| /* | |
| * 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 "libavutil/channel_layout.h" | |
| #include "libavutil/mem_internal.h" | |
| #include "libavutil/thread.h" | |
| #include "libavutil/tx.h" | |
| #define BITSTREAM_READER_LE | |
| #include "avcodec.h" | |
| #include "get_bits.h" | |
| #include "bytestream.h" | |
| #include "codec_internal.h" | |
| #include "decode.h" | |
| #include "mpegaudio.h" | |
| #include "mpegaudiodsp.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 FFTTone { | |
| float level; | |
| AVComplexFloat *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, AVComplexFloat, complex)[MPA_MAX_CHANNELS][256 + 1]; | |
| DECLARE_ALIGNED(32, AVComplexFloat, temp)[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]; | |
| AVTXContext *rdft_ctx; | |
| av_tx_fn rdft_fn; | |
| 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; | |
| AVComplexFloat 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(AVComplexFloat)); | |
| } | |
| /* 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; | |
| AVComplexFloat 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; | |
| q->fft.complex[channel][0].re *= 2.0f; | |
| q->fft.complex[channel][0].im = 0.0f; | |
| q->fft.complex[channel][q->fft_size].re = 0.0f; | |
| q->fft.complex[channel][q->fft_size].im = 0.0f; | |
| q->rdft_fn(q->rdft_ctx, q->fft.temp[channel], q->fft.complex[channel], | |
| sizeof(AVComplexFloat)); | |
| /* add samples to output buffer */ | |
| for (int i = 0; i < FFALIGN(q->fft_size, 8); i++) { | |
| out[0] += q->fft.temp[channel][i].re * gain; | |
| out[q->channels] += q->fft.temp[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 ret, tmp_val, tmp, size; | |
| float scale = 1.0f / 2.0f; | |
| 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); | |
| 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; | |
| } | |
| av_channel_layout_uninit(&avctx->ch_layout); | |
| av_channel_layout_default(&avctx->ch_layout, s->channels); | |
| 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; | |
| } | |
| ret = av_tx_init(&s->rdft_ctx, &s->rdft_fn, AV_TX_FLOAT_RDFT, 1, 2*s->fft_size, &scale, 0); | |
| if (ret < 0) | |
| return ret; | |
| 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; | |
| av_tx_uninit(&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, AVFrame *frame, | |
| int *got_frame_ptr, AVPacket *avpkt) | |
| { | |
| 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; | |
| } | |
| const FFCodec ff_qdm2_decoder = { | |
| .p.name = "qdm2", | |
| CODEC_LONG_NAME("QDesign Music Codec 2"), | |
| .p.type = AVMEDIA_TYPE_AUDIO, | |
| .p.id = AV_CODEC_ID_QDM2, | |
| .priv_data_size = sizeof(QDM2Context), | |
| .init = qdm2_decode_init, | |
| .close = qdm2_decode_close, | |
| FF_CODEC_DECODE_CB(qdm2_decode_frame), | |
| .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF, | |
| }; |