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ffmpeg / libavcodec / qdm2.c @ 621d7fe9

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/*
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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 qdm2.c
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 * QDM2 decoder
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 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
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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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#define ALT_BITSTREAM_READER_LE
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#include "avcodec.h"
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#include "bitstream.h"
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#include "dsputil.h"
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#ifdef CONFIG_MPEGAUDIO_HP
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#define USE_HIGHPRECISION
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#endif
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#include "mpegaudio.h"
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#include "qdm2data.h"
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#undef NDEBUG
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#include <assert.h>
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53

    
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#define SOFTCLIP_THRESHOLD 27600
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#define HARDCLIP_THRESHOLD 35716
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57

    
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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 BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
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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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82
#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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85

    
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typedef int8_t sb_int8_array[2][30][64];
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/**
89
 * Subpacket
90
 */
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typedef struct {
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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
99
 */
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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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105
typedef struct {
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    float level;
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    float *samples_im;
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    float *samples_re;
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    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 {
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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;
124

    
125
typedef struct {
126
    float re;
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    float im;
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} QDM2Complex;
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130
typedef struct {
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    DECLARE_ALIGNED_16(QDM2Complex, complex[256 + 1]);
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    float       samples_im[MPA_MAX_CHANNELS][256];
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    float       samples_re[MPA_MAX_CHANNELS][256];
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} QDM2FFT;
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136
/**
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 * QDM2 decoder context
138
 */
139
typedef struct {
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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 fft_frame_size;      ///< size of fft frame, in components (1 comples = re + im)
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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
158
    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
166
    FFTTone fft_tones[1000];
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    int fft_tone_start;
168
    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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    FFTContext fft_ctx;
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    FFTComplex exptab[128];
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    QDM2FFT fft;
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    /// I/O data
179
    uint8_t *compressed_data;
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    int compressed_size;
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    float output_buffer[1024];
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    /// Synthesis filter
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    DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512*2]);
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    int synth_buf_offset[MPA_MAX_CHANNELS];
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    DECLARE_ALIGNED_16(int32_t, sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT]);
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    /// Mixed temporary data used in decoding
189
    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
200
    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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204
    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 uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
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211
static VLC vlc_tab_level;
212
static VLC vlc_tab_diff;
213
static VLC vlc_tab_run;
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static VLC fft_level_exp_alt_vlc;
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static VLC fft_level_exp_vlc;
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static VLC fft_stereo_exp_vlc;
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static VLC fft_stereo_phase_vlc;
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static VLC vlc_tab_tone_level_idx_hi1;
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static VLC vlc_tab_tone_level_idx_mid;
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static VLC vlc_tab_tone_level_idx_hi2;
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static VLC vlc_tab_type30;
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static VLC vlc_tab_type34;
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static VLC vlc_tab_fft_tone_offset[5];
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static uint16_t softclip_table[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1];
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static float noise_table[4096];
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static uint8_t random_dequant_index[256][5];
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static uint8_t random_dequant_type24[128][3];
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static float noise_samples[128];
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231
static DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]);
232

    
233

    
234
static void softclip_table_init(void) {
235
    int i;
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    double dfl = SOFTCLIP_THRESHOLD - 32767;
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    float delta = 1.0 / -dfl;
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    for (i = 0; i < HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1; i++)
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        softclip_table[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF);
240
}
241

    
242

    
243
// random generated table
244
static void rnd_table_init(void) {
245
    int i,j;
246
    uint32_t ldw,hdw;
247
    uint64_t tmp64_1;
248
    uint64_t random_seed = 0;
249
    float delta = 1.0 / 16384.0;
250
    for(i = 0; i < 4096 ;i++) {
251
        random_seed = random_seed * 214013 + 2531011;
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        noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3;
253
    }
254

    
255
    for (i = 0; i < 256 ;i++) {
256
        random_seed = 81;
257
        ldw = i;
258
        for (j = 0; j < 5 ;j++) {
259
            random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
260
            ldw = (uint32_t)ldw % (uint32_t)random_seed;
261
            tmp64_1 = (random_seed * 0x55555556);
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            hdw = (uint32_t)(tmp64_1 >> 32);
263
            random_seed = (uint64_t)(hdw + (ldw >> 31));
264
        }
265
    }
266
    for (i = 0; i < 128 ;i++) {
267
        random_seed = 25;
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        ldw = i;
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        for (j = 0; j < 3 ;j++) {
270
            random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
271
            ldw = (uint32_t)ldw % (uint32_t)random_seed;
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            tmp64_1 = (random_seed * 0x66666667);
273
            hdw = (uint32_t)(tmp64_1 >> 33);
274
            random_seed = hdw + (ldw >> 31);
275
        }
276
    }
277
}
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279

    
280
static void init_noise_samples(void) {
281
    int i;
282
    int random_seed = 0;
283
    float delta = 1.0 / 16384.0;
284
    for (i = 0; i < 128;i++) {
285
        random_seed = random_seed * 214013 + 2531011;
286
        noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0);
287
    }
288
}
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291
static void qdm2_init_vlc(void)
292
{
293
    init_vlc (&vlc_tab_level, 8, 24,
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        vlc_tab_level_huffbits, 1, 1,
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        vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
296

    
297
    init_vlc (&vlc_tab_diff, 8, 37,
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        vlc_tab_diff_huffbits, 1, 1,
299
        vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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301
    init_vlc (&vlc_tab_run, 5, 6,
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        vlc_tab_run_huffbits, 1, 1,
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        vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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305
    init_vlc (&fft_level_exp_alt_vlc, 8, 28,
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        fft_level_exp_alt_huffbits, 1, 1,
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        fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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309
    init_vlc (&fft_level_exp_vlc, 8, 20,
310
        fft_level_exp_huffbits, 1, 1,
311
        fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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313
    init_vlc (&fft_stereo_exp_vlc, 6, 7,
314
        fft_stereo_exp_huffbits, 1, 1,
315
        fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
316

    
317
    init_vlc (&fft_stereo_phase_vlc, 6, 9,
318
        fft_stereo_phase_huffbits, 1, 1,
319
        fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
320

    
321
    init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
322
        vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
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        vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
324

    
325
    init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
326
        vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
327
        vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
328

    
329
    init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
330
        vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
331
        vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
332

    
333
    init_vlc (&vlc_tab_type30, 6, 9,
334
        vlc_tab_type30_huffbits, 1, 1,
335
        vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
336

    
337
    init_vlc (&vlc_tab_type34, 5, 10,
338
        vlc_tab_type34_huffbits, 1, 1,
339
        vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
340

    
341
    init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
342
        vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
343
        vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
344

    
345
    init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
346
        vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
347
        vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
348

    
349
    init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
350
        vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
351
        vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
352

    
353
    init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
354
        vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
355
        vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
356

    
357
    init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
358
        vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
359
        vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
360
}
361

    
362

    
363
/* for floating point to fixed point conversion */
364
static float f2i_scale = (float) (1 << (FRAC_BITS - 15));
365

    
366

    
367
static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
368
{
369
    int value;
370

    
371
    value = get_vlc2(gb, vlc->table, vlc->bits, depth);
372

    
373
    /* stage-2, 3 bits exponent escape sequence */
374
    if (value-- == 0)
375
        value = get_bits (gb, get_bits (gb, 3) + 1);
376

    
377
    /* stage-3, optional */
378
    if (flag) {
379
        int tmp = vlc_stage3_values[value];
380

    
381
        if ((value & ~3) > 0)
382
            tmp += get_bits (gb, (value >> 2));
383
        value = tmp;
384
    }
385

    
386
    return value;
387
}
388

    
389

    
390
static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
391
{
392
    int value = qdm2_get_vlc (gb, vlc, 0, depth);
393

    
394
    return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
395
}
396

    
397

    
398
/**
399
 * QDM2 checksum
400
 *
401
 * @param data      pointer to data to be checksum'ed
402
 * @param length    data length
403
 * @param value     checksum value
404
 *
405
 * @return          0 if checksum is OK
406
 */
407
static uint16_t qdm2_packet_checksum (uint8_t *data, int length, int value) {
408
    int i;
409

    
410
    for (i=0; i < length; i++)
411
        value -= data[i];
412

    
413
    return (uint16_t)(value & 0xffff);
414
}
415

    
416

    
417
/**
418
 * Fills a QDM2SubPacket structure with packet type, size, and data pointer.
419
 *
420
 * @param gb            bitreader context
421
 * @param sub_packet    packet under analysis
422
 */
423
static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
424
{
425
    sub_packet->type = get_bits (gb, 8);
426

    
427
    if (sub_packet->type == 0) {
428
        sub_packet->size = 0;
429
        sub_packet->data = NULL;
430
    } else {
431
        sub_packet->size = get_bits (gb, 8);
432

    
433
      if (sub_packet->type & 0x80) {
434
          sub_packet->size <<= 8;
435
          sub_packet->size  |= get_bits (gb, 8);
436
          sub_packet->type  &= 0x7f;
437
      }
438

    
439
      if (sub_packet->type == 0x7f)
440
          sub_packet->type |= (get_bits (gb, 8) << 8);
441

    
442
      sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
443
    }
444

    
445
    av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
446
        sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
447
}
448

    
449

    
450
/**
451
 * Return node pointer to first packet of requested type in list.
452
 *
453
 * @param list    list of subpackets to be scanned
454
 * @param type    type of searched subpacket
455
 * @return        node pointer for subpacket if found, else NULL
456
 */
457
static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
458
{
459
    while (list != NULL && list->packet != NULL) {
460
        if (list->packet->type == type)
461
            return list;
462
        list = list->next;
463
    }
464
    return NULL;
465
}
466

    
467

    
468
/**
469
 * Replaces 8 elements with their average value.
470
 * Called by qdm2_decode_superblock before starting subblock decoding.
471
 *
472
 * @param q       context
473
 */
474
static void average_quantized_coeffs (QDM2Context *q)
475
{
476
    int i, j, n, ch, sum;
477

    
478
    n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
479

    
480
    for (ch = 0; ch < q->nb_channels; ch++)
481
        for (i = 0; i < n; i++) {
482
            sum = 0;
483

    
484
            for (j = 0; j < 8; j++)
485
                sum += q->quantized_coeffs[ch][i][j];
486

    
487
            sum /= 8;
488
            if (sum > 0)
489
                sum--;
490

    
491
            for (j=0; j < 8; j++)
492
                q->quantized_coeffs[ch][i][j] = sum;
493
        }
494
}
495

    
496

    
497
/**
498
 * Build subband samples with noise weighted by q->tone_level.
499
 * Called by synthfilt_build_sb_samples.
500
 *
501
 * @param q     context
502
 * @param sb    subband index
503
 */
504
static void build_sb_samples_from_noise (QDM2Context *q, int sb)
505
{
506
    int ch, j;
507

    
508
    FIX_NOISE_IDX(q->noise_idx);
509

    
510
    if (!q->nb_channels)
511
        return;
512

    
513
    for (ch = 0; ch < q->nb_channels; ch++)
514
        for (j = 0; j < 64; j++) {
515
            q->sb_samples[ch][j * 2][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
516
            q->sb_samples[ch][j * 2 + 1][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
517
        }
518
}
519

    
520

    
521
/**
522
 * Called while processing data from subpackets 11 and 12.
523
 * Used after making changes to coding_method array.
524
 *
525
 * @param sb               subband index
526
 * @param channels         number of channels
527
 * @param coding_method    q->coding_method[0][0][0]
528
 */
529
static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
530
{
531
    int j,k;
532
    int ch;
533
    int run, case_val;
534
    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};
535

    
536
    for (ch = 0; ch < channels; ch++) {
537
        for (j = 0; j < 64; ) {
538
            if((coding_method[ch][sb][j] - 8) > 22) {
539
                run = 1;
540
                case_val = 8;
541
            } else {
542
                switch (switchtable[coding_method[ch][sb][j]-8]) {
543
                    case 0: run = 10; case_val = 10; break;
544
                    case 1: run = 1; case_val = 16; break;
545
                    case 2: run = 5; case_val = 24; break;
546
                    case 3: run = 3; case_val = 30; break;
547
                    case 4: run = 1; case_val = 30; break;
548
                    case 5: run = 1; case_val = 8; break;
549
                    default: run = 1; case_val = 8; break;
550
                }
551
            }
552
            for (k = 0; k < run; k++)
553
                if (j + k < 128)
554
                    if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
555
                        if (k > 0) {
556
                           SAMPLES_NEEDED
557
                            //not debugged, almost never used
558
                            memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
559
                            memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
560
                        }
561
            j += run;
562
        }
563
    }
564
}
565

    
566

    
567
/**
568
 * Related to synthesis filter
569
 * Called by process_subpacket_10
570
 *
571
 * @param q       context
572
 * @param flag    1 if called after getting data from subpacket 10, 0 if no subpacket 10
573
 */
574
static void fill_tone_level_array (QDM2Context *q, int flag)
575
{
576
    int i, sb, ch, sb_used;
577
    int tmp, tab;
578

    
579
    // This should never happen
580
    if (q->nb_channels <= 0)
581
        return;
582

    
583
    for (ch = 0; ch < q->nb_channels; ch++)
584
        for (sb = 0; sb < 30; sb++)
585
            for (i = 0; i < 8; i++) {
586
                if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
587
                    tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
588
                          q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
589
                else
590
                    tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
591
                if(tmp < 0)
592
                    tmp += 0xff;
593
                q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
594
            }
595

    
596
    sb_used = QDM2_SB_USED(q->sub_sampling);
597

    
598
    if ((q->superblocktype_2_3 != 0) && !flag) {
599
        for (sb = 0; sb < sb_used; sb++)
600
            for (ch = 0; ch < q->nb_channels; ch++)
601
                for (i = 0; i < 64; i++) {
602
                    q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
603
                    if (q->tone_level_idx[ch][sb][i] < 0)
604
                        q->tone_level[ch][sb][i] = 0;
605
                    else
606
                        q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
607
                }
608
    } else {
609
        tab = q->superblocktype_2_3 ? 0 : 1;
610
        for (sb = 0; sb < sb_used; sb++) {
611
            if ((sb >= 4) && (sb <= 23)) {
612
                for (ch = 0; ch < q->nb_channels; ch++)
613
                    for (i = 0; i < 64; i++) {
614
                        tmp = q->tone_level_idx_base[ch][sb][i / 8] -
615
                              q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
616
                              q->tone_level_idx_mid[ch][sb - 4][i / 8] -
617
                              q->tone_level_idx_hi2[ch][sb - 4];
618
                        q->tone_level_idx[ch][sb][i] = tmp & 0xff;
619
                        if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
620
                            q->tone_level[ch][sb][i] = 0;
621
                        else
622
                            q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
623
                }
624
            } else {
625
                if (sb > 4) {
626
                    for (ch = 0; ch < q->nb_channels; ch++)
627
                        for (i = 0; i < 64; i++) {
628
                            tmp = q->tone_level_idx_base[ch][sb][i / 8] -
629
                                  q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
630
                                  q->tone_level_idx_hi2[ch][sb - 4];
631
                            q->tone_level_idx[ch][sb][i] = tmp & 0xff;
632
                            if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
633
                                q->tone_level[ch][sb][i] = 0;
634
                            else
635
                                q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
636
                    }
637
                } else {
638
                    for (ch = 0; ch < q->nb_channels; ch++)
639
                        for (i = 0; i < 64; i++) {
640
                            tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
641
                            if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
642
                                q->tone_level[ch][sb][i] = 0;
643
                            else
644
                                q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
645
                        }
646
                }
647
            }
648
        }
649
    }
650

    
651
    return;
652
}
653

    
654

    
655
/**
656
 * Related to synthesis filter
657
 * Called by process_subpacket_11
658
 * c is built with data from subpacket 11
659
 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
660
 *
661
 * @param tone_level_idx
662
 * @param tone_level_idx_temp
663
 * @param coding_method        q->coding_method[0][0][0]
664
 * @param nb_channels          number of channels
665
 * @param c                    coming from subpacket 11, passed as 8*c
666
 * @param superblocktype_2_3   flag based on superblock packet type
667
 * @param cm_table_select      q->cm_table_select
668
 */
669
static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
670
                sb_int8_array coding_method, int nb_channels,
671
                int c, int superblocktype_2_3, int cm_table_select)
672
{
673
    int ch, sb, j;
674
    int tmp, acc, esp_40, comp;
675
    int add1, add2, add3, add4;
676
    int64_t multres;
677

    
678
    // This should never happen
679
    if (nb_channels <= 0)
680
        return;
681

    
682
    if (!superblocktype_2_3) {
683
        /* This case is untested, no samples available */
684
        SAMPLES_NEEDED
685
        for (ch = 0; ch < nb_channels; ch++)
686
            for (sb = 0; sb < 30; sb++) {
687
                for (j = 1; j < 64; j++) {
688
                    add1 = tone_level_idx[ch][sb][j] - 10;
689
                    if (add1 < 0)
690
                        add1 = 0;
691
                    add2 = add3 = add4 = 0;
692
                    if (sb > 1) {
693
                        add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
694
                        if (add2 < 0)
695
                            add2 = 0;
696
                    }
697
                    if (sb > 0) {
698
                        add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
699
                        if (add3 < 0)
700
                            add3 = 0;
701
                    }
702
                    if (sb < 29) {
703
                        add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
704
                        if (add4 < 0)
705
                            add4 = 0;
706
                    }
707
                    tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
708
                    if (tmp < 0)
709
                        tmp = 0;
710
                    tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
711
                }
712
                tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
713
            }
714
            acc = 0;
715
            for (ch = 0; ch < nb_channels; ch++)
716
                for (sb = 0; sb < 30; sb++)
717
                    for (j = 0; j < 64; j++)
718
                        acc += tone_level_idx_temp[ch][sb][j];
719
            if (acc)
720
                tmp = c * 256 / (acc & 0xffff);
721
            multres = 0x66666667 * (acc * 10);
722
            esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
723
            for (ch = 0;  ch < nb_channels; ch++)
724
                for (sb = 0; sb < 30; sb++)
725
                    for (j = 0; j < 64; j++) {
726
                        comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
727
                        if (comp < 0)
728
                            comp += 0xff;
729
                        comp /= 256; // signed shift
730
                        switch(sb) {
731
                            case 0:
732
                                if (comp < 30)
733
                                    comp = 30;
734
                                comp += 15;
735
                                break;
736
                            case 1:
737
                                if (comp < 24)
738
                                    comp = 24;
739
                                comp += 10;
740
                                break;
741
                            case 2:
742
                            case 3:
743
                            case 4:
744
                                if (comp < 16)
745
                                    comp = 16;
746
                        }
747
                        if (comp <= 5)
748
                            tmp = 0;
749
                        else if (comp <= 10)
750
                            tmp = 10;
751
                        else if (comp <= 16)
752
                            tmp = 16;
753
                        else if (comp <= 24)
754
                            tmp = -1;
755
                        else
756
                            tmp = 0;
757
                        coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
758
                    }
759
            for (sb = 0; sb < 30; sb++)
760
                fix_coding_method_array(sb, nb_channels, coding_method);
761
            for (ch = 0; ch < nb_channels; ch++)
762
                for (sb = 0; sb < 30; sb++)
763
                    for (j = 0; j < 64; j++)
764
                        if (sb >= 10) {
765
                            if (coding_method[ch][sb][j] < 10)
766
                                coding_method[ch][sb][j] = 10;
767
                        } else {
768
                            if (sb >= 2) {
769
                                if (coding_method[ch][sb][j] < 16)
770
                                    coding_method[ch][sb][j] = 16;
771
                            } else {
772
                                if (coding_method[ch][sb][j] < 30)
773
                                    coding_method[ch][sb][j] = 30;
774
                            }
775
                        }
776
    } else { // superblocktype_2_3 != 0
777
        for (ch = 0; ch < nb_channels; ch++)
778
            for (sb = 0; sb < 30; sb++)
779
                for (j = 0; j < 64; j++)
780
                    coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
781
    }
782

    
783
    return;
784
}
785

    
786

    
787
/**
788
 *
789
 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
790
 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
791
 *
792
 * @param q         context
793
 * @param gb        bitreader context
794
 * @param length    packet length in bits
795
 * @param sb_min    lower subband processed (sb_min included)
796
 * @param sb_max    higher subband processed (sb_max excluded)
797
 */
798
static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
799
{
800
    int sb, j, k, n, ch, run, channels;
801
    int joined_stereo, zero_encoding, chs;
802
    int type34_first;
803
    float type34_div = 0;
804
    float type34_predictor;
805
    float samples[10], sign_bits[16];
806

    
807
    if (length == 0) {
808
        // If no data use noise
809
        for (sb=sb_min; sb < sb_max; sb++)
810
            build_sb_samples_from_noise (q, sb);
811

    
812
        return;
813
    }
814

    
815
    for (sb = sb_min; sb < sb_max; sb++) {
816
        FIX_NOISE_IDX(q->noise_idx);
817

    
818
        channels = q->nb_channels;
819

    
820
        if (q->nb_channels <= 1 || sb < 12)
821
            joined_stereo = 0;
822
        else if (sb >= 24)
823
            joined_stereo = 1;
824
        else
825
            joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
826

    
827
        if (joined_stereo) {
828
            if (BITS_LEFT(length,gb) >= 16)
829
                for (j = 0; j < 16; j++)
830
                    sign_bits[j] = get_bits1 (gb);
831

    
832
            for (j = 0; j < 64; j++)
833
                if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
834
                    q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
835

    
836
            fix_coding_method_array(sb, q->nb_channels, q->coding_method);
837
            channels = 1;
838
        }
839

    
840
        for (ch = 0; ch < channels; ch++) {
841
            zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
842
            type34_predictor = 0.0;
843
            type34_first = 1;
844

    
845
            for (j = 0; j < 128; ) {
846
                switch (q->coding_method[ch][sb][j / 2]) {
847
                    case 8:
848
                        if (BITS_LEFT(length,gb) >= 10) {
849
                            if (zero_encoding) {
850
                                for (k = 0; k < 5; k++) {
851
                                    if ((j + 2 * k) >= 128)
852
                                        break;
853
                                    samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
854
                                }
855
                            } else {
856
                                n = get_bits(gb, 8);
857
                                for (k = 0; k < 5; k++)
858
                                    samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
859
                            }
860
                            for (k = 0; k < 5; k++)
861
                                samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
862
                        } else {
863
                            for (k = 0; k < 10; k++)
864
                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
865
                        }
866
                        run = 10;
867
                        break;
868

    
869
                    case 10:
870
                        if (BITS_LEFT(length,gb) >= 1) {
871
                            float f = 0.81;
872

    
873
                            if (get_bits1(gb))
874
                                f = -f;
875
                            f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
876
                            samples[0] = f;
877
                        } else {
878
                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
879
                        }
880
                        run = 1;
881
                        break;
882

    
883
                    case 16:
884
                        if (BITS_LEFT(length,gb) >= 10) {
885
                            if (zero_encoding) {
886
                                for (k = 0; k < 5; k++) {
887
                                    if ((j + k) >= 128)
888
                                        break;
889
                                    samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
890
                                }
891
                            } else {
892
                                n = get_bits (gb, 8);
893
                                for (k = 0; k < 5; k++)
894
                                    samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
895
                            }
896
                        } else {
897
                            for (k = 0; k < 5; k++)
898
                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
899
                        }
900
                        run = 5;
901
                        break;
902

    
903
                    case 24:
904
                        if (BITS_LEFT(length,gb) >= 7) {
905
                            n = get_bits(gb, 7);
906
                            for (k = 0; k < 3; k++)
907
                                samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
908
                        } else {
909
                            for (k = 0; k < 3; k++)
910
                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
911
                        }
912
                        run = 3;
913
                        break;
914

    
915
                    case 30:
916
                        if (BITS_LEFT(length,gb) >= 4)
917
                            samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
918
                        else
919
                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
920

    
921
                        run = 1;
922
                        break;
923

    
924
                    case 34:
925
                        if (BITS_LEFT(length,gb) >= 7) {
926
                            if (type34_first) {
927
                                type34_div = (float)(1 << get_bits(gb, 2));
928
                                samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
929
                                type34_predictor = samples[0];
930
                                type34_first = 0;
931
                            } else {
932
                                samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
933
                                type34_predictor = samples[0];
934
                            }
935
                        } else {
936
                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
937
                        }
938
                        run = 1;
939
                        break;
940

    
941
                    default:
942
                        samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
943
                        run = 1;
944
                        break;
945
                }
946

    
947
                if (joined_stereo) {
948
                    float tmp[10][MPA_MAX_CHANNELS];
949

    
950
                    for (k = 0; k < run; k++) {
951
                        tmp[k][0] = samples[k];
952
                        tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
953
                    }
954
                    for (chs = 0; chs < q->nb_channels; chs++)
955
                        for (k = 0; k < run; k++)
956
                            if ((j + k) < 128)
957
                                q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
958
                } else {
959
                    for (k = 0; k < run; k++)
960
                        if ((j + k) < 128)
961
                            q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
962
                }
963

    
964
                j += run;
965
            } // j loop
966
        } // channel loop
967
    } // subband loop
968
}
969

    
970

    
971
/**
972
 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
973
 * This is similar to process_subpacket_9, but for a single channel and for element [0]
974
 * same VLC tables as process_subpacket_9 are used.
975
 *
976
 * @param q         context
977
 * @param quantized_coeffs    pointer to quantized_coeffs[ch][0]
978
 * @param gb        bitreader context
979
 * @param length    packet length in bits
980
 */
981
static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
982
{
983
    int i, k, run, level, diff;
984

    
985
    if (BITS_LEFT(length,gb) < 16)
986
        return;
987
    level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
988

    
989
    quantized_coeffs[0] = level;
990

    
991
    for (i = 0; i < 7; ) {
992
        if (BITS_LEFT(length,gb) < 16)
993
            break;
994
        run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
995

    
996
        if (BITS_LEFT(length,gb) < 16)
997
            break;
998
        diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
999

    
1000
        for (k = 1; k <= run; k++)
1001
            quantized_coeffs[i + k] = (level + ((k * diff) / run));
1002

    
1003
        level += diff;
1004
        i += run;
1005
    }
1006
}
1007

    
1008

    
1009
/**
1010
 * Related to synthesis filter, process data from packet 10
1011
 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
1012
 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
1013
 *
1014
 * @param q         context
1015
 * @param gb        bitreader context
1016
 * @param length    packet length in bits
1017
 */
1018
static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
1019
{
1020
    int sb, j, k, n, ch;
1021

    
1022
    for (ch = 0; ch < q->nb_channels; ch++) {
1023
        init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
1024

    
1025
        if (BITS_LEFT(length,gb) < 16) {
1026
            memset(q->quantized_coeffs[ch][0], 0, 8);
1027
            break;
1028
        }
1029
    }
1030

    
1031
    n = q->sub_sampling + 1;
1032

    
1033
    for (sb = 0; sb < n; sb++)
1034
        for (ch = 0; ch < q->nb_channels; ch++)
1035
            for (j = 0; j < 8; j++) {
1036
                if (BITS_LEFT(length,gb) < 1)
1037
                    break;
1038
                if (get_bits1(gb)) {
1039
                    for (k=0; k < 8; k++) {
1040
                        if (BITS_LEFT(length,gb) < 16)
1041
                            break;
1042
                        q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1043
                    }
1044
                } else {
1045
                    for (k=0; k < 8; k++)
1046
                        q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1047
                }
1048
            }
1049

    
1050
    n = QDM2_SB_USED(q->sub_sampling) - 4;
1051

    
1052
    for (sb = 0; sb < n; sb++)
1053
        for (ch = 0; ch < q->nb_channels; ch++) {
1054
            if (BITS_LEFT(length,gb) < 16)
1055
                break;
1056
            q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1057
            if (sb > 19)
1058
                q->tone_level_idx_hi2[ch][sb] -= 16;
1059
            else
1060
                for (j = 0; j < 8; j++)
1061
                    q->tone_level_idx_mid[ch][sb][j] = -16;
1062
        }
1063

    
1064
    n = QDM2_SB_USED(q->sub_sampling) - 5;
1065

    
1066
    for (sb = 0; sb < n; sb++)
1067
        for (ch = 0; ch < q->nb_channels; ch++)
1068
            for (j = 0; j < 8; j++) {
1069
                if (BITS_LEFT(length,gb) < 16)
1070
                    break;
1071
                q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1072
            }
1073
}
1074

    
1075
/**
1076
 * Process subpacket 9, init quantized_coeffs with data from it
1077
 *
1078
 * @param q       context
1079
 * @param node    pointer to node with packet
1080
 */
1081
static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
1082
{
1083
    GetBitContext gb;
1084
    int i, j, k, n, ch, run, level, diff;
1085

    
1086
    init_get_bits(&gb, node->packet->data, node->packet->size*8);
1087

    
1088
    n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1089

    
1090
    for (i = 1; i < n; i++)
1091
        for (ch=0; ch < q->nb_channels; ch++) {
1092
            level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1093
            q->quantized_coeffs[ch][i][0] = level;
1094

    
1095
            for (j = 0; j < (8 - 1); ) {
1096
                run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1097
                diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1098

    
1099
                for (k = 1; k <= run; k++)
1100
                    q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1101

    
1102
                level += diff;
1103
                j += run;
1104
            }
1105
        }
1106

    
1107
    for (ch = 0; ch < q->nb_channels; ch++)
1108
        for (i = 0; i < 8; i++)
1109
            q->quantized_coeffs[ch][0][i] = 0;
1110
}
1111

    
1112

    
1113
/**
1114
 * Process subpacket 10 if not null, else
1115
 *
1116
 * @param q         context
1117
 * @param node      pointer to node with packet
1118
 * @param length    packet length in bits
1119
 */
1120
static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
1121
{
1122
    GetBitContext gb;
1123

    
1124
    init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1125

    
1126
    if (length != 0) {
1127
        init_tone_level_dequantization(q, &gb, length);
1128
        fill_tone_level_array(q, 1);
1129
    } else {
1130
        fill_tone_level_array(q, 0);
1131
    }
1132
}
1133

    
1134

    
1135
/**
1136
 * Process subpacket 11
1137
 *
1138
 * @param q         context
1139
 * @param node      pointer to node with packet
1140
 * @param length    packet length in bit
1141
 */
1142
static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
1143
{
1144
    GetBitContext gb;
1145

    
1146
    init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1147
    if (length >= 32) {
1148
        int c = get_bits (&gb, 13);
1149

    
1150
        if (c > 3)
1151
            fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
1152
                                      q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
1153
    }
1154

    
1155
    synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1156
}
1157

    
1158

    
1159
/**
1160
 * Process subpacket 12
1161
 *
1162
 * @param q         context
1163
 * @param node      pointer to node with packet
1164
 * @param length    packet length in bits
1165
 */
1166
static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
1167
{
1168
    GetBitContext gb;
1169

    
1170
    init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1171
    synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1172
}
1173

    
1174
/*
1175
 * Process new subpackets for synthesis filter
1176
 *
1177
 * @param q       context
1178
 * @param list    list with synthesis filter packets (list D)
1179
 */
1180
static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
1181
{
1182
    QDM2SubPNode *nodes[4];
1183

    
1184
    nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1185
    if (nodes[0] != NULL)
1186
        process_subpacket_9(q, nodes[0]);
1187

    
1188
    nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1189
    if (nodes[1] != NULL)
1190
        process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
1191
    else
1192
        process_subpacket_10(q, NULL, 0);
1193

    
1194
    nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1195
    if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1196
        process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
1197
    else
1198
        process_subpacket_11(q, NULL, 0);
1199

    
1200
    nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1201
    if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1202
        process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
1203
    else
1204
        process_subpacket_12(q, NULL, 0);
1205
}
1206

    
1207

    
1208
/*
1209
 * Decode superblock, fill packet lists.
1210
 *
1211
 * @param q    context
1212
 */
1213
static void qdm2_decode_super_block (QDM2Context *q)
1214
{
1215
    GetBitContext gb;
1216
    QDM2SubPacket header, *packet;
1217
    int i, packet_bytes, sub_packet_size, sub_packets_D;
1218
    unsigned int next_index = 0;
1219

    
1220
    memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1221
    memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1222
    memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1223

    
1224
    q->sub_packets_B = 0;
1225
    sub_packets_D = 0;
1226

    
1227
    average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1228

    
1229
    init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
1230
    qdm2_decode_sub_packet_header(&gb, &header);
1231

    
1232
    if (header.type < 2 || header.type >= 8) {
1233
        q->has_errors = 1;
1234
        av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1235
        return;
1236
    }
1237

    
1238
    q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1239
    packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1240

    
1241
    init_get_bits(&gb, header.data, header.size*8);
1242

    
1243
    if (header.type == 2 || header.type == 4 || header.type == 5) {
1244
        int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8);
1245

    
1246
        csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1247

    
1248
        if (csum != 0) {
1249
            q->has_errors = 1;
1250
            av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1251
            return;
1252
        }
1253
    }
1254

    
1255
    q->sub_packet_list_B[0].packet = NULL;
1256
    q->sub_packet_list_D[0].packet = NULL;
1257

    
1258
    for (i = 0; i < 6; i++)
1259
        if (--q->fft_level_exp[i] < 0)
1260
            q->fft_level_exp[i] = 0;
1261

    
1262
    for (i = 0; packet_bytes > 0; i++) {
1263
        int j;
1264

    
1265
        q->sub_packet_list_A[i].next = NULL;
1266

    
1267
        if (i > 0) {
1268
            q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1269

    
1270
            /* seek to next block */
1271
            init_get_bits(&gb, header.data, header.size*8);
1272
            skip_bits(&gb, next_index*8);
1273

    
1274
            if (next_index >= header.size)
1275
                break;
1276
        }
1277

    
1278
        /* decode subpacket */
1279
        packet = &q->sub_packets[i];
1280
        qdm2_decode_sub_packet_header(&gb, packet);
1281
        next_index = packet->size + get_bits_count(&gb) / 8;
1282
        sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1283

    
1284
        if (packet->type == 0)
1285
            break;
1286

    
1287
        if (sub_packet_size > packet_bytes) {
1288
            if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1289
                break;
1290
            packet->size += packet_bytes - sub_packet_size;
1291
        }
1292

    
1293
        packet_bytes -= sub_packet_size;
1294

    
1295
        /* add subpacket to 'all subpackets' list */
1296
        q->sub_packet_list_A[i].packet = packet;
1297

    
1298
        /* add subpacket to related list */
1299
        if (packet->type == 8) {
1300
            SAMPLES_NEEDED_2("packet type 8");
1301
            return;
1302
        } else if (packet->type >= 9 && packet->type <= 12) {
1303
            /* packets for MPEG Audio like Synthesis Filter */
1304
            QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1305
        } else if (packet->type == 13) {
1306
            for (j = 0; j < 6; j++)
1307
                q->fft_level_exp[j] = get_bits(&gb, 6);
1308
        } else if (packet->type == 14) {
1309
            for (j = 0; j < 6; j++)
1310
                q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1311
        } else if (packet->type == 15) {
1312
            SAMPLES_NEEDED_2("packet type 15")
1313
            return;
1314
        } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1315
            /* packets for FFT */
1316
            QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1317
        }
1318
    } // Packet bytes loop
1319

    
1320
/* **************************************************************** */
1321
    if (q->sub_packet_list_D[0].packet != NULL) {
1322
        process_synthesis_subpackets(q, q->sub_packet_list_D);
1323
        q->do_synth_filter = 1;
1324
    } else if (q->do_synth_filter) {
1325
        process_subpacket_10(q, NULL, 0);
1326
        process_subpacket_11(q, NULL, 0);
1327
        process_subpacket_12(q, NULL, 0);
1328
    }
1329
/* **************************************************************** */
1330
}
1331

    
1332

    
1333
static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1334
                       int offset, int duration, int channel,
1335
                       int exp, int phase)
1336
{
1337
    if (q->fft_coefs_min_index[duration] < 0)
1338
        q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1339

    
1340
    q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1341
    q->fft_coefs[q->fft_coefs_index].channel = channel;
1342
    q->fft_coefs[q->fft_coefs_index].offset = offset;
1343
    q->fft_coefs[q->fft_coefs_index].exp = exp;
1344
    q->fft_coefs[q->fft_coefs_index].phase = phase;
1345
    q->fft_coefs_index++;
1346
}
1347

    
1348

    
1349
static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
1350
{
1351
    int channel, stereo, phase, exp;
1352
    int local_int_4,  local_int_8,  stereo_phase,  local_int_10;
1353
    int local_int_14, stereo_exp, local_int_20, local_int_28;
1354
    int n, offset;
1355

    
1356
    local_int_4 = 0;
1357
    local_int_28 = 0;
1358
    local_int_20 = 2;
1359
    local_int_8 = (4 - duration);
1360
    local_int_10 = 1 << (q->group_order - duration - 1);
1361
    offset = 1;
1362

    
1363
    while (1) {
1364
        if (q->superblocktype_2_3) {
1365
            while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1366
                offset = 1;
1367
                if (n == 0) {
1368
                    local_int_4 += local_int_10;
1369
                    local_int_28 += (1 << local_int_8);
1370
                } else {
1371
                    local_int_4 += 8*local_int_10;
1372
                    local_int_28 += (8 << local_int_8);
1373
                }
1374
            }
1375
            offset += (n - 2);
1376
        } else {
1377
            offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1378
            while (offset >= (local_int_10 - 1)) {
1379
                offset += (1 - (local_int_10 - 1));
1380
                local_int_4  += local_int_10;
1381
                local_int_28 += (1 << local_int_8);
1382
            }
1383
        }
1384

    
1385
        if (local_int_4 >= q->group_size)
1386
            return;
1387

    
1388
        local_int_14 = (offset >> local_int_8);
1389

    
1390
        if (q->nb_channels > 1) {
1391
            channel = get_bits1(gb);
1392
            stereo = get_bits1(gb);
1393
        } else {
1394
            channel = 0;
1395
            stereo = 0;
1396
        }
1397

    
1398
        exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1399
        exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1400
        exp = (exp < 0) ? 0 : exp;
1401

    
1402
        phase = get_bits(gb, 3);
1403
        stereo_exp = 0;
1404
        stereo_phase = 0;
1405

    
1406
        if (stereo) {
1407
            stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1408
            stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1409
            if (stereo_phase < 0)
1410
                stereo_phase += 8;
1411
        }
1412

    
1413
        if (q->frequency_range > (local_int_14 + 1)) {
1414
            int sub_packet = (local_int_20 + local_int_28);
1415

    
1416
            qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1417
            if (stereo)
1418
                qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1419
        }
1420

    
1421
        offset++;
1422
    }
1423
}
1424

    
1425

    
1426
static void qdm2_decode_fft_packets (QDM2Context *q)
1427
{
1428
    int i, j, min, max, value, type, unknown_flag;
1429
    GetBitContext gb;
1430

    
1431
    if (q->sub_packet_list_B[0].packet == NULL)
1432
        return;
1433

    
1434
    /* reset minimum indices for FFT coefficients */
1435
    q->fft_coefs_index = 0;
1436
    for (i=0; i < 5; i++)
1437
        q->fft_coefs_min_index[i] = -1;
1438

    
1439
    /* process subpackets ordered by type, largest type first */
1440
    for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1441
        QDM2SubPacket *packet;
1442

    
1443
        /* find subpacket with largest type less than max */
1444
        for (j = 0, min = 0, packet = NULL; j < q->sub_packets_B; j++) {
1445
            value = q->sub_packet_list_B[j].packet->type;
1446
            if (value > min && value < max) {
1447
                min = value;
1448
                packet = q->sub_packet_list_B[j].packet;
1449
            }
1450
        }
1451

    
1452
        max = min;
1453

    
1454
        /* check for errors (?) */
1455
        if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1456
            return;
1457

    
1458
        /* decode FFT tones */
1459
        init_get_bits (&gb, packet->data, packet->size*8);
1460

    
1461
        if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1462
            unknown_flag = 1;
1463
        else
1464
            unknown_flag = 0;
1465

    
1466
        type = packet->type;
1467

    
1468
        if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1469
            int duration = q->sub_sampling + 5 - (type & 15);
1470

    
1471
            if (duration >= 0 && duration < 4)
1472
                qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1473
        } else if (type == 31) {
1474
            for (j=0; j < 4; j++)
1475
                qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1476
        } else if (type == 46) {
1477
            for (j=0; j < 6; j++)
1478
                q->fft_level_exp[j] = get_bits(&gb, 6);
1479
            for (j=0; j < 4; j++)
1480
            qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1481
        }
1482
    } // Loop on B packets
1483

    
1484
    /* calculate maximum indices for FFT coefficients */
1485
    for (i = 0, j = -1; i < 5; i++)
1486
        if (q->fft_coefs_min_index[i] >= 0) {
1487
            if (j >= 0)
1488
                q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1489
            j = i;
1490
        }
1491
    if (j >= 0)
1492
        q->fft_coefs_max_index[j] = q->fft_coefs_index;
1493
}
1494

    
1495

    
1496
static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
1497
{
1498
   float level, f[6];
1499
   int i;
1500
   QDM2Complex c;
1501
   const double iscale = 2.0*M_PI / 512.0;
1502

    
1503
    tone->phase += tone->phase_shift;
1504

    
1505
    /* calculate current level (maximum amplitude) of tone */
1506
    level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1507
    c.im = level * sin(tone->phase*iscale);
1508
    c.re = level * cos(tone->phase*iscale);
1509

    
1510
    /* generate FFT coefficients for tone */
1511
    if (tone->duration >= 3 || tone->cutoff >= 3) {
1512
        tone->samples_im[0] += c.im;
1513
        tone->samples_re[0] += c.re;
1514
        tone->samples_im[1] -= c.im;
1515
        tone->samples_re[1] -= c.re;
1516
    } else {
1517
        f[1] = -tone->table[4];
1518
        f[0] =  tone->table[3] - tone->table[0];
1519
        f[2] =  1.0 - tone->table[2] - tone->table[3];
1520
        f[3] =  tone->table[1] + tone->table[4] - 1.0;
1521
        f[4] =  tone->table[0] - tone->table[1];
1522
        f[5] =  tone->table[2];
1523
        for (i = 0; i < 2; i++) {
1524
            tone->samples_re[fft_cutoff_index_table[tone->cutoff][i]] += c.re * f[i];
1525
            tone->samples_im[fft_cutoff_index_table[tone->cutoff][i]] += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1526
        }
1527
        for (i = 0; i < 4; i++) {
1528
            tone->samples_re[i] += c.re * f[i+2];
1529
            tone->samples_im[i] += c.im * f[i+2];
1530
        }
1531
    }
1532

    
1533
    /* copy the tone if it has not yet died out */
1534
    if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1535
      memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1536
      q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1537
    }
1538
}
1539

    
1540

    
1541
static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1542
{
1543
    int i, j, ch;
1544
    const double iscale = 0.25 * M_PI;
1545

    
1546
    for (ch = 0; ch < q->channels; ch++) {
1547
        memset(q->fft.samples_im[ch], 0, q->fft_size * sizeof(float));
1548
        memset(q->fft.samples_re[ch], 0, q->fft_size * sizeof(float));
1549
    }
1550

    
1551

    
1552
    /* apply FFT tones with duration 4 (1 FFT period) */
1553
    if (q->fft_coefs_min_index[4] >= 0)
1554
        for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1555
            float level;
1556
            QDM2Complex c;
1557

    
1558
            if (q->fft_coefs[i].sub_packet != sub_packet)
1559
                break;
1560

    
1561
            ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1562
            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];
1563

    
1564
            c.re = level * cos(q->fft_coefs[i].phase * iscale);
1565
            c.im = level * sin(q->fft_coefs[i].phase * iscale);
1566
            q->fft.samples_re[ch][q->fft_coefs[i].offset + 0] += c.re;
1567
            q->fft.samples_im[ch][q->fft_coefs[i].offset + 0] += c.im;
1568
            q->fft.samples_re[ch][q->fft_coefs[i].offset + 1] -= c.re;
1569
            q->fft.samples_im[ch][q->fft_coefs[i].offset + 1] -= c.im;
1570
        }
1571

    
1572
    /* generate existing FFT tones */
1573
    for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1574
        qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1575
        q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1576
    }
1577

    
1578
    /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1579
    for (i = 0; i < 4; i++)
1580
        if (q->fft_coefs_min_index[i] >= 0) {
1581
            for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1582
                int offset, four_i;
1583
                FFTTone tone;
1584

    
1585
                if (q->fft_coefs[j].sub_packet != sub_packet)
1586
                    break;
1587

    
1588
                four_i = (4 - i);
1589
                offset = q->fft_coefs[j].offset >> four_i;
1590
                ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1591

    
1592
                if (offset < q->frequency_range) {
1593
                    if (offset < 2)
1594
                        tone.cutoff = offset;
1595
                    else
1596
                        tone.cutoff = (offset >= 60) ? 3 : 2;
1597

    
1598
                    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];
1599
                    tone.samples_im = &q->fft.samples_im[ch][offset];
1600
                    tone.samples_re = &q->fft.samples_re[ch][offset];
1601
                    tone.table = (float*)fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1602
                    tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1603
                    tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1604
                    tone.duration = i;
1605
                    tone.time_index = 0;
1606

    
1607
                    qdm2_fft_generate_tone(q, &tone);
1608
                }
1609
            }
1610
            q->fft_coefs_min_index[i] = j;
1611
        }
1612
}
1613

    
1614

    
1615
static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1616
{
1617
    const int n = 1 << (q->fft_order - 1);
1618
    const int n2 = n >> 1;
1619
    const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.25f : 0.50f;
1620
    float c, s, f0, f1, f2, f3;
1621
    int i, j;
1622

    
1623
    /* prerotation (or something like that) */
1624
    for (i=1; i < n2; i++) {
1625
        j  = (n - i);
1626
        c = q->exptab[i].re;
1627
        s = -q->exptab[i].im;
1628
        f0 = (q->fft.samples_re[channel][i] - q->fft.samples_re[channel][j]) * gain;
1629
        f1 = (q->fft.samples_im[channel][i] + q->fft.samples_im[channel][j]) * gain;
1630
        f2 = (q->fft.samples_re[channel][i] + q->fft.samples_re[channel][j]) * gain;
1631
        f3 = (q->fft.samples_im[channel][i] - q->fft.samples_im[channel][j]) * gain;
1632
        q->fft.complex[i].re =  s * f0 - c * f1 + f2;
1633
        q->fft.complex[i].im =  c * f0 + s * f1 + f3;
1634
        q->fft.complex[j].re = -s * f0 + c * f1 + f2;
1635
        q->fft.complex[j].im =  c * f0 + s * f1 - f3;
1636
    }
1637

    
1638
    q->fft.complex[ 0].re =  q->fft.samples_re[channel][ 0] * gain * 2.0;
1639
    q->fft.complex[ 0].im =  q->fft.samples_re[channel][ 0] * gain * 2.0;
1640
    q->fft.complex[n2].re =  q->fft.samples_re[channel][n2] * gain * 2.0;
1641
    q->fft.complex[n2].im = -q->fft.samples_im[channel][n2] * gain * 2.0;
1642

    
1643
    ff_fft_permute(&q->fft_ctx, (FFTComplex *) q->fft.complex);
1644
    ff_fft_calc (&q->fft_ctx, (FFTComplex *) q->fft.complex);
1645
    /* add samples to output buffer */
1646
    for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
1647
        q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex)[i];
1648
}
1649

    
1650

    
1651
/**
1652
 * @param q        context
1653
 * @param index    subpacket number
1654
 */
1655
static void qdm2_synthesis_filter (QDM2Context *q, int index)
1656
{
1657
    OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
1658
    int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1659

    
1660
    /* copy sb_samples */
1661
    sb_used = QDM2_SB_USED(q->sub_sampling);
1662

    
1663
    for (ch = 0; ch < q->channels; ch++)
1664
        for (i = 0; i < 8; i++)
1665
            for (k=sb_used; k < SBLIMIT; k++)
1666
                q->sb_samples[ch][(8 * index) + i][k] = 0;
1667

    
1668
    for (ch = 0; ch < q->nb_channels; ch++) {
1669
        OUT_INT *samples_ptr = samples + ch;
1670

    
1671
        for (i = 0; i < 8; i++) {
1672
            ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1673
                mpa_window, &dither_state,
1674
                samples_ptr, q->nb_channels,
1675
                q->sb_samples[ch][(8 * index) + i]);
1676
            samples_ptr += 32 * q->nb_channels;
1677
        }
1678
    }
1679

    
1680
    /* add samples to output buffer */
1681
    sub_sampling = (4 >> q->sub_sampling);
1682

    
1683
    for (ch = 0; ch < q->channels; ch++)
1684
        for (i = 0; i < q->frame_size; i++)
1685
            q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
1686
}
1687

    
1688

    
1689
/**
1690
 * Init static data (does not depend on specific file)
1691
 *
1692
 * @param q    context
1693
 */
1694
static void qdm2_init(QDM2Context *q) {
1695
    static int inited = 0;
1696

    
1697
    if (inited != 0)
1698
        return;
1699
    inited = 1;
1700

    
1701
    qdm2_init_vlc();
1702
    ff_mpa_synth_init(mpa_window);
1703
    softclip_table_init();
1704
    rnd_table_init();
1705
    init_noise_samples();
1706

    
1707
    av_log(NULL, AV_LOG_DEBUG, "init done\n");
1708
}
1709

    
1710

    
1711
#if 0
1712
static void dump_context(QDM2Context *q)
1713
{
1714
    int i;
1715
#define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
1716
    PRINT("compressed_data",q->compressed_data);
1717
    PRINT("compressed_size",q->compressed_size);
1718
    PRINT("frame_size",q->frame_size);
1719
    PRINT("checksum_size",q->checksum_size);
1720
    PRINT("channels",q->channels);
1721
    PRINT("nb_channels",q->nb_channels);
1722
    PRINT("fft_frame_size",q->fft_frame_size);
1723
    PRINT("fft_size",q->fft_size);
1724
    PRINT("sub_sampling",q->sub_sampling);
1725
    PRINT("fft_order",q->fft_order);
1726
    PRINT("group_order",q->group_order);
1727
    PRINT("group_size",q->group_size);
1728
    PRINT("sub_packet",q->sub_packet);
1729
    PRINT("frequency_range",q->frequency_range);
1730
    PRINT("has_errors",q->has_errors);
1731
    PRINT("fft_tone_end",q->fft_tone_end);
1732
    PRINT("fft_tone_start",q->fft_tone_start);
1733
    PRINT("fft_coefs_index",q->fft_coefs_index);
1734
    PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
1735
    PRINT("cm_table_select",q->cm_table_select);
1736
    PRINT("noise_idx",q->noise_idx);
1737

1738
    for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
1739
    {
1740
    FFTTone *t = &q->fft_tones[i];
1741

1742
    av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
1743
    av_log(NULL,AV_LOG_DEBUG,"  level = %f\n", t->level);
1744
//  PRINT(" level", t->level);
1745
    PRINT(" phase", t->phase);
1746
    PRINT(" phase_shift", t->phase_shift);
1747
    PRINT(" duration", t->duration);
1748
    PRINT(" samples_im", t->samples_im);
1749
    PRINT(" samples_re", t->samples_re);
1750
    PRINT(" table", t->table);
1751
    }
1752

1753
}
1754
#endif
1755

    
1756

    
1757
/**
1758
 * Init parameters from codec extradata
1759
 */
1760
static int qdm2_decode_init(AVCodecContext *avctx)
1761
{
1762
    QDM2Context *s = avctx->priv_data;
1763
    uint8_t *extradata;
1764
    int extradata_size;
1765
    int tmp_val, tmp, size;
1766
    int i;
1767
    float alpha;
1768

    
1769
    /* extradata parsing
1770

1771
    Structure:
1772
    wave {
1773
        frma (QDM2)
1774
        QDCA
1775
        QDCP
1776
    }
1777

1778
    32  size (including this field)
1779
    32  tag (=frma)
1780
    32  type (=QDM2 or QDMC)
1781

1782
    32  size (including this field, in bytes)
1783
    32  tag (=QDCA) // maybe mandatory parameters
1784
    32  unknown (=1)
1785
    32  channels (=2)
1786
    32  samplerate (=44100)
1787
    32  bitrate (=96000)
1788
    32  block size (=4096)
1789
    32  frame size (=256) (for one channel)
1790
    32  packet size (=1300)
1791

1792
    32  size (including this field, in bytes)
1793
    32  tag (=QDCP) // maybe some tuneable parameters
1794
    32  float1 (=1.0)
1795
    32  zero ?
1796
    32  float2 (=1.0)
1797
    32  float3 (=1.0)
1798
    32  unknown (27)
1799
    32  unknown (8)
1800
    32  zero ?
1801
    */
1802

    
1803
    if (!avctx->extradata || (avctx->extradata_size < 48)) {
1804
        av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1805
        return -1;
1806
    }
1807

    
1808
    extradata = avctx->extradata;
1809
    extradata_size = avctx->extradata_size;
1810

    
1811
    while (extradata_size > 7) {
1812
        if (!memcmp(extradata, "frmaQDM", 7))
1813
            break;
1814
        extradata++;
1815
        extradata_size--;
1816
    }
1817

    
1818
    if (extradata_size < 12) {
1819
        av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1820
               extradata_size);
1821
        return -1;
1822
    }
1823

    
1824
    if (memcmp(extradata, "frmaQDM", 7)) {
1825
        av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1826
        return -1;
1827
    }
1828

    
1829
    if (extradata[7] == 'C') {
1830
//        s->is_qdmc = 1;
1831
        av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1832
        return -1;
1833
    }
1834

    
1835
    extradata += 8;
1836
    extradata_size -= 8;
1837

    
1838
    size = AV_RB32(extradata);
1839

    
1840
    if(size > extradata_size){
1841
        av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1842
               extradata_size, size);
1843
        return -1;
1844
    }
1845

    
1846
    extradata += 4;
1847
    av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1848
    if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1849
        av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1850
        return -1;
1851
    }
1852

    
1853
    extradata += 8;
1854

    
1855
    avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1856
    extradata += 4;
1857

    
1858
    avctx->sample_rate = AV_RB32(extradata);
1859
    extradata += 4;
1860

    
1861
    avctx->bit_rate = AV_RB32(extradata);
1862
    extradata += 4;
1863

    
1864
    s->group_size = AV_RB32(extradata);
1865
    extradata += 4;
1866

    
1867
    s->fft_size = AV_RB32(extradata);
1868
    extradata += 4;
1869

    
1870
    s->checksum_size = AV_RB32(extradata);
1871
    extradata += 4;
1872

    
1873
    s->fft_order = av_log2(s->fft_size) + 1;
1874
    s->fft_frame_size = 2 * s->fft_size; // complex has two floats
1875

    
1876
    // something like max decodable tones
1877
    s->group_order = av_log2(s->group_size) + 1;
1878
    s->frame_size = s->group_size / 16; // 16 iterations per super block
1879

    
1880
    s->sub_sampling = s->fft_order - 7;
1881
    s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1882

    
1883
    switch ((s->sub_sampling * 2 + s->channels - 1)) {
1884
        case 0: tmp = 40; break;
1885
        case 1: tmp = 48; break;
1886
        case 2: tmp = 56; break;
1887
        case 3: tmp = 72; break;
1888
        case 4: tmp = 80; break;
1889
        case 5: tmp = 100;break;
1890
        default: tmp=s->sub_sampling; break;
1891
    }
1892
    tmp_val = 0;
1893
    if ((tmp * 1000) < avctx->bit_rate)  tmp_val = 1;
1894
    if ((tmp * 1440) < avctx->bit_rate)  tmp_val = 2;
1895
    if ((tmp * 1760) < avctx->bit_rate)  tmp_val = 3;
1896
    if ((tmp * 2240) < avctx->bit_rate)  tmp_val = 4;
1897
    s->cm_table_select = tmp_val;
1898

    
1899
    if (s->sub_sampling == 0)
1900
        tmp = 7999;
1901
    else
1902
        tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1903
    /*
1904
    0: 7999 -> 0
1905
    1: 20000 -> 2
1906
    2: 28000 -> 2
1907
    */
1908
    if (tmp < 8000)
1909
        s->coeff_per_sb_select = 0;
1910
    else if (tmp <= 16000)
1911
        s->coeff_per_sb_select = 1;
1912
    else
1913
        s->coeff_per_sb_select = 2;
1914

    
1915
    // Fail on unknown fft order, if it's > 9 it can overflow s->exptab[]
1916
    if ((s->fft_order < 7) || (s->fft_order > 9)) {
1917
        av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1918
        return -1;
1919
    }
1920

    
1921
    ff_fft_init(&s->fft_ctx, s->fft_order - 1, 1);
1922

    
1923
    for (i = 1; i < (1 << (s->fft_order - 2)); i++) {
1924
        alpha = 2 * M_PI * (float)i / (float)(1 << (s->fft_order - 1));
1925
        s->exptab[i].re = cos(alpha);
1926
        s->exptab[i].im = sin(alpha);
1927
    }
1928

    
1929
    qdm2_init(s);
1930

    
1931
//    dump_context(s);
1932
    return 0;
1933
}
1934

    
1935

    
1936
static int qdm2_decode_close(AVCodecContext *avctx)
1937
{
1938
    QDM2Context *s = avctx->priv_data;
1939

    
1940
    ff_fft_end(&s->fft_ctx);
1941

    
1942
    return 0;
1943
}
1944

    
1945

    
1946
static void qdm2_decode (QDM2Context *q, uint8_t *in, int16_t *out)
1947
{
1948
    int ch, i;
1949
    const int frame_size = (q->frame_size * q->channels);
1950

    
1951
    /* select input buffer */
1952
    q->compressed_data = in;
1953
    q->compressed_size = q->checksum_size;
1954

    
1955
//  dump_context(q);
1956

    
1957
    /* copy old block, clear new block of output samples */
1958
    memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1959
    memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1960

    
1961
    /* decode block of QDM2 compressed data */
1962
    if (q->sub_packet == 0) {
1963
        q->has_errors = 0; // zero it for a new super block
1964
        av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1965
        qdm2_decode_super_block(q);
1966
    }
1967

    
1968
    /* parse subpackets */
1969
    if (!q->has_errors) {
1970
        if (q->sub_packet == 2)
1971
            qdm2_decode_fft_packets(q);
1972

    
1973
        qdm2_fft_tone_synthesizer(q, q->sub_packet);
1974
    }
1975

    
1976
    /* sound synthesis stage 1 (FFT) */
1977
    for (ch = 0; ch < q->channels; ch++) {
1978
        qdm2_calculate_fft(q, ch, q->sub_packet);
1979

    
1980
        if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1981
            SAMPLES_NEEDED_2("has errors, and C list is not empty")
1982
            return;
1983
        }
1984
    }
1985

    
1986
    /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1987
    if (!q->has_errors && q->do_synth_filter)
1988
        qdm2_synthesis_filter(q, q->sub_packet);
1989

    
1990
    q->sub_packet = (q->sub_packet + 1) % 16;
1991

    
1992
    /* clip and convert output float[] to 16bit signed samples */
1993
    for (i = 0; i < frame_size; i++) {
1994
        int value = (int)q->output_buffer[i];
1995

    
1996
        if (value > SOFTCLIP_THRESHOLD)
1997
            value = (value >  HARDCLIP_THRESHOLD) ?  32767 :  softclip_table[ value - SOFTCLIP_THRESHOLD];
1998
        else if (value < -SOFTCLIP_THRESHOLD)
1999
            value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
2000

    
2001
        out[i] = value;
2002
    }
2003
}
2004

    
2005

    
2006
static int qdm2_decode_frame(AVCodecContext *avctx,
2007
            void *data, int *data_size,
2008
            uint8_t *buf, int buf_size)
2009
{
2010
    QDM2Context *s = avctx->priv_data;
2011

    
2012
    if(!buf)
2013
        return 0;
2014
    if(buf_size < s->checksum_size)
2015
        return -1;
2016

    
2017
    *data_size = s->channels * s->frame_size * sizeof(int16_t);
2018

    
2019
    av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
2020
       buf_size, buf, s->checksum_size, data, *data_size);
2021

    
2022
    qdm2_decode(s, buf, data);
2023

    
2024
    // reading only when next superblock found
2025
    if (s->sub_packet == 0) {
2026
        return s->checksum_size;
2027
    }
2028

    
2029
    return 0;
2030
}
2031

    
2032
AVCodec qdm2_decoder =
2033
{
2034
    .name = "qdm2",
2035
    .type = CODEC_TYPE_AUDIO,
2036
    .id = CODEC_ID_QDM2,
2037
    .priv_data_size = sizeof(QDM2Context),
2038
    .init = qdm2_decode_init,
2039
    .close = qdm2_decode_close,
2040
    .decode = qdm2_decode_frame,
2041
};