32(&gb); |
| 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 = bytestream2_get_be32(&gb); |
| avctx->bit_rate = bytestream2_get_be32(&gb); |
| s->group_size = bytestream2_get_be32(&gb); |
| s->fft_size = bytestream2_get_be32(&gb); |
| s->checksum_size = bytestream2_get_be32(&gb); |
| if (s->checksum_size >= 1U << 28 || s->checksum_size <= 1) { |
| av_log(avctx, AV_LOG_ERROR, "data block size invalid (%u)\n", s->checksum_size); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| s->fft_order = av_log2(s->fft_size) + 1; |
| |
| // Fail on unknown fft order |
| if ((s->fft_order < 7) || (s->fft_order > 9)) { |
| avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order); |
| return AVERROR_PATCHWELCOME; |
| } |
| |
| // 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)); |
| |
| if (s->frame_size * 4 >> s->sub_sampling > MPA_FRAME_SIZE) { |
| avpriv_request_sample(avctx, "large frames"); |
| return AVERROR_PATCHWELCOME; |
| } |
| |
| 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 (avctx->bit_rate <= 8000) |
| s->coeff_per_sb_select = 0; |
| else if (avctx->bit_rate < 16000) |
| s->coeff_per_sb_select = 1; |
| else |
| s->coeff_per_sb_select = 2; |
| |
| if (s->fft_size != (1 << (s->fft_order - 1))) { |
| av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R); |
| ff_mpadsp_init(&s->mpadsp); |
| |
| avctx->sample_fmt = AV_SAMPLE_FMT_S16; |
| |
| ff_thread_once(&init_static_once, qdm2_init_static_data); |
| |
| 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) { |
| 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 16-bit 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 ((ret = qdm2_decode(s, buf, out)) < 0) |
| return ret; |
| out += s->channels * s->frame_size; |
| } |
| |
| *got_frame_ptr = 1; |
| |
| return s->checksum_size; |
| } |
| |
| AVCodec ff_qdm2_decoder = { |
| .name = "qdm2", |
| .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), |
| .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 = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF, |
| .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, |
| }; |
