1887 lines
63 KiB
C
Executable file
1887 lines
63 KiB
C
Executable file
/*
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* QDM2 compatible decoder
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* Copyright (c) 2003 Ewald Snel
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* Copyright (c) 2005 Benjamin Larsson
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* Copyright (c) 2005 Alex Beregszaszi
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* Copyright (c) 2005 Roberto Togni
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* QDM2 decoder
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* @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
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*
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* The decoder is not perfect yet, there are still some distortions
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* especially on files encoded with 16 or 8 subbands.
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*/
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#include <math.h>
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#include <stddef.h>
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#include <stdio.h>
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#include "libavutil/channel_layout.h"
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#define BITSTREAM_READER_LE
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#include "avcodec.h"
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#include "get_bits.h"
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#include "bytestream.h"
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#include "internal.h"
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#include "mpegaudio.h"
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#include "mpegaudiodsp.h"
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#include "rdft.h"
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#include "qdm2_tablegen.h"
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#define QDM2_LIST_ADD(list, size, packet) \
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do { \
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if (size > 0) { \
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list[size - 1].next = &list[size]; \
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} \
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list[size].packet = packet; \
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list[size].next = NULL; \
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size++; \
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} while(0)
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// Result is 8, 16 or 30
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#define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
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#define FIX_NOISE_IDX(noise_idx) \
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if ((noise_idx) >= 3840) \
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(noise_idx) -= 3840; \
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#define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
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#define SAMPLES_NEEDED \
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av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
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#define SAMPLES_NEEDED_2(why) \
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av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
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#define QDM2_MAX_FRAME_SIZE 512
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typedef int8_t sb_int8_array[2][30][64];
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/**
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* Subpacket
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*/
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typedef struct QDM2SubPacket {
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int type; ///< subpacket type
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unsigned int size; ///< subpacket size
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const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
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} QDM2SubPacket;
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/**
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* A node in the subpacket list
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*/
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typedef struct QDM2SubPNode {
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QDM2SubPacket *packet; ///< packet
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struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
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} QDM2SubPNode;
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typedef struct QDM2Complex {
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float re;
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float im;
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} QDM2Complex;
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typedef struct FFTTone {
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float level;
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QDM2Complex *complex;
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const float *table;
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int phase;
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int phase_shift;
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int duration;
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short time_index;
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short cutoff;
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} FFTTone;
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typedef struct FFTCoefficient {
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int16_t sub_packet;
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uint8_t channel;
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int16_t offset;
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int16_t exp;
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uint8_t phase;
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} FFTCoefficient;
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typedef struct QDM2FFT {
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DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
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} QDM2FFT;
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/**
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* QDM2 decoder context
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*/
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typedef struct QDM2Context {
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/// Parameters from codec header, do not change during playback
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int nb_channels; ///< number of channels
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int channels; ///< number of channels
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int group_size; ///< size of frame group (16 frames per group)
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int fft_size; ///< size of FFT, in complex numbers
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int checksum_size; ///< size of data block, used also for checksum
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/// Parameters built from header parameters, do not change during playback
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int group_order; ///< order of frame group
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int fft_order; ///< order of FFT (actually fftorder+1)
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int frame_size; ///< size of data frame
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int frequency_range;
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int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
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int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
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int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
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/// Packets and packet lists
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QDM2SubPacket sub_packets[16]; ///< the packets themselves
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QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
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QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
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int sub_packets_B; ///< number of packets on 'B' list
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QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
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QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
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/// FFT and tones
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FFTTone fft_tones[1000];
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int fft_tone_start;
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int fft_tone_end;
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FFTCoefficient fft_coefs[1000];
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int fft_coefs_index;
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int fft_coefs_min_index[5];
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int fft_coefs_max_index[5];
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int fft_level_exp[6];
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RDFTContext rdft_ctx;
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QDM2FFT fft;
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/// I/O data
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const uint8_t *compressed_data;
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int compressed_size;
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float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
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/// Synthesis filter
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MPADSPContext mpadsp;
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DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
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int synth_buf_offset[MPA_MAX_CHANNELS];
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DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
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DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
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/// Mixed temporary data used in decoding
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float tone_level[MPA_MAX_CHANNELS][30][64];
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int8_t coding_method[MPA_MAX_CHANNELS][30][64];
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int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
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int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
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int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
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int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
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int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
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int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
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int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
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// Flags
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int has_errors; ///< packet has errors
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int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
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int do_synth_filter; ///< used to perform or skip synthesis filter
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int sub_packet;
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int noise_idx; ///< index for dithering noise table
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} QDM2Context;
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static const int switchtable[23] = {
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0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4
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};
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static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
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{
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int value;
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value = get_vlc2(gb, vlc->table, vlc->bits, depth);
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/* stage-2, 3 bits exponent escape sequence */
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if (value-- == 0)
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value = get_bits(gb, get_bits(gb, 3) + 1);
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/* stage-3, optional */
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if (flag) {
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int tmp;
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if (value >= 60) {
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av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
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return 0;
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}
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tmp= vlc_stage3_values[value];
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if ((value & ~3) > 0)
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tmp += get_bits(gb, (value >> 2));
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value = tmp;
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}
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return value;
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}
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static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth)
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{
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int value = qdm2_get_vlc(gb, vlc, 0, depth);
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return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
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}
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/**
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* QDM2 checksum
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*
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* @param data pointer to data to be checksummed
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* @param length data length
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* @param value checksum value
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*
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* @return 0 if checksum is OK
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*/
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static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value)
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{
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int i;
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for (i = 0; i < length; i++)
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value -= data[i];
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return (uint16_t)(value & 0xffff);
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}
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/**
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* Fill a QDM2SubPacket structure with packet type, size, and data pointer.
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*
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* @param gb bitreader context
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* @param sub_packet packet under analysis
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*/
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static void qdm2_decode_sub_packet_header(GetBitContext *gb,
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QDM2SubPacket *sub_packet)
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{
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sub_packet->type = get_bits(gb, 8);
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if (sub_packet->type == 0) {
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sub_packet->size = 0;
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sub_packet->data = NULL;
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} else {
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sub_packet->size = get_bits(gb, 8);
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if (sub_packet->type & 0x80) {
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sub_packet->size <<= 8;
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sub_packet->size |= get_bits(gb, 8);
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sub_packet->type &= 0x7f;
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}
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if (sub_packet->type == 0x7f)
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sub_packet->type |= (get_bits(gb, 8) << 8);
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// FIXME: this depends on bitreader-internal data
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sub_packet->data = &gb->buffer[get_bits_count(gb) / 8];
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}
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av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n",
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sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
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}
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/**
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* Return node pointer to first packet of requested type in list.
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*
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* @param list list of subpackets to be scanned
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* @param type type of searched subpacket
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* @return node pointer for subpacket if found, else NULL
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*/
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static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list,
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int type)
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{
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while (list && list->packet) {
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if (list->packet->type == type)
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return list;
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list = list->next;
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}
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return NULL;
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}
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/**
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* Replace 8 elements with their average value.
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* Called by qdm2_decode_superblock before starting subblock decoding.
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*
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* @param q context
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*/
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static void average_quantized_coeffs(QDM2Context *q)
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{
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int i, j, n, ch, sum;
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n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
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for (ch = 0; ch < q->nb_channels; ch++)
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for (i = 0; i < n; i++) {
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sum = 0;
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for (j = 0; j < 8; j++)
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sum += q->quantized_coeffs[ch][i][j];
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sum /= 8;
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if (sum > 0)
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sum--;
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for (j = 0; j < 8; j++)
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q->quantized_coeffs[ch][i][j] = sum;
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}
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}
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/**
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* Build subband samples with noise weighted by q->tone_level.
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* Called by synthfilt_build_sb_samples.
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*
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* @param q context
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* @param sb subband index
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*/
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static void build_sb_samples_from_noise(QDM2Context *q, int sb)
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{
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int ch, j;
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FIX_NOISE_IDX(q->noise_idx);
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if (!q->nb_channels)
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return;
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for (ch = 0; ch < q->nb_channels; ch++) {
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for (j = 0; j < 64; j++) {
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q->sb_samples[ch][j * 2][sb] =
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SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
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q->sb_samples[ch][j * 2 + 1][sb] =
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SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
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}
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}
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}
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/**
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* Called while processing data from subpackets 11 and 12.
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* Used after making changes to coding_method array.
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*
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* @param sb subband index
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* @param channels number of channels
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* @param coding_method q->coding_method[0][0][0]
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*/
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static int fix_coding_method_array(int sb, int channels,
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sb_int8_array coding_method)
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{
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int j, k;
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int ch;
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int run, case_val;
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for (ch = 0; ch < channels; ch++) {
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for (j = 0; j < 64; ) {
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if (coding_method[ch][sb][j] < 8)
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return -1;
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if ((coding_method[ch][sb][j] - 8) > 22) {
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run = 1;
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case_val = 8;
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} else {
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switch (switchtable[coding_method[ch][sb][j] - 8]) {
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case 0: run = 10;
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case_val = 10;
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break;
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case 1: run = 1;
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case_val = 16;
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break;
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case 2: run = 5;
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case_val = 24;
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break;
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case 3: run = 3;
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case_val = 30;
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break;
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case 4: run = 1;
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case_val = 30;
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break;
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case 5: run = 1;
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case_val = 8;
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break;
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default: run = 1;
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case_val = 8;
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break;
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}
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}
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for (k = 0; k < run; k++) {
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if (j + k < 128) {
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int sbjk = sb + (j + k) / 64;
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if (sbjk > 29) {
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SAMPLES_NEEDED
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continue;
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}
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if (coding_method[ch][sbjk][(j + k) % 64] > coding_method[ch][sb][j]) {
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if (k > 0) {
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SAMPLES_NEEDED
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//not debugged, almost never used
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memset(&coding_method[ch][sb][j + k], case_val,
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k *sizeof(int8_t));
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memset(&coding_method[ch][sb][j + k], case_val,
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3 * sizeof(int8_t));
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}
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}
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}
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}
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j += run;
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}
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}
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return 0;
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}
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/**
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* Related to synthesis filter
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* Called by process_subpacket_10
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*
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* @param q context
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* @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
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*/
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static void fill_tone_level_array(QDM2Context *q, int flag)
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{
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int i, sb, ch, sb_used;
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int tmp, tab;
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for (ch = 0; ch < q->nb_channels; ch++)
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for (sb = 0; sb < 30; sb++)
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for (i = 0; i < 8; i++) {
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if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
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tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
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q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
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else
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tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
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if(tmp < 0)
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tmp += 0xff;
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q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
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}
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sb_used = QDM2_SB_USED(q->sub_sampling);
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if ((q->superblocktype_2_3 != 0) && !flag) {
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for (sb = 0; sb < sb_used; sb++)
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for (ch = 0; ch < q->nb_channels; ch++)
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for (i = 0; i < 64; i++) {
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q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
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if (q->tone_level_idx[ch][sb][i] < 0)
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q->tone_level[ch][sb][i] = 0;
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else
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q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
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}
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} else {
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tab = q->superblocktype_2_3 ? 0 : 1;
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for (sb = 0; sb < sb_used; sb++) {
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if ((sb >= 4) && (sb <= 23)) {
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for (ch = 0; ch < q->nb_channels; ch++)
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for (i = 0; i < 64; i++) {
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tmp = q->tone_level_idx_base[ch][sb][i / 8] -
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q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
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q->tone_level_idx_mid[ch][sb - 4][i / 8] -
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q->tone_level_idx_hi2[ch][sb - 4];
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q->tone_level_idx[ch][sb][i] = tmp & 0xff;
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if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
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q->tone_level[ch][sb][i] = 0;
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else
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q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
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|
}
|
|
} 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)
|
|
*
|
|
* @param q context
|
|
*/
|
|
static av_cold void qdm2_init_static_data(void) {
|
|
static int done;
|
|
|
|
if(done)
|
|
return;
|
|
|
|
qdm2_init_vlc();
|
|
ff_mpa_synth_init_float(ff_mpa_synth_window_float);
|
|
softclip_table_init();
|
|
rnd_table_init();
|
|
init_noise_samples();
|
|
|
|
done = 1;
|
|
}
|
|
|
|
/**
|
|
* Init parameters from codec extradata
|
|
*/
|
|
static av_cold int qdm2_decode_init(AVCodecContext *avctx)
|
|
{
|
|
QDM2Context *s = avctx->priv_data;
|
|
int tmp_val, tmp, size;
|
|
GetByteContext gb;
|
|
|
|
qdm2_init_static_data();
|
|
|
|
/* 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;
|
|
|
|
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,
|
|
};
|