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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 libavcodec/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 "get_bits.h"
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#include "dsputil.h"
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#include "mpegaudio.h"
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43
#include "qdm2data.h"
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#undef NDEBUG
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#include <assert.h>
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48

    
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#define SOFTCLIP_THRESHOLD 27600
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#define HARDCLIP_THRESHOLD 35716
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52

    
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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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77
#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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80

    
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typedef int8_t sb_int8_array[2][30][64];
82

    
83
/**
84
 * Subpacket
85
 */
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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)
90
} QDM2SubPacket;
91

    
92
/**
93
 * A node in the subpacket list
94
 */
95
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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100
typedef struct {
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    float re;
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    float im;
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} QDM2Complex;
104

    
105
typedef struct {
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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;
115

    
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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;
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124
typedef struct {
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    DECLARE_ALIGNED_16(QDM2Complex, complex[MPA_MAX_CHANNELS][256]);
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} QDM2FFT;
127

    
128
/**
129
 * QDM2 decoder context
130
 */
131
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
150
    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
158
    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
170
    const uint8_t *compressed_data;
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    int compressed_size;
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    float output_buffer[1024];
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174
    /// Synthesis filter
175
    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
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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];
186
    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
191
    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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195
    int sub_packet;
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    int noise_idx; ///< index for dithering noise table
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} QDM2Context;
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199

    
200
static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
201

    
202
static VLC vlc_tab_level;
203
static VLC vlc_tab_diff;
204
static VLC vlc_tab_run;
205
static VLC fft_level_exp_alt_vlc;
206
static VLC fft_level_exp_vlc;
207
static VLC fft_stereo_exp_vlc;
208
static VLC fft_stereo_phase_vlc;
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static VLC vlc_tab_tone_level_idx_hi1;
210
static VLC vlc_tab_tone_level_idx_mid;
211
static VLC vlc_tab_tone_level_idx_hi2;
212
static VLC vlc_tab_type30;
213
static VLC vlc_tab_type34;
214
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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static DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]);
223

    
224

    
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static av_cold void softclip_table_init(void) {
226
    int i;
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    double dfl = SOFTCLIP_THRESHOLD - 32767;
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    float delta = 1.0 / -dfl;
229
    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);
231
}
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233

    
234
// random generated table
235
static av_cold void rnd_table_init(void) {
236
    int i,j;
237
    uint32_t ldw,hdw;
238
    uint64_t tmp64_1;
239
    uint64_t random_seed = 0;
240
    float delta = 1.0 / 16384.0;
241
    for(i = 0; i < 4096 ;i++) {
242
        random_seed = random_seed * 214013 + 2531011;
243
        noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3;
244
    }
245

    
246
    for (i = 0; i < 256 ;i++) {
247
        random_seed = 81;
248
        ldw = i;
249
        for (j = 0; j < 5 ;j++) {
250
            random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
251
            ldw = (uint32_t)ldw % (uint32_t)random_seed;
252
            tmp64_1 = (random_seed * 0x55555556);
253
            hdw = (uint32_t)(tmp64_1 >> 32);
254
            random_seed = (uint64_t)(hdw + (ldw >> 31));
255
        }
256
    }
257
    for (i = 0; i < 128 ;i++) {
258
        random_seed = 25;
259
        ldw = i;
260
        for (j = 0; j < 3 ;j++) {
261
            random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
262
            ldw = (uint32_t)ldw % (uint32_t)random_seed;
263
            tmp64_1 = (random_seed * 0x66666667);
264
            hdw = (uint32_t)(tmp64_1 >> 33);
265
            random_seed = hdw + (ldw >> 31);
266
        }
267
    }
268
}
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270

    
271
static av_cold void init_noise_samples(void) {
272
    int i;
273
    int random_seed = 0;
274
    float delta = 1.0 / 16384.0;
275
    for (i = 0; i < 128;i++) {
276
        random_seed = random_seed * 214013 + 2531011;
277
        noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0);
278
    }
279
}
280

    
281

    
282
static av_cold void qdm2_init_vlc(void)
283
{
284
    init_vlc (&vlc_tab_level, 8, 24,
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        vlc_tab_level_huffbits, 1, 1,
286
        vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
287

    
288
    init_vlc (&vlc_tab_diff, 8, 37,
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        vlc_tab_diff_huffbits, 1, 1,
290
        vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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292
    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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296
    init_vlc (&fft_level_exp_alt_vlc, 8, 28,
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        fft_level_exp_alt_huffbits, 1, 1,
298
        fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
299

    
300
    init_vlc (&fft_level_exp_vlc, 8, 20,
301
        fft_level_exp_huffbits, 1, 1,
302
        fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
303

    
304
    init_vlc (&fft_stereo_exp_vlc, 6, 7,
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        fft_stereo_exp_huffbits, 1, 1,
306
        fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
307

    
308
    init_vlc (&fft_stereo_phase_vlc, 6, 9,
309
        fft_stereo_phase_huffbits, 1, 1,
310
        fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
311

    
312
    init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
313
        vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
314
        vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
315

    
316
    init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
317
        vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
318
        vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
319

    
320
    init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
321
        vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
322
        vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
323

    
324
    init_vlc (&vlc_tab_type30, 6, 9,
325
        vlc_tab_type30_huffbits, 1, 1,
326
        vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
327

    
328
    init_vlc (&vlc_tab_type34, 5, 10,
329
        vlc_tab_type34_huffbits, 1, 1,
330
        vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
331

    
332
    init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
333
        vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
334
        vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
335

    
336
    init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
337
        vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
338
        vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
339

    
340
    init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
341
        vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
342
        vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
343

    
344
    init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
345
        vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
346
        vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
347

    
348
    init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
349
        vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
350
        vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
351
}
352

    
353

    
354
/* for floating point to fixed point conversion */
355
static const float f2i_scale = (float) (1 << (FRAC_BITS - 15));
356

    
357

    
358
static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
359
{
360
    int value;
361

    
362
    value = get_vlc2(gb, vlc->table, vlc->bits, depth);
363

    
364
    /* stage-2, 3 bits exponent escape sequence */
365
    if (value-- == 0)
366
        value = get_bits (gb, get_bits (gb, 3) + 1);
367

    
368
    /* stage-3, optional */
369
    if (flag) {
370
        int tmp = vlc_stage3_values[value];
371

    
372
        if ((value & ~3) > 0)
373
            tmp += get_bits (gb, (value >> 2));
374
        value = tmp;
375
    }
376

    
377
    return value;
378
}
379

    
380

    
381
static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
382
{
383
    int value = qdm2_get_vlc (gb, vlc, 0, depth);
384

    
385
    return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
386
}
387

    
388

    
389
/**
390
 * QDM2 checksum
391
 *
392
 * @param data      pointer to data to be checksum'ed
393
 * @param length    data length
394
 * @param value     checksum value
395
 *
396
 * @return          0 if checksum is OK
397
 */
398
static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
399
    int i;
400

    
401
    for (i=0; i < length; i++)
402
        value -= data[i];
403

    
404
    return (uint16_t)(value & 0xffff);
405
}
406

    
407

    
408
/**
409
 * Fills a QDM2SubPacket structure with packet type, size, and data pointer.
410
 *
411
 * @param gb            bitreader context
412
 * @param sub_packet    packet under analysis
413
 */
414
static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
415
{
416
    sub_packet->type = get_bits (gb, 8);
417

    
418
    if (sub_packet->type == 0) {
419
        sub_packet->size = 0;
420
        sub_packet->data = NULL;
421
    } else {
422
        sub_packet->size = get_bits (gb, 8);
423

    
424
      if (sub_packet->type & 0x80) {
425
          sub_packet->size <<= 8;
426
          sub_packet->size  |= get_bits (gb, 8);
427
          sub_packet->type  &= 0x7f;
428
      }
429

    
430
      if (sub_packet->type == 0x7f)
431
          sub_packet->type |= (get_bits (gb, 8) << 8);
432

    
433
      sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
434
    }
435

    
436
    av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
437
        sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
438
}
439

    
440

    
441
/**
442
 * Return node pointer to first packet of requested type in list.
443
 *
444
 * @param list    list of subpackets to be scanned
445
 * @param type    type of searched subpacket
446
 * @return        node pointer for subpacket if found, else NULL
447
 */
448
static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
449
{
450
    while (list != NULL && list->packet != NULL) {
451
        if (list->packet->type == type)
452
            return list;
453
        list = list->next;
454
    }
455
    return NULL;
456
}
457

    
458

    
459
/**
460
 * Replaces 8 elements with their average value.
461
 * Called by qdm2_decode_superblock before starting subblock decoding.
462
 *
463
 * @param q       context
464
 */
465
static void average_quantized_coeffs (QDM2Context *q)
466
{
467
    int i, j, n, ch, sum;
468

    
469
    n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
470

    
471
    for (ch = 0; ch < q->nb_channels; ch++)
472
        for (i = 0; i < n; i++) {
473
            sum = 0;
474

    
475
            for (j = 0; j < 8; j++)
476
                sum += q->quantized_coeffs[ch][i][j];
477

    
478
            sum /= 8;
479
            if (sum > 0)
480
                sum--;
481

    
482
            for (j=0; j < 8; j++)
483
                q->quantized_coeffs[ch][i][j] = sum;
484
        }
485
}
486

    
487

    
488
/**
489
 * Build subband samples with noise weighted by q->tone_level.
490
 * Called by synthfilt_build_sb_samples.
491
 *
492
 * @param q     context
493
 * @param sb    subband index
494
 */
495
static void build_sb_samples_from_noise (QDM2Context *q, int sb)
496
{
497
    int ch, j;
498

    
499
    FIX_NOISE_IDX(q->noise_idx);
500

    
501
    if (!q->nb_channels)
502
        return;
503

    
504
    for (ch = 0; ch < q->nb_channels; ch++)
505
        for (j = 0; j < 64; j++) {
506
            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);
507
            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);
508
        }
509
}
510

    
511

    
512
/**
513
 * Called while processing data from subpackets 11 and 12.
514
 * Used after making changes to coding_method array.
515
 *
516
 * @param sb               subband index
517
 * @param channels         number of channels
518
 * @param coding_method    q->coding_method[0][0][0]
519
 */
520
static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
521
{
522
    int j,k;
523
    int ch;
524
    int run, case_val;
525
    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};
526

    
527
    for (ch = 0; ch < channels; ch++) {
528
        for (j = 0; j < 64; ) {
529
            if((coding_method[ch][sb][j] - 8) > 22) {
530
                run = 1;
531
                case_val = 8;
532
            } else {
533
                switch (switchtable[coding_method[ch][sb][j]-8]) {
534
                    case 0: run = 10; case_val = 10; break;
535
                    case 1: run = 1; case_val = 16; break;
536
                    case 2: run = 5; case_val = 24; break;
537
                    case 3: run = 3; case_val = 30; break;
538
                    case 4: run = 1; case_val = 30; break;
539
                    case 5: run = 1; case_val = 8; break;
540
                    default: run = 1; case_val = 8; break;
541
                }
542
            }
543
            for (k = 0; k < run; k++)
544
                if (j + k < 128)
545
                    if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
546
                        if (k > 0) {
547
                           SAMPLES_NEEDED
548
                            //not debugged, almost never used
549
                            memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
550
                            memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
551
                        }
552
            j += run;
553
        }
554
    }
555
}
556

    
557

    
558
/**
559
 * Related to synthesis filter
560
 * Called by process_subpacket_10
561
 *
562
 * @param q       context
563
 * @param flag    1 if called after getting data from subpacket 10, 0 if no subpacket 10
564
 */
565
static void fill_tone_level_array (QDM2Context *q, int flag)
566
{
567
    int i, sb, ch, sb_used;
568
    int tmp, tab;
569

    
570
    // This should never happen
571
    if (q->nb_channels <= 0)
572
        return;
573

    
574
    for (ch = 0; ch < q->nb_channels; ch++)
575
        for (sb = 0; sb < 30; sb++)
576
            for (i = 0; i < 8; i++) {
577
                if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
578
                    tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
579
                          q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
580
                else
581
                    tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
582
                if(tmp < 0)
583
                    tmp += 0xff;
584
                q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
585
            }
586

    
587
    sb_used = QDM2_SB_USED(q->sub_sampling);
588

    
589
    if ((q->superblocktype_2_3 != 0) && !flag) {
590
        for (sb = 0; sb < sb_used; sb++)
591
            for (ch = 0; ch < q->nb_channels; ch++)
592
                for (i = 0; i < 64; i++) {
593
                    q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
594
                    if (q->tone_level_idx[ch][sb][i] < 0)
595
                        q->tone_level[ch][sb][i] = 0;
596
                    else
597
                        q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
598
                }
599
    } else {
600
        tab = q->superblocktype_2_3 ? 0 : 1;
601
        for (sb = 0; sb < sb_used; sb++) {
602
            if ((sb >= 4) && (sb <= 23)) {
603
                for (ch = 0; ch < q->nb_channels; ch++)
604
                    for (i = 0; i < 64; i++) {
605
                        tmp = q->tone_level_idx_base[ch][sb][i / 8] -
606
                              q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
607
                              q->tone_level_idx_mid[ch][sb - 4][i / 8] -
608
                              q->tone_level_idx_hi2[ch][sb - 4];
609
                        q->tone_level_idx[ch][sb][i] = tmp & 0xff;
610
                        if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
611
                            q->tone_level[ch][sb][i] = 0;
612
                        else
613
                            q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
614
                }
615
            } else {
616
                if (sb > 4) {
617
                    for (ch = 0; ch < q->nb_channels; ch++)
618
                        for (i = 0; i < 64; i++) {
619
                            tmp = q->tone_level_idx_base[ch][sb][i / 8] -
620
                                  q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
621
                                  q->tone_level_idx_hi2[ch][sb - 4];
622
                            q->tone_level_idx[ch][sb][i] = tmp & 0xff;
623
                            if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
624
                                q->tone_level[ch][sb][i] = 0;
625
                            else
626
                                q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
627
                    }
628
                } else {
629
                    for (ch = 0; ch < q->nb_channels; ch++)
630
                        for (i = 0; i < 64; i++) {
631
                            tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
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
                }
638
            }
639
        }
640
    }
641

    
642
    return;
643
}
644

    
645

    
646
/**
647
 * Related to synthesis filter
648
 * Called by process_subpacket_11
649
 * c is built with data from subpacket 11
650
 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
651
 *
652
 * @param tone_level_idx
653
 * @param tone_level_idx_temp
654
 * @param coding_method        q->coding_method[0][0][0]
655
 * @param nb_channels          number of channels
656
 * @param c                    coming from subpacket 11, passed as 8*c
657
 * @param superblocktype_2_3   flag based on superblock packet type
658
 * @param cm_table_select      q->cm_table_select
659
 */
660
static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
661
                sb_int8_array coding_method, int nb_channels,
662
                int c, int superblocktype_2_3, int cm_table_select)
663
{
664
    int ch, sb, j;
665
    int tmp, acc, esp_40, comp;
666
    int add1, add2, add3, add4;
667
    int64_t multres;
668

    
669
    // This should never happen
670
    if (nb_channels <= 0)
671
        return;
672

    
673
    if (!superblocktype_2_3) {
674
        /* This case is untested, no samples available */
675
        SAMPLES_NEEDED
676
        for (ch = 0; ch < nb_channels; ch++)
677
            for (sb = 0; sb < 30; sb++) {
678
                for (j = 1; j < 63; j++) {  // The loop only iterates to 63 so the code doesn't overflow the buffer
679
                    add1 = tone_level_idx[ch][sb][j] - 10;
680
                    if (add1 < 0)
681
                        add1 = 0;
682
                    add2 = add3 = add4 = 0;
683
                    if (sb > 1) {
684
                        add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
685
                        if (add2 < 0)
686
                            add2 = 0;
687
                    }
688
                    if (sb > 0) {
689
                        add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
690
                        if (add3 < 0)
691
                            add3 = 0;
692
                    }
693
                    if (sb < 29) {
694
                        add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
695
                        if (add4 < 0)
696
                            add4 = 0;
697
                    }
698
                    tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
699
                    if (tmp < 0)
700
                        tmp = 0;
701
                    tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
702
                }
703
                tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
704
            }
705
            acc = 0;
706
            for (ch = 0; ch < nb_channels; ch++)
707
                for (sb = 0; sb < 30; sb++)
708
                    for (j = 0; j < 64; j++)
709
                        acc += tone_level_idx_temp[ch][sb][j];
710
            if (acc)
711
                tmp = c * 256 / (acc & 0xffff);
712
            multres = 0x66666667 * (acc * 10);
713
            esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
714
            for (ch = 0;  ch < nb_channels; ch++)
715
                for (sb = 0; sb < 30; sb++)
716
                    for (j = 0; j < 64; j++) {
717
                        comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
718
                        if (comp < 0)
719
                            comp += 0xff;
720
                        comp /= 256; // signed shift
721
                        switch(sb) {
722
                            case 0:
723
                                if (comp < 30)
724
                                    comp = 30;
725
                                comp += 15;
726
                                break;
727
                            case 1:
728
                                if (comp < 24)
729
                                    comp = 24;
730
                                comp += 10;
731
                                break;
732
                            case 2:
733
                            case 3:
734
                            case 4:
735
                                if (comp < 16)
736
                                    comp = 16;
737
                        }
738
                        if (comp <= 5)
739
                            tmp = 0;
740
                        else if (comp <= 10)
741
                            tmp = 10;
742
                        else if (comp <= 16)
743
                            tmp = 16;
744
                        else if (comp <= 24)
745
                            tmp = -1;
746
                        else
747
                            tmp = 0;
748
                        coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
749
                    }
750
            for (sb = 0; sb < 30; sb++)
751
                fix_coding_method_array(sb, nb_channels, coding_method);
752
            for (ch = 0; ch < nb_channels; ch++)
753
                for (sb = 0; sb < 30; sb++)
754
                    for (j = 0; j < 64; j++)
755
                        if (sb >= 10) {
756
                            if (coding_method[ch][sb][j] < 10)
757
                                coding_method[ch][sb][j] = 10;
758
                        } else {
759
                            if (sb >= 2) {
760
                                if (coding_method[ch][sb][j] < 16)
761
                                    coding_method[ch][sb][j] = 16;
762
                            } else {
763
                                if (coding_method[ch][sb][j] < 30)
764
                                    coding_method[ch][sb][j] = 30;
765
                            }
766
                        }
767
    } else { // superblocktype_2_3 != 0
768
        for (ch = 0; ch < nb_channels; ch++)
769
            for (sb = 0; sb < 30; sb++)
770
                for (j = 0; j < 64; j++)
771
                    coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
772
    }
773

    
774
    return;
775
}
776

    
777

    
778
/**
779
 *
780
 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
781
 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
782
 *
783
 * @param q         context
784
 * @param gb        bitreader context
785
 * @param length    packet length in bits
786
 * @param sb_min    lower subband processed (sb_min included)
787
 * @param sb_max    higher subband processed (sb_max excluded)
788
 */
789
static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
790
{
791
    int sb, j, k, n, ch, run, channels;
792
    int joined_stereo, zero_encoding, chs;
793
    int type34_first;
794
    float type34_div = 0;
795
    float type34_predictor;
796
    float samples[10], sign_bits[16];
797

    
798
    if (length == 0) {
799
        // If no data use noise
800
        for (sb=sb_min; sb < sb_max; sb++)
801
            build_sb_samples_from_noise (q, sb);
802

    
803
        return;
804
    }
805

    
806
    for (sb = sb_min; sb < sb_max; sb++) {
807
        FIX_NOISE_IDX(q->noise_idx);
808

    
809
        channels = q->nb_channels;
810

    
811
        if (q->nb_channels <= 1 || sb < 12)
812
            joined_stereo = 0;
813
        else if (sb >= 24)
814
            joined_stereo = 1;
815
        else
816
            joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
817

    
818
        if (joined_stereo) {
819
            if (BITS_LEFT(length,gb) >= 16)
820
                for (j = 0; j < 16; j++)
821
                    sign_bits[j] = get_bits1 (gb);
822

    
823
            for (j = 0; j < 64; j++)
824
                if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
825
                    q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
826

    
827
            fix_coding_method_array(sb, q->nb_channels, q->coding_method);
828
            channels = 1;
829
        }
830

    
831
        for (ch = 0; ch < channels; ch++) {
832
            zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
833
            type34_predictor = 0.0;
834
            type34_first = 1;
835

    
836
            for (j = 0; j < 128; ) {
837
                switch (q->coding_method[ch][sb][j / 2]) {
838
                    case 8:
839
                        if (BITS_LEFT(length,gb) >= 10) {
840
                            if (zero_encoding) {
841
                                for (k = 0; k < 5; k++) {
842
                                    if ((j + 2 * k) >= 128)
843
                                        break;
844
                                    samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
845
                                }
846
                            } else {
847
                                n = get_bits(gb, 8);
848
                                for (k = 0; k < 5; k++)
849
                                    samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
850
                            }
851
                            for (k = 0; k < 5; k++)
852
                                samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
853
                        } else {
854
                            for (k = 0; k < 10; k++)
855
                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
856
                        }
857
                        run = 10;
858
                        break;
859

    
860
                    case 10:
861
                        if (BITS_LEFT(length,gb) >= 1) {
862
                            float f = 0.81;
863

    
864
                            if (get_bits1(gb))
865
                                f = -f;
866
                            f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
867
                            samples[0] = f;
868
                        } else {
869
                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
870
                        }
871
                        run = 1;
872
                        break;
873

    
874
                    case 16:
875
                        if (BITS_LEFT(length,gb) >= 10) {
876
                            if (zero_encoding) {
877
                                for (k = 0; k < 5; k++) {
878
                                    if ((j + k) >= 128)
879
                                        break;
880
                                    samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
881
                                }
882
                            } else {
883
                                n = get_bits (gb, 8);
884
                                for (k = 0; k < 5; k++)
885
                                    samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
886
                            }
887
                        } else {
888
                            for (k = 0; k < 5; k++)
889
                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
890
                        }
891
                        run = 5;
892
                        break;
893

    
894
                    case 24:
895
                        if (BITS_LEFT(length,gb) >= 7) {
896
                            n = get_bits(gb, 7);
897
                            for (k = 0; k < 3; k++)
898
                                samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
899
                        } else {
900
                            for (k = 0; k < 3; k++)
901
                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
902
                        }
903
                        run = 3;
904
                        break;
905

    
906
                    case 30:
907
                        if (BITS_LEFT(length,gb) >= 4)
908
                            samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
909
                        else
910
                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
911

    
912
                        run = 1;
913
                        break;
914

    
915
                    case 34:
916
                        if (BITS_LEFT(length,gb) >= 7) {
917
                            if (type34_first) {
918
                                type34_div = (float)(1 << get_bits(gb, 2));
919
                                samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
920
                                type34_predictor = samples[0];
921
                                type34_first = 0;
922
                            } else {
923
                                samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
924
                                type34_predictor = samples[0];
925
                            }
926
                        } else {
927
                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
928
                        }
929
                        run = 1;
930
                        break;
931

    
932
                    default:
933
                        samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
934
                        run = 1;
935
                        break;
936
                }
937

    
938
                if (joined_stereo) {
939
                    float tmp[10][MPA_MAX_CHANNELS];
940

    
941
                    for (k = 0; k < run; k++) {
942
                        tmp[k][0] = samples[k];
943
                        tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
944
                    }
945
                    for (chs = 0; chs < q->nb_channels; chs++)
946
                        for (k = 0; k < run; k++)
947
                            if ((j + k) < 128)
948
                                q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
949
                } else {
950
                    for (k = 0; k < run; k++)
951
                        if ((j + k) < 128)
952
                            q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
953
                }
954

    
955
                j += run;
956
            } // j loop
957
        } // channel loop
958
    } // subband loop
959
}
960

    
961

    
962
/**
963
 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
964
 * This is similar to process_subpacket_9, but for a single channel and for element [0]
965
 * same VLC tables as process_subpacket_9 are used.
966
 *
967
 * @param q         context
968
 * @param quantized_coeffs    pointer to quantized_coeffs[ch][0]
969
 * @param gb        bitreader context
970
 * @param length    packet length in bits
971
 */
972
static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
973
{
974
    int i, k, run, level, diff;
975

    
976
    if (BITS_LEFT(length,gb) < 16)
977
        return;
978
    level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
979

    
980
    quantized_coeffs[0] = level;
981

    
982
    for (i = 0; i < 7; ) {
983
        if (BITS_LEFT(length,gb) < 16)
984
            break;
985
        run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
986

    
987
        if (BITS_LEFT(length,gb) < 16)
988
            break;
989
        diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
990

    
991
        for (k = 1; k <= run; k++)
992
            quantized_coeffs[i + k] = (level + ((k * diff) / run));
993

    
994
        level += diff;
995
        i += run;
996
    }
997
}
998

    
999

    
1000
/**
1001
 * Related to synthesis filter, process data from packet 10
1002
 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
1003
 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
1004
 *
1005
 * @param q         context
1006
 * @param gb        bitreader context
1007
 * @param length    packet length in bits
1008
 */
1009
static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
1010
{
1011
    int sb, j, k, n, ch;
1012

    
1013
    for (ch = 0; ch < q->nb_channels; ch++) {
1014
        init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
1015

    
1016
        if (BITS_LEFT(length,gb) < 16) {
1017
            memset(q->quantized_coeffs[ch][0], 0, 8);
1018
            break;
1019
        }
1020
    }
1021

    
1022
    n = q->sub_sampling + 1;
1023

    
1024
    for (sb = 0; sb < n; sb++)
1025
        for (ch = 0; ch < q->nb_channels; ch++)
1026
            for (j = 0; j < 8; j++) {
1027
                if (BITS_LEFT(length,gb) < 1)
1028
                    break;
1029
                if (get_bits1(gb)) {
1030
                    for (k=0; k < 8; k++) {
1031
                        if (BITS_LEFT(length,gb) < 16)
1032
                            break;
1033
                        q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1034
                    }
1035
                } else {
1036
                    for (k=0; k < 8; k++)
1037
                        q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1038
                }
1039
            }
1040

    
1041
    n = QDM2_SB_USED(q->sub_sampling) - 4;
1042

    
1043
    for (sb = 0; sb < n; sb++)
1044
        for (ch = 0; ch < q->nb_channels; ch++) {
1045
            if (BITS_LEFT(length,gb) < 16)
1046
                break;
1047
            q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1048
            if (sb > 19)
1049
                q->tone_level_idx_hi2[ch][sb] -= 16;
1050
            else
1051
                for (j = 0; j < 8; j++)
1052
                    q->tone_level_idx_mid[ch][sb][j] = -16;
1053
        }
1054

    
1055
    n = QDM2_SB_USED(q->sub_sampling) - 5;
1056

    
1057
    for (sb = 0; sb < n; sb++)
1058
        for (ch = 0; ch < q->nb_channels; ch++)
1059
            for (j = 0; j < 8; j++) {
1060
                if (BITS_LEFT(length,gb) < 16)
1061
                    break;
1062
                q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1063
            }
1064
}
1065

    
1066
/**
1067
 * Process subpacket 9, init quantized_coeffs with data from it
1068
 *
1069
 * @param q       context
1070
 * @param node    pointer to node with packet
1071
 */
1072
static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
1073
{
1074
    GetBitContext gb;
1075
    int i, j, k, n, ch, run, level, diff;
1076

    
1077
    init_get_bits(&gb, node->packet->data, node->packet->size*8);
1078

    
1079
    n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1080

    
1081
    for (i = 1; i < n; i++)
1082
        for (ch=0; ch < q->nb_channels; ch++) {
1083
            level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1084
            q->quantized_coeffs[ch][i][0] = level;
1085

    
1086
            for (j = 0; j < (8 - 1); ) {
1087
                run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1088
                diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1089

    
1090
                for (k = 1; k <= run; k++)
1091
                    q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1092

    
1093
                level += diff;
1094
                j += run;
1095
            }
1096
        }
1097

    
1098
    for (ch = 0; ch < q->nb_channels; ch++)
1099
        for (i = 0; i < 8; i++)
1100
            q->quantized_coeffs[ch][0][i] = 0;
1101
}
1102

    
1103

    
1104
/**
1105
 * Process subpacket 10 if not null, else
1106
 *
1107
 * @param q         context
1108
 * @param node      pointer to node with packet
1109
 * @param length    packet length in bits
1110
 */
1111
static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
1112
{
1113
    GetBitContext gb;
1114

    
1115
    init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1116

    
1117
    if (length != 0) {
1118
        init_tone_level_dequantization(q, &gb, length);
1119
        fill_tone_level_array(q, 1);
1120
    } else {
1121
        fill_tone_level_array(q, 0);
1122
    }
1123
}
1124

    
1125

    
1126
/**
1127
 * Process subpacket 11
1128
 *
1129
 * @param q         context
1130
 * @param node      pointer to node with packet
1131
 * @param length    packet length in bit
1132
 */
1133
static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
1134
{
1135
    GetBitContext gb;
1136

    
1137
    init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1138
    if (length >= 32) {
1139
        int c = get_bits (&gb, 13);
1140

    
1141
        if (c > 3)
1142
            fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
1143
                                      q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
1144
    }
1145

    
1146
    synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1147
}
1148

    
1149

    
1150
/**
1151
 * Process subpacket 12
1152
 *
1153
 * @param q         context
1154
 * @param node      pointer to node with packet
1155
 * @param length    packet length in bits
1156
 */
1157
static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
1158
{
1159
    GetBitContext gb;
1160

    
1161
    init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1162
    synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1163
}
1164

    
1165
/*
1166
 * Process new subpackets for synthesis filter
1167
 *
1168
 * @param q       context
1169
 * @param list    list with synthesis filter packets (list D)
1170
 */
1171
static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
1172
{
1173
    QDM2SubPNode *nodes[4];
1174

    
1175
    nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1176
    if (nodes[0] != NULL)
1177
        process_subpacket_9(q, nodes[0]);
1178

    
1179
    nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1180
    if (nodes[1] != NULL)
1181
        process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
1182
    else
1183
        process_subpacket_10(q, NULL, 0);
1184

    
1185
    nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1186
    if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1187
        process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
1188
    else
1189
        process_subpacket_11(q, NULL, 0);
1190

    
1191
    nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1192
    if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1193
        process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
1194
    else
1195
        process_subpacket_12(q, NULL, 0);
1196
}
1197

    
1198

    
1199
/*
1200
 * Decode superblock, fill packet lists.
1201
 *
1202
 * @param q    context
1203
 */
1204
static void qdm2_decode_super_block (QDM2Context *q)
1205
{
1206
    GetBitContext gb;
1207
    QDM2SubPacket header, *packet;
1208
    int i, packet_bytes, sub_packet_size, sub_packets_D;
1209
    unsigned int next_index = 0;
1210

    
1211
    memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1212
    memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1213
    memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1214

    
1215
    q->sub_packets_B = 0;
1216
    sub_packets_D = 0;
1217

    
1218
    average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1219

    
1220
    init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
1221
    qdm2_decode_sub_packet_header(&gb, &header);
1222

    
1223
    if (header.type < 2 || header.type >= 8) {
1224
        q->has_errors = 1;
1225
        av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1226
        return;
1227
    }
1228

    
1229
    q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1230
    packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1231

    
1232
    init_get_bits(&gb, header.data, header.size*8);
1233

    
1234
    if (header.type == 2 || header.type == 4 || header.type == 5) {
1235
        int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8);
1236

    
1237
        csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1238

    
1239
        if (csum != 0) {
1240
            q->has_errors = 1;
1241
            av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1242
            return;
1243
        }
1244
    }
1245

    
1246
    q->sub_packet_list_B[0].packet = NULL;
1247
    q->sub_packet_list_D[0].packet = NULL;
1248

    
1249
    for (i = 0; i < 6; i++)
1250
        if (--q->fft_level_exp[i] < 0)
1251
            q->fft_level_exp[i] = 0;
1252

    
1253
    for (i = 0; packet_bytes > 0; i++) {
1254
        int j;
1255

    
1256
        q->sub_packet_list_A[i].next = NULL;
1257

    
1258
        if (i > 0) {
1259
            q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1260

    
1261
            /* seek to next block */
1262
            init_get_bits(&gb, header.data, header.size*8);
1263
            skip_bits(&gb, next_index*8);
1264

    
1265
            if (next_index >= header.size)
1266
                break;
1267
        }
1268

    
1269
        /* decode subpacket */
1270
        packet = &q->sub_packets[i];
1271
        qdm2_decode_sub_packet_header(&gb, packet);
1272
        next_index = packet->size + get_bits_count(&gb) / 8;
1273
        sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1274

    
1275
        if (packet->type == 0)
1276
            break;
1277

    
1278
        if (sub_packet_size > packet_bytes) {
1279
            if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1280
                break;
1281
            packet->size += packet_bytes - sub_packet_size;
1282
        }
1283

    
1284
        packet_bytes -= sub_packet_size;
1285

    
1286
        /* add subpacket to 'all subpackets' list */
1287
        q->sub_packet_list_A[i].packet = packet;
1288

    
1289
        /* add subpacket to related list */
1290
        if (packet->type == 8) {
1291
            SAMPLES_NEEDED_2("packet type 8");
1292
            return;
1293
        } else if (packet->type >= 9 && packet->type <= 12) {
1294
            /* packets for MPEG Audio like Synthesis Filter */
1295
            QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1296
        } else if (packet->type == 13) {
1297
            for (j = 0; j < 6; j++)
1298
                q->fft_level_exp[j] = get_bits(&gb, 6);
1299
        } else if (packet->type == 14) {
1300
            for (j = 0; j < 6; j++)
1301
                q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1302
        } else if (packet->type == 15) {
1303
            SAMPLES_NEEDED_2("packet type 15")
1304
            return;
1305
        } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1306
            /* packets for FFT */
1307
            QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1308
        }
1309
    } // Packet bytes loop
1310

    
1311
/* **************************************************************** */
1312
    if (q->sub_packet_list_D[0].packet != NULL) {
1313
        process_synthesis_subpackets(q, q->sub_packet_list_D);
1314
        q->do_synth_filter = 1;
1315
    } else if (q->do_synth_filter) {
1316
        process_subpacket_10(q, NULL, 0);
1317
        process_subpacket_11(q, NULL, 0);
1318
        process_subpacket_12(q, NULL, 0);
1319
    }
1320
/* **************************************************************** */
1321
}
1322

    
1323

    
1324
static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1325
                       int offset, int duration, int channel,
1326
                       int exp, int phase)
1327
{
1328
    if (q->fft_coefs_min_index[duration] < 0)
1329
        q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1330

    
1331
    q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1332
    q->fft_coefs[q->fft_coefs_index].channel = channel;
1333
    q->fft_coefs[q->fft_coefs_index].offset = offset;
1334
    q->fft_coefs[q->fft_coefs_index].exp = exp;
1335
    q->fft_coefs[q->fft_coefs_index].phase = phase;
1336
    q->fft_coefs_index++;
1337
}
1338

    
1339

    
1340
static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
1341
{
1342
    int channel, stereo, phase, exp;
1343
    int local_int_4,  local_int_8,  stereo_phase,  local_int_10;
1344
    int local_int_14, stereo_exp, local_int_20, local_int_28;
1345
    int n, offset;
1346

    
1347
    local_int_4 = 0;
1348
    local_int_28 = 0;
1349
    local_int_20 = 2;
1350
    local_int_8 = (4 - duration);
1351
    local_int_10 = 1 << (q->group_order - duration - 1);
1352
    offset = 1;
1353

    
1354
    while (1) {
1355
        if (q->superblocktype_2_3) {
1356
            while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1357
                offset = 1;
1358
                if (n == 0) {
1359
                    local_int_4 += local_int_10;
1360
                    local_int_28 += (1 << local_int_8);
1361
                } else {
1362
                    local_int_4 += 8*local_int_10;
1363
                    local_int_28 += (8 << local_int_8);
1364
                }
1365
            }
1366
            offset += (n - 2);
1367
        } else {
1368
            offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1369
            while (offset >= (local_int_10 - 1)) {
1370
                offset += (1 - (local_int_10 - 1));
1371
                local_int_4  += local_int_10;
1372
                local_int_28 += (1 << local_int_8);
1373
            }
1374
        }
1375

    
1376
        if (local_int_4 >= q->group_size)
1377
            return;
1378

    
1379
        local_int_14 = (offset >> local_int_8);
1380

    
1381
        if (q->nb_channels > 1) {
1382
            channel = get_bits1(gb);
1383
            stereo = get_bits1(gb);
1384
        } else {
1385
            channel = 0;
1386
            stereo = 0;
1387
        }
1388

    
1389
        exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1390
        exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1391
        exp = (exp < 0) ? 0 : exp;
1392

    
1393
        phase = get_bits(gb, 3);
1394
        stereo_exp = 0;
1395
        stereo_phase = 0;
1396

    
1397
        if (stereo) {
1398
            stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1399
            stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1400
            if (stereo_phase < 0)
1401
                stereo_phase += 8;
1402
        }
1403

    
1404
        if (q->frequency_range > (local_int_14 + 1)) {
1405
            int sub_packet = (local_int_20 + local_int_28);
1406

    
1407
            qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1408
            if (stereo)
1409
                qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1410
        }
1411

    
1412
        offset++;
1413
    }
1414
}
1415

    
1416

    
1417
static void qdm2_decode_fft_packets (QDM2Context *q)
1418
{
1419
    int i, j, min, max, value, type, unknown_flag;
1420
    GetBitContext gb;
1421

    
1422
    if (q->sub_packet_list_B[0].packet == NULL)
1423
        return;
1424

    
1425
    /* reset minimum indexes for FFT coefficients */
1426
    q->fft_coefs_index = 0;
1427
    for (i=0; i < 5; i++)
1428
        q->fft_coefs_min_index[i] = -1;
1429

    
1430
    /* process subpackets ordered by type, largest type first */
1431
    for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1432
        QDM2SubPacket *packet= NULL;
1433

    
1434
        /* find subpacket with largest type less than max */
1435
        for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1436
            value = q->sub_packet_list_B[j].packet->type;
1437
            if (value > min && value < max) {
1438
                min = value;
1439
                packet = q->sub_packet_list_B[j].packet;
1440
            }
1441
        }
1442

    
1443
        max = min;
1444

    
1445
        /* check for errors (?) */
1446
        if (!packet)
1447
            return;
1448

    
1449
        if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1450
            return;
1451

    
1452
        /* decode FFT tones */
1453
        init_get_bits (&gb, packet->data, packet->size*8);
1454

    
1455
        if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1456
            unknown_flag = 1;
1457
        else
1458
            unknown_flag = 0;
1459

    
1460
        type = packet->type;
1461

    
1462
        if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1463
            int duration = q->sub_sampling + 5 - (type & 15);
1464

    
1465
            if (duration >= 0 && duration < 4)
1466
                qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1467
        } else if (type == 31) {
1468
            for (j=0; j < 4; j++)
1469
                qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1470
        } else if (type == 46) {
1471
            for (j=0; j < 6; j++)
1472
                q->fft_level_exp[j] = get_bits(&gb, 6);
1473
            for (j=0; j < 4; j++)
1474
            qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1475
        }
1476
    } // Loop on B packets
1477

    
1478
    /* calculate maximum indexes for FFT coefficients */
1479
    for (i = 0, j = -1; i < 5; i++)
1480
        if (q->fft_coefs_min_index[i] >= 0) {
1481
            if (j >= 0)
1482
                q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1483
            j = i;
1484
        }
1485
    if (j >= 0)
1486
        q->fft_coefs_max_index[j] = q->fft_coefs_index;
1487
}
1488

    
1489

    
1490
static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
1491
{
1492
   float level, f[6];
1493
   int i;
1494
   QDM2Complex c;
1495
   const double iscale = 2.0*M_PI / 512.0;
1496

    
1497
    tone->phase += tone->phase_shift;
1498

    
1499
    /* calculate current level (maximum amplitude) of tone */
1500
    level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1501
    c.im = level * sin(tone->phase*iscale);
1502
    c.re = level * cos(tone->phase*iscale);
1503

    
1504
    /* generate FFT coefficients for tone */
1505
    if (tone->duration >= 3 || tone->cutoff >= 3) {
1506
        tone->complex[0].im += c.im;
1507
        tone->complex[0].re += c.re;
1508
        tone->complex[1].im -= c.im;
1509
        tone->complex[1].re -= c.re;
1510
    } else {
1511
        f[1] = -tone->table[4];
1512
        f[0] =  tone->table[3] - tone->table[0];
1513
        f[2] =  1.0 - tone->table[2] - tone->table[3];
1514
        f[3] =  tone->table[1] + tone->table[4] - 1.0;
1515
        f[4] =  tone->table[0] - tone->table[1];
1516
        f[5] =  tone->table[2];
1517
        for (i = 0; i < 2; i++) {
1518
            tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
1519
            tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1520
        }
1521
        for (i = 0; i < 4; i++) {
1522
            tone->complex[i].re += c.re * f[i+2];
1523
            tone->complex[i].im += c.im * f[i+2];
1524
        }
1525
    }
1526

    
1527
    /* copy the tone if it has not yet died out */
1528
    if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1529
      memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1530
      q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1531
    }
1532
}
1533

    
1534

    
1535
static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1536
{
1537
    int i, j, ch;
1538
    const double iscale = 0.25 * M_PI;
1539

    
1540
    for (ch = 0; ch < q->channels; ch++) {
1541
        memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1542
    }
1543

    
1544

    
1545
    /* apply FFT tones with duration 4 (1 FFT period) */
1546
    if (q->fft_coefs_min_index[4] >= 0)
1547
        for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1548
            float level;
1549
            QDM2Complex c;
1550

    
1551
            if (q->fft_coefs[i].sub_packet != sub_packet)
1552
                break;
1553

    
1554
            ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1555
            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];
1556

    
1557
            c.re = level * cos(q->fft_coefs[i].phase * iscale);
1558
            c.im = level * sin(q->fft_coefs[i].phase * iscale);
1559
            q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1560
            q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1561
            q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1562
            q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1563
        }
1564

    
1565
    /* generate existing FFT tones */
1566
    for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1567
        qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1568
        q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1569
    }
1570

    
1571
    /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1572
    for (i = 0; i < 4; i++)
1573
        if (q->fft_coefs_min_index[i] >= 0) {
1574
            for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1575
                int offset, four_i;
1576
                FFTTone tone;
1577

    
1578
                if (q->fft_coefs[j].sub_packet != sub_packet)
1579
                    break;
1580

    
1581
                four_i = (4 - i);
1582
                offset = q->fft_coefs[j].offset >> four_i;
1583
                ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1584

    
1585
                if (offset < q->frequency_range) {
1586
                    if (offset < 2)
1587
                        tone.cutoff = offset;
1588
                    else
1589
                        tone.cutoff = (offset >= 60) ? 3 : 2;
1590

    
1591
                    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];
1592
                    tone.complex = &q->fft.complex[ch][offset];
1593
                    tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1594
                    tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1595
                    tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1596
                    tone.duration = i;
1597
                    tone.time_index = 0;
1598

    
1599
                    qdm2_fft_generate_tone(q, &tone);
1600
                }
1601
            }
1602
            q->fft_coefs_min_index[i] = j;
1603
        }
1604
}
1605

    
1606

    
1607
static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1608
{
1609
    const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1610
    int i;
1611
    q->fft.complex[channel][0].re *= 2.0f;
1612
    q->fft.complex[channel][0].im = 0.0f;
1613
    ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1614
    /* add samples to output buffer */
1615
    for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
1616
        q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
1617
}
1618

    
1619

    
1620
/**
1621
 * @param q        context
1622
 * @param index    subpacket number
1623
 */
1624
static void qdm2_synthesis_filter (QDM2Context *q, int index)
1625
{
1626
    OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
1627
    int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1628

    
1629
    /* copy sb_samples */
1630
    sb_used = QDM2_SB_USED(q->sub_sampling);
1631

    
1632
    for (ch = 0; ch < q->channels; ch++)
1633
        for (i = 0; i < 8; i++)
1634
            for (k=sb_used; k < SBLIMIT; k++)
1635
                q->sb_samples[ch][(8 * index) + i][k] = 0;
1636

    
1637
    for (ch = 0; ch < q->nb_channels; ch++) {
1638
        OUT_INT *samples_ptr = samples + ch;
1639

    
1640
        for (i = 0; i < 8; i++) {
1641
            ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1642
                mpa_window, &dither_state,
1643
                samples_ptr, q->nb_channels,
1644
                q->sb_samples[ch][(8 * index) + i]);
1645
            samples_ptr += 32 * q->nb_channels;
1646
        }
1647
    }
1648

    
1649
    /* add samples to output buffer */
1650
    sub_sampling = (4 >> q->sub_sampling);
1651

    
1652
    for (ch = 0; ch < q->channels; ch++)
1653
        for (i = 0; i < q->frame_size; i++)
1654
            q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
1655
}
1656

    
1657

    
1658
/**
1659
 * Init static data (does not depend on specific file)
1660
 *
1661
 * @param q    context
1662
 */
1663
static av_cold void qdm2_init(QDM2Context *q) {
1664
    static int initialized = 0;
1665

    
1666
    if (initialized != 0)
1667
        return;
1668
    initialized = 1;
1669

    
1670
    qdm2_init_vlc();
1671
    ff_mpa_synth_init(mpa_window);
1672
    softclip_table_init();
1673
    rnd_table_init();
1674
    init_noise_samples();
1675

    
1676
    av_log(NULL, AV_LOG_DEBUG, "init done\n");
1677
}
1678

    
1679

    
1680
#if 0
1681
static void dump_context(QDM2Context *q)
1682
{
1683
    int i;
1684
#define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
1685
    PRINT("compressed_data",q->compressed_data);
1686
    PRINT("compressed_size",q->compressed_size);
1687
    PRINT("frame_size",q->frame_size);
1688
    PRINT("checksum_size",q->checksum_size);
1689
    PRINT("channels",q->channels);
1690
    PRINT("nb_channels",q->nb_channels);
1691
    PRINT("fft_frame_size",q->fft_frame_size);
1692
    PRINT("fft_size",q->fft_size);
1693
    PRINT("sub_sampling",q->sub_sampling);
1694
    PRINT("fft_order",q->fft_order);
1695
    PRINT("group_order",q->group_order);
1696
    PRINT("group_size",q->group_size);
1697
    PRINT("sub_packet",q->sub_packet);
1698
    PRINT("frequency_range",q->frequency_range);
1699
    PRINT("has_errors",q->has_errors);
1700
    PRINT("fft_tone_end",q->fft_tone_end);
1701
    PRINT("fft_tone_start",q->fft_tone_start);
1702
    PRINT("fft_coefs_index",q->fft_coefs_index);
1703
    PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
1704
    PRINT("cm_table_select",q->cm_table_select);
1705
    PRINT("noise_idx",q->noise_idx);
1706

1707
    for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
1708
    {
1709
    FFTTone *t = &q->fft_tones[i];
1710

1711
    av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
1712
    av_log(NULL,AV_LOG_DEBUG,"  level = %f\n", t->level);
1713
//  PRINT(" level", t->level);
1714
    PRINT(" phase", t->phase);
1715
    PRINT(" phase_shift", t->phase_shift);
1716
    PRINT(" duration", t->duration);
1717
    PRINT(" samples_im", t->samples_im);
1718
    PRINT(" samples_re", t->samples_re);
1719
    PRINT(" table", t->table);
1720
    }
1721

1722
}
1723
#endif
1724

    
1725

    
1726
/**
1727
 * Init parameters from codec extradata
1728
 */
1729
static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1730
{
1731
    QDM2Context *s = avctx->priv_data;
1732
    uint8_t *extradata;
1733
    int extradata_size;
1734
    int tmp_val, tmp, size;
1735

    
1736
    /* extradata parsing
1737

1738
    Structure:
1739
    wave {
1740
        frma (QDM2)
1741
        QDCA
1742
        QDCP
1743
    }
1744

1745
    32  size (including this field)
1746
    32  tag (=frma)
1747
    32  type (=QDM2 or QDMC)
1748

1749
    32  size (including this field, in bytes)
1750
    32  tag (=QDCA) // maybe mandatory parameters
1751
    32  unknown (=1)
1752
    32  channels (=2)
1753
    32  samplerate (=44100)
1754
    32  bitrate (=96000)
1755
    32  block size (=4096)
1756
    32  frame size (=256) (for one channel)
1757
    32  packet size (=1300)
1758

1759
    32  size (including this field, in bytes)
1760
    32  tag (=QDCP) // maybe some tuneable parameters
1761
    32  float1 (=1.0)
1762
    32  zero ?
1763
    32  float2 (=1.0)
1764
    32  float3 (=1.0)
1765
    32  unknown (27)
1766
    32  unknown (8)
1767
    32  zero ?
1768
    */
1769

    
1770
    if (!avctx->extradata || (avctx->extradata_size < 48)) {
1771
        av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1772
        return -1;
1773
    }
1774

    
1775
    extradata = avctx->extradata;
1776
    extradata_size = avctx->extradata_size;
1777

    
1778
    while (extradata_size > 7) {
1779
        if (!memcmp(extradata, "frmaQDM", 7))
1780
            break;
1781
        extradata++;
1782
        extradata_size--;
1783
    }
1784

    
1785
    if (extradata_size < 12) {
1786
        av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1787
               extradata_size);
1788
        return -1;
1789
    }
1790

    
1791
    if (memcmp(extradata, "frmaQDM", 7)) {
1792
        av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1793
        return -1;
1794
    }
1795

    
1796
    if (extradata[7] == 'C') {
1797
//        s->is_qdmc = 1;
1798
        av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1799
        return -1;
1800
    }
1801

    
1802
    extradata += 8;
1803
    extradata_size -= 8;
1804

    
1805
    size = AV_RB32(extradata);
1806

    
1807
    if(size > extradata_size){
1808
        av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1809
               extradata_size, size);
1810
        return -1;
1811
    }
1812

    
1813
    extradata += 4;
1814
    av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1815
    if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1816
        av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1817
        return -1;
1818
    }
1819

    
1820
    extradata += 8;
1821

    
1822
    avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1823
    extradata += 4;
1824

    
1825
    avctx->sample_rate = AV_RB32(extradata);
1826
    extradata += 4;
1827

    
1828
    avctx->bit_rate = AV_RB32(extradata);
1829
    extradata += 4;
1830

    
1831
    s->group_size = AV_RB32(extradata);
1832
    extradata += 4;
1833

    
1834
    s->fft_size = AV_RB32(extradata);
1835
    extradata += 4;
1836

    
1837
    s->checksum_size = AV_RB32(extradata);
1838
    extradata += 4;
1839

    
1840
    s->fft_order = av_log2(s->fft_size) + 1;
1841
    s->fft_frame_size = 2 * s->fft_size; // complex has two floats
1842

    
1843
    // something like max decodable tones
1844
    s->group_order = av_log2(s->group_size) + 1;
1845
    s->frame_size = s->group_size / 16; // 16 iterations per super block
1846

    
1847
    s->sub_sampling = s->fft_order - 7;
1848
    s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1849

    
1850
    switch ((s->sub_sampling * 2 + s->channels - 1)) {
1851
        case 0: tmp = 40; break;
1852
        case 1: tmp = 48; break;
1853
        case 2: tmp = 56; break;
1854
        case 3: tmp = 72; break;
1855
        case 4: tmp = 80; break;
1856
        case 5: tmp = 100;break;
1857
        default: tmp=s->sub_sampling; break;
1858
    }
1859
    tmp_val = 0;
1860
    if ((tmp * 1000) < avctx->bit_rate)  tmp_val = 1;
1861
    if ((tmp * 1440) < avctx->bit_rate)  tmp_val = 2;
1862
    if ((tmp * 1760) < avctx->bit_rate)  tmp_val = 3;
1863
    if ((tmp * 2240) < avctx->bit_rate)  tmp_val = 4;
1864
    s->cm_table_select = tmp_val;
1865

    
1866
    if (s->sub_sampling == 0)
1867
        tmp = 7999;
1868
    else
1869
        tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1870
    /*
1871
    0: 7999 -> 0
1872
    1: 20000 -> 2
1873
    2: 28000 -> 2
1874
    */
1875
    if (tmp < 8000)
1876
        s->coeff_per_sb_select = 0;
1877
    else if (tmp <= 16000)
1878
        s->coeff_per_sb_select = 1;
1879
    else
1880
        s->coeff_per_sb_select = 2;
1881

    
1882
    // Fail on unknown fft order
1883
    if ((s->fft_order < 7) || (s->fft_order > 9)) {
1884
        av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1885
        return -1;
1886
    }
1887

    
1888
    ff_rdft_init(&s->rdft_ctx, s->fft_order, IRDFT);
1889

    
1890
    qdm2_init(s);
1891

    
1892
    avctx->sample_fmt = SAMPLE_FMT_S16;
1893

    
1894
//    dump_context(s);
1895
    return 0;
1896
}
1897

    
1898

    
1899
static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1900
{
1901
    QDM2Context *s = avctx->priv_data;
1902

    
1903
    ff_rdft_end(&s->rdft_ctx);
1904

    
1905
    return 0;
1906
}
1907

    
1908

    
1909
static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
1910
{
1911
    int ch, i;
1912
    const int frame_size = (q->frame_size * q->channels);
1913

    
1914
    /* select input buffer */
1915
    q->compressed_data = in;
1916
    q->compressed_size = q->checksum_size;
1917

    
1918
//  dump_context(q);
1919

    
1920
    /* copy old block, clear new block of output samples */
1921
    memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1922
    memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1923

    
1924
    /* decode block of QDM2 compressed data */
1925
    if (q->sub_packet == 0) {
1926
        q->has_errors = 0; // zero it for a new super block
1927
        av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1928
        qdm2_decode_super_block(q);
1929
    }
1930

    
1931
    /* parse subpackets */
1932
    if (!q->has_errors) {
1933
        if (q->sub_packet == 2)
1934
            qdm2_decode_fft_packets(q);
1935

    
1936
        qdm2_fft_tone_synthesizer(q, q->sub_packet);
1937
    }
1938

    
1939
    /* sound synthesis stage 1 (FFT) */
1940
    for (ch = 0; ch < q->channels; ch++) {
1941
        qdm2_calculate_fft(q, ch, q->sub_packet);
1942

    
1943
        if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1944
            SAMPLES_NEEDED_2("has errors, and C list is not empty")
1945
            return;
1946
        }
1947
    }
1948

    
1949
    /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1950
    if (!q->has_errors && q->do_synth_filter)
1951
        qdm2_synthesis_filter(q, q->sub_packet);
1952

    
1953
    q->sub_packet = (q->sub_packet + 1) % 16;
1954

    
1955
    /* clip and convert output float[] to 16bit signed samples */
1956
    for (i = 0; i < frame_size; i++) {
1957
        int value = (int)q->output_buffer[i];
1958

    
1959
        if (value > SOFTCLIP_THRESHOLD)
1960
            value = (value >  HARDCLIP_THRESHOLD) ?  32767 :  softclip_table[ value - SOFTCLIP_THRESHOLD];
1961
        else if (value < -SOFTCLIP_THRESHOLD)
1962
            value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1963

    
1964
        out[i] = value;
1965
    }
1966
}
1967

    
1968

    
1969
static int qdm2_decode_frame(AVCodecContext *avctx,
1970
            void *data, int *data_size,
1971
            AVPacket *avpkt)
1972
{
1973
    const uint8_t *buf = avpkt->data;
1974
    int buf_size = avpkt->size;
1975
    QDM2Context *s = avctx->priv_data;
1976

    
1977
    if(!buf)
1978
        return 0;
1979
    if(buf_size < s->checksum_size)
1980
        return -1;
1981

    
1982
    *data_size = s->channels * s->frame_size * sizeof(int16_t);
1983

    
1984
    av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
1985
       buf_size, buf, s->checksum_size, data, *data_size);
1986

    
1987
    qdm2_decode(s, buf, data);
1988

    
1989
    // reading only when next superblock found
1990
    if (s->sub_packet == 0) {
1991
        return s->checksum_size;
1992
    }
1993

    
1994
    return 0;
1995
}
1996

    
1997
AVCodec qdm2_decoder =
1998
{
1999
    .name = "qdm2",
2000
    .type = CODEC_TYPE_AUDIO,
2001
    .id = CODEC_ID_QDM2,
2002
    .priv_data_size = sizeof(QDM2Context),
2003
    .init = qdm2_decode_init,
2004
    .close = qdm2_decode_close,
2005
    .decode = qdm2_decode_frame,
2006
    .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
2007
};